- sr-users - kamailio.org

Re: [Serusers] 400 ul_add: flags expected
by Samuel Osorio Calvo 24 Jun '05

24 Jun '05

There is one field missing after replicate, called flag. It is a boolean parameter that I think it determines wether the UA is natted or not. So, the right command is: bash#serctl fifo ul_add {table} {AoR} {Contact} {Expires} {q} {replicate} {flags} so, as an example: serctl fifo ul_add location bob(a)biloxy.com bob(a)192.168.2.45 3600 1 0 0 Other useful commands are: bash#serctl fifo ul_rm {table} {AoR}--> which will remove all contacts for the given AoR and bash#serctl fifo ul_rm_contact {table} {AoR} {Contact} --> remove only the given contact for the AoR Hope it helps, Samuel Unclassified. >>> "Aisling" <ashling.odriscoll(a)cit.ie> 06/23/05 05:12PM >>> Hello, Has anyone experienced the above message when adding a permanent user location via the serctl fifo interface? I did a search and another post suggested it was due to a mismatch in the database due to customizing SER or because of serweb but because I haven't done anything like that, does anyone have another explanation? Also would anyone know the exact format for setting the q value with this command? According to previous correspondence this might be possible ("I think the current usrloc fifo commands provide enough flexibility to modify the q value, I don't know now the exact spelling but something like: serctl ul add location user@domain user@ip {expires} {q} {replicate}") Many Thanks, Aisling. _______________________________________ Aisling O' Driscoll Research Student Centre for Adaptive Wireless Systems Cork Institute of Technology. Email: ashling.odriscoll(a)cit.ie -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] google summer of code for opensource
by Iqbal 24 Jun '05

24 Jun '05

seeing that we have been having lots of discussions about opensource , and forking etc, maybe we could get some resources from here if anyone is interested http://code.google.com/summerofcode.html Iqbal

2 1

[Serusers] 486 Busy Here
by Ramu Yadav 24 Jun '05

24 Jun '05

Hi all, I am facing a problem. i.e: When I dial a number ex: (from 9005 to 9010). In this case 9010 reject that call then that guy will send "486 Busy Here" or "603 Decline" . (This is the first phone) So far so good. When I reject the 2nd phone will ring immediately but it didn't, it is ringing after 20 seconds. Here my "fr_inv_timer", 20 seconds. Thanks -- Ramu Yadav

1 0

[Serusers] Onsip doc
by harry gaillac 24 Jun '05

24 Jun '05

Hello, May be a good link for ONSIP http://www.tech-invite.com/ Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 1

[Serusers] call test generator
by Ernesto Lucero 24 Jun '05

24 Jun '05

Hi Does anybody has or knows of a very good TEST CALL GENERATOR i am trying to make test to the network where i am deploying my sip service Any help will be great Thanks moyotlan

3 2

[Serusers] SER losing SIP registrations
by Juan Carlos Castro y Castro 24 Jun '05

24 Jun '05

Our company has a call center implementation on a client with up to 100 support personnel using X-Lite version 1050 softphones running under Windows 98. Until a few days ago, the softphones were logged in directly on our PBXs. Now, they log onto a separate SER 0.8.14 box and the PBXs forward calls to SER. That was needed to unify queue management. It works. But some softphones are being randomly kicked out of SER, it seems SER isn't receiving the refresh REGISTER messages from the softphones. The re-register timeout is set to 500 seconds on the softphones. There's a lot of "removing spare zombie" and "Binding '<user>','<url>' has expired" messages in /var/log/messages. For now, we're instructing the client to increase the timeout to 10 hours on the softphones in which the problem happens most often. I don't know if that's really the right thing to do, I think we should somehow make sure the re-registers are done in a timely fashion and retried, but I could not find ant SER configuration option related to that. What should I do? Another thing: the softphones take a horribly high amount of time to log in, but that may be because X-Lite sucks. Regards, Juan

7 9

[Serusers] forking stable, unstable....just a thought how it could be done
by Iqbal 23 Jun '05

23 Jun '05

hi without wanting to start up last week discussions i thought this might be useful ------------------------------------- HEAD and STABLE In the early development of Asterisk, there was a single branch. Releases were few and far between, and the generally accepted way to obtain and use Asterisk was via CVS. While this is fine for people developing Asterisk, the others who wished to run the software in a production environment sometimes had to contend with broken branches, and not really sure what they were getting from one day to the next. At Astricon (http://www.astricon.net), the first official STABLE version 1.0 was released. The STABLE branch was born. Now there were two separate branches of Asterisk. There was HEAD which the developers would use to test patches and new features, and there was STABLE, which people could run in production and where only bug fixes and proven features could be ported over. However this caused two independent versions of code. While this is good for stability, reliability and development, it also doubles the development overhead. Every bug fix which is submitted and resolved on the bug tracker (http://bugs.digium.com) must then be ported to the stable branch. And the longer that the development and stable branches co-exist, the more diverse their code bases become. This can cause patches which apply cleanly to HEAD to not apply to STABLE. This is where the Asterisk Development Proxy comes in. Russell Bryant is the maintainer of the STABLE branch, and thus, must port every single resolved bug from HEAD into STABLE. This is an enormous under taking, and the purpose of formation of the Asterisk Development Proxy. Many of these patches can be ported from HEAD to STABLE with basic or average coding skills. It is also an excellent opportunity for developers unfamiliar with the Asterisk code base to get involved and help with something while they are getting their feet wet. So join the Asterisk Development Proxy and Asterisk Documentation Project team and get involved with Asterisk today! Retrieved from "http://dev.asteriskdocs.org/index.php/HEAD_and_STABLE" ------------------------------------------------------------ Iqbal

1 0

[Users] unified pseudo-variables in avpops and xlog
by Daniel-Constantin Mierla 23 Jun '05

23 Jun '05

Hello, the avpops and xlog module are using now same format for specifiers to print formatted strings. avpops has a new function avp_printf(dest, format) which stores in the AVP 'dest' the string resulted after substitution of the specifiers from "format" parameter. The old marker for specifiers '%' has been replaced by '$' which is more common for variables. To specify a header or an AVP you have to use now $hdr(header_name[index]) or $avp(avp_id[index]). The list with all available pseudo-variables in OpenSER is posted at: http://openser.org/docs/pseudo-variables.html avp_printf() allow to compose a value from many parts of a SIP message, other avps and many other values taken from system. For example: avp_printf("i:20", "the time is $Tf, the request is a $rm and has the from header $hdr(from)"); The documentation for avpops and xlog module has been updated on CVS, too. Note that these features are available only on CVS head (development version - http://openser.org/index.php#download). Online documentation site for modules (http://openser.org/docs/) has been split in 0.9x (stable) and 0.10.x (devel) directories. Follows the integration of same format for pseudo-variables in textops, acc and tm. We will appreciate any feedback. Daniel

1 0

[Serusers] AVP on subst_uri()
by Miguel Angel Villar 23 Jun '05

23 Jun '05

Hi There. I'm using subst_uri() to make some URI transformation and is working great: subst_uri('/^sip:([0-9]+)@(.*)$/sip:011\1@\2/i'); Now I'm trying to insert an AVP value in the substitution but is not working: avp_push("011","var"); subst_uri('/^sip:([0-9]+)@(.*)$/sip:$var\1@\2/i'); When I do this the replaced URI has the string $var instead of the value of it (011). Can I do something like this in subst_uri()? Looking at the Textops User's guide I can see that there is some kind of parameter association for subst_uri(). <<subst_uri().txt>> Regards. MaV.-

1 0

[Users] two issues: ser.cfg or openser.cfg and mysql password?
by Jose Bertuzzi 23 Jun '05

23 Jun '05

Hello guys, first of all congrats for this new idea of openser. I completly agree with you. I tried on june 14, the debian stable package and found that after installing it the config file created is openser.cfg but the program calls ser.cfg. I've just renamed it and start working. The other issue is that i don't know what is the mysql password to add users using openserctl add. I tried with heslo, but failed. So I changed the passwords on mysql and used the command modparam("acc", "db_url", "mysql://user:password@localhost/openser") but did not work. I also tried with the FreeBSD port and compiling src, with same results. What can I do? Thanks a lot for your help. Pablo. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

4 5

[Serusers] 400 ul_add: flags expected
by Aisling 23 Jun '05

23 Jun '05

Hello, Has anyone experienced the above message when adding a permanent user location via the serctl fifo interface? I did a search and another post suggested it was due to a mismatch in the database due to customizing SER or because of serweb but because I haven't done anything like that, does anyone have another explanation? Also would anyone know the exact format for setting the q value with this command? According to previous correspondence this might be possible ("I think the current usrloc fifo commands provide enough flexibility to modify the q value, I don't know now the exact spelling but something like: serctl ul add location user@domain user@ip {expires} {q} {replicate}") Many Thanks, Aisling. _______________________________________ Aisling O' Driscoll Research Student Centre for Adaptive Wireless Systems Cork Institute of Technology. Email: ashling.odriscoll(a)cit.ie -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] problems when going to pstn
by Lucas Aimaretto 23 Jun '05

23 Jun '05

Hi all, I'm writting to ask for suggestion. I'm able to communicate sip-to-sip using SER. But when one user wants to talk with somebody in the pstn, I receive two different errors, depending on the gateway. Look ... This first message came from a Mera sip server. U IP_MERA:5060 -> IP_SER:5060 SIP/2.0 415 Unsupported Media Type. v: SIP/2.0/UDP IP_SER:5060;branch=z9hG4bKad03.0a8597a1.0. v: SIP/2.0/UDP IP_UA:15172;branch=z9hG4bK2413941036. Record-Route: <sip:IP_MERA:5060;maddr=38.117.22.34>. Record-Route: <sip:IP_SER;ftag=2859935538;lr=on>. f: "1991117" <sip:1991117@IP_SER>;tag=2859935538. t: <sip:011543515684778@IP_SER>;tag=2eff4b002937ff10ff0000045a5b1432. i: 6ad7a895fbb61ed09f5b2ea36b6612ce(a)192.168.1.30. CSeq: 1 INVITE. P-Session-Timeout: 7200. Server: MERA MSIP v.1.0.1. l: 0. This one from a cisco AS5350 U IP_CISCO:5060 -> IP_SER:5060 SIP/2.0 400 Bad Request - 'Invalid SDP information'. v: SIP/2.0/UDP IP_SER;branch=z9hG4bK2817.1aa690e7.0. v: SIP/2.0/UDP IP_UA:15188;branch=z9hG4bK3914033240. f: "1991117" <sip:1991117@IP_SER>;tag=2962826014. t: <sip:3055383346@IP_SER>. i: 332ee2e17a14427caeb2feaf2da22a3a(a)192.168.1.30. CSeq: 2 INVITE. P-Session-Timeout: 7200. l: 0. I do not get it. Why could I communicate sip-to-sip just fine, but, when going to pstn, I receive this errors. And, moreover, they are different. I'm only having this error with this particular user, which is behind a router doing NAT. Do you have any ideas of what could I do to solve this ... ? Or what could be the problem ? Best regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.7.11/26 - Release Date: 22/06/2005

1 0

[Serusers] rtpproxy + nathelper + radius + mediaproxy + voicemail + asterisk + serweb + ser-0.9.3-fc1.0 and it is running fine. (ito.)
by vm_8000＠voice.dyndns.tv 23 Jun '05

23 Jun '05

rtpproxy + nathelper + radius + mediaproxy + voicemail + asterisk + serweb + ser-0.9.3-fc1.0 and it is running fine. (ito.) -rw-r--r-- 1 root root 835211 6月 23 18:35 ser-0.9.3-fc1.0.i386.rpm -rw-r--r-- 1 root root 1909414 6月 23 18:35 ser-0.9.3-fc1.0.src.rpm -rw-r--r-- 1 root root 1902353 6月 23 18:41 ser-0.9.3_src.tar.gz -rw-r--r-- 1 root root 3470729 6月 23 18:35 ser-debuginfo-0.9.3-fc1.0.i386 .rpm -rw-r--r-- 1 root root 64601 6月 23 18:35 ser-jabber-0.9.3-fc1.0.i386.rpm -rw-r--r-- 1 root root 111690 6月 23 18:35 ser-mysql-0.9.3-fc1.0.i386.rpm -rw-r--r-- 1 root root 21406 6月 23 18:35 ser-radius-0.9.3-fc1.0.i386.rpm ///////////////////////////////////////////// [cycle #: 10; if constant make sure server lives and fifo is on] Server: Sip EXpress router (0.9.3-0 (i386/linux)) Now: Thu Jun 23 22:04:39 2005 Up Since: Thu Jun 23 21:34:44 2005 Up time: 1795 [sec] Transaction Statistics Current: 0 (0 waiting) Total: 0 (0 local) Replied localy: 0 Completion status 6xx: 0, 5xx: 0, 4xx: 0, 3xx: 0,2xx: 0 Stateless Server Statistics 200: 19 202: 0 2xx: 0 300: 0 301: 0 302: 0 3xx: 0 400: 0 401: 19 403: 0 404: 0 407: 0 408: 0 483: 0 4xx: 0 500: 0 5xx: 0 6xx: 0 xxx: 0 failures: 0 UsrLoc Stats Domain Registered Expired 'aliases' 19 0 'location' 10 1 ///////////////////////////////////////////// http://www.voice.dyndns.tv sip:admin at iptel.org Saitama Japan Sip EXpress router Support Team voice at wmail.plala.or.jp /////////////////////////////////////////////

1 0

[serusers] calling problem
by michael p 23 Jun '05

23 Jun '05

hi in my table location of SER i have a field flags, someone knows for what is? now i have three users registered and two can call and their flags are "0" and one can't call and is flags field is "1" is that the reason why it don't work? thanks M.

2 1

[Serusers] Implementing Module with Osip
by Alexandre Passito 23 Jun '05

23 Jun '05

Hi, I'm trying to implement a module to Openser that should use parts of the Osip (sip stack implementation). That module could be a "black list", "follow me" and etc. Did somebody implement it already? Does somebody have an ideia how to begin to implement? Where can I find some information about it? Best regards, -- Alexandre Passito Universidade Federal do Amazonas Departamento de Ciência da Computação LabVoIP - GRSD

1 0

[Serusers] Server header in SER responses
by Fermín Galán Márquez 23 Jun '05

23 Jun '05

Hello, SER includes a "Server" header in responses (for examples, in the 200 OK responding a REGISTER when SER acting as registrar server). Server: Sip Express router (0.8.14 (i386/linux)) Is there any way to modify or/and remove the header? I have looked through the SIP Admin Guide (http://www.iptel.org/ser/doc/seruser/seruser.html) but I haven't found nothing related with this. (I know I could modify the "Server" header editing the SERVER_HDR directive in config.h and recompile, but I think is too radical :) Thanks in advance! Best regards, ------ Fermín Agora Systems, S. A.

2 1

[Serusers] PRACK
by Dana Turcanu 23 Jun '05

23 Jun '05

Hello, I have a problem with tow Cisco ATA 186 both having version v3.2. When I try to make a call on my side I hear that the phone is ringing 3 times and then the call is disconnected. On the other side, the telephone is ringing 10 times. I belive that the problem is when one Cisco sends a PRACK packet and and it receive "420 Bad request" from de sip server. Then the call is disconnected with error "504 Gateway Time-out". Below I put the logs from the tow ATAs. Any help in solving this problem will be appreciated. Thanks. Call from: 3327xxxx to 33271yyyy (log from the calling ATA): ===================================================== 050620|11:48:28 Calling 0332yyyy 050620|11:48:28 SCC Cmd[0:0]CLIP 050620|11:48:28 SCC Cmd[0:0]CALL 050620|11:48:28 [0]INVITE Retry 0 050620|11:48:28 [0:0]Tx Msg to ip_sip_proxy:5060 050620|11:48:28 050620|11:48:28 INVITE sip:0332yyyy@domain.com;user=phone SIP/2.0 050620|11:48:28 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bK5ce290af84b14296 050620|11:48:28 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:28 To: <sip:0332yyyy@domain.com;user=phone> 050620|11:48:28 Call-ID: 2133411222@ip_ata_1 050620|11:48:28 CSeq: 1 INVITE 050620|11:48:28 Contact: <sip:332xxxx@ip_ata_1:5060;user=phone;transport=udp> 050620|11:48:28 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:28 Expires: 300 050620|11:48:28 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:28 Supported: 100rel,replaces 050620|11:48:28 Content-Length: 236 050620|11:48:28 Content-Type: application/sdp 050620|11:48:28 050620|11:48:28 v=0 050620|11:48:28 o=332xxxx 57793347 57793347 IN IP4 ip_ata_1 050620|11:48:28 s=ATA186 Call 050620|11:48:28 c=IN IP4 ip_ata_1 050620|11:48:28 t=0 0 050620|11:48:28 m=audio 16384 RTP/AVP 8 0 101 050620|11:48:28 a=rtpmap:8 PCMA/8000/1 050620|11:48:28 a=rtpmap:0 PCMU/8000/1 050620|11:48:28 a=rtpmap:101 telephone-event/8000 050620|11:48:28 a=fmtp:101 0-15 050620|11:48:28 050620|11:48:28 [0]Rx Msg from ip_sip_server:5060 050620|11:48:28 050620|11:48:28 SIP/2.0 407 Proxy Authentication Required 050620|11:48:28 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bK5ce290af84b14296;rport=5060;received=ip_ata_1_nat 050620|11:48:28 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:28 To: <sip:0332yyyy@domain.com;user=phone>;tag=fee886471600e72424ce85e67d30b332.6542 050620|11:48:28 Call-ID: 2133411222@ip_ata_1 050620|11:48:28 CSeq: 1 INVITE 050620|11:48:28 Proxy-Authenticate: Digest realm="domain.com", nonce="rrrr" 050620|11:48:28 Server: Sip EXpress router (0.8.14-5 (i386/linux)) 050620|11:48:28 Content-Length: 0 050620|11:48:28 050620|11:48:28 050620|11:48:28 Response=407; Proxy Authentication Required 050620|11:48:28 [0]Tx INVITE to ip_sip_server:5060 050620|11:48:28 050620|11:48:28 ACK sip:0332yyyy@domain.com;user=phone SIP/2.0 050620|11:48:28 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bK5ce290af84b14296 050620|11:48:28 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:28 To: <sip:0332yyyy@domain.com;user=phone>;tag=fee886471600e72424ce85e67d30b332.6542 050620|11:48:28 Call-ID: 2133411222@ip_ata_1 050620|11:48:28 CSeq: 1 ACK 050620|11:48:28 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:28 Content-Length: 0 050620|11:48:28 050620|11:48:28 050620|11:48:28 [0]INVITE Retry 0 050620|11:48:28 [0:0]Tx Msg to ip_sip_server:5060 050620|11:48:28 050620|11:48:28 INVITE sip:0332yyyy@domain.com;user=phone SIP/2.0 050620|11:48:28 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bKf33e54a2a402e527 050620|11:48:28 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:28 To: <sip:0332yyyy@domain.com;user=phone> 050620|11:48:28 Call-ID: 2133411222@ip_ata_1 050620|11:48:28 CSeq: 2 INVITE 050620|11:48:28 Contact: <sip:332xxxx@ip_ata_1:5060;user=phone;transport=udp> 050620|11:48:28 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:28 Proxy-Authorization: Digest username="332xxxx",realm="domain.com",nonce="rrrr",uri="sip:03 32yyyy(a)domain.com",response="ihuhuh" 050620|11:48:28 Expires: 300 050620|11:48:28 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:28 Supported: 100rel,replaces 050620|11:48:28 Content-Length: 236 050620|11:48:28 Content-Type: application/sdp 050620|11:48:28 050620|11:48:28 v=0 050620|11:48:28 o=332xxxx 57793355 57793355 IN IP4 ip_ata_1 050620|11:48:28 s=ATA186 Call 050620|11:48:28 c=IN IP4 ip_ata_1 050620|11:48:28 t=0 0 050620|11:48:28 m=audio 16384 RTP/AVP 8 0 101 050620|11:48:28 a=rtpmap:8 PCMA/8000/1 050620|11:48:28 a=rtpmap:0 PCMU/8000/1 050620|11:48:28 a=rtpmap:101 telephone-event/8000 050620|11:48:28 a=fmtp:101 0-15 050620|11:48:28 050620|11:48:28 [0]Rx Msg from ip_sip_server:5060 050620|11:48:28 050620|11:48:28 SIP/2.0 100 Trying 050620|11:48:28 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bKf33e54a2a402e527;rport=5060;received=ip_ata_1_nat 050620|11:48:28 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:28 To: <sip:0332yyyy@domain.com;user=phone> 050620|11:48:28 Call-ID: 2133411222@ip_ata_1 050620|11:48:28 CSeq: 2 INVITE 050620|11:48:28 Server: Sip EXpress router (0.8.14-5 (i386/linux)) 050620|11:48:28 Content-Length: 0 050620|11:48:28 050620|11:48:28 050620|11:48:28 Response=100; Trying 050620|11:48:29 [0]Rx Msg from ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 100 trying -- your call is important to us 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bKf33e54a2a402e527;rport=5060;received=ip_ata_1_nat 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone> 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Server: Sip EXpress router (0.8.14-5 (i386/linux)) 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 Response=100; Trying 050620|11:48:29 [0]Rx Msg from ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 180 Ringing 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Require: 100rel 050620|11:48:29 RSeq: 1 050620|11:48:29 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:29 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 Response=180; Ringing 050620|11:48:29 SCC:ev=10[0:0] 3 0 050620|11:48:29 [0]StartTone 3 050620|11:48:29 [0]PRACK Retry 0 050620|11:48:29 [0:0]Tx Msg to ip_sip_server:5060 050620|11:48:29 050620|11:48:29 PRACK sip:0332yyyy@domain.com;user=phone SIP/2.0 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bK2b0f7845ea7a8edb 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 3 PRACK 050620|11:48:29 RAck: 1 2 INVITE 050620|11:48:29 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Proxy-Authorization: Digest username="332xxxx",realm="domain.com",nonce="rrrr",uri="sip:03 32yyyy(a)domain.com",response="tttt" 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 [0]Rx Msg from ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 400 Bad request 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bK2b0f7845ea7a8edb;rport=5060;received=ip_ata_1_nat 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 3 PRACK 050620|11:48:29 Server: Sip EXpress router (0.8.14-5 (i386/linux)) 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 [0]Rx Msg from ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 180 Ringing 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Require: 100rel 050620|11:48:29 RSeq: 1 050620|11:48:29 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:29 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 Response=180; Ringing 050620|11:48:30 [0]Rx Msg from ip_sip_server:5060 050620|11:48:30 050620|11:48:30 SIP/2.0 180 Ringing 050620|11:48:30 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:30 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:30 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:30 Call-ID: 2133411222@ip_ata_1 050620|11:48:30 CSeq: 2 INVITE 050620|11:48:30 Require: 100rel 050620|11:48:30 RSeq: 1 050620|11:48:30 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:30 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:30 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:30 Content-Length: 0 050620|11:48:30 050620|11:48:30 050620|11:48:30 Response=180; Ringing 050620|11:48:32 [0]Rx Msg from ip_sip_server:5060 050620|11:48:32 050620|11:48:32 SIP/2.0 180 Ringing 050620|11:48:32 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:32 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:32 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:32 Call-ID: 2133411222@ip_ata_1 050620|11:48:32 CSeq: 2 INVITE 050620|11:48:32 Require: 100rel 050620|11:48:32 RSeq: 1 050620|11:48:32 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:32 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:32 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:32 Content-Length: 0 050620|11:48:32 050620|11:48:32 050620|11:48:33 Response=180; Ringing 050620|11:48:35 [0]Rx Msg from ip_sip_server:5060 050620|11:48:35 050620|11:48:35 SIP/2.0 180 Ringing 050620|11:48:35 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:35 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:35 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:35 Call-ID: 2133411222@ip_ata_1 050620|11:48:35 CSeq: 2 INVITE 050620|11:48:35 Require: 100rel 050620|11:48:35 RSeq: 1 050620|11:48:35 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:35 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:35 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:35 Content-Length: 0 050620|11:48:35 050620|11:48:35 050620|11:48:35 Response=180; Ringing 050620|11:48:37 [0]Rx Msg from ip_sip_server:5060 050620|11:48:37 050620|11:48:37 SIP/2.0 180 Ringing 050620|11:48:37 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:37 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:37 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:37 Call-ID: 2133411222@ip_ata_1 050620|11:48:37 CSeq: 2 INVITE 050620|11:48:37 Require: 100rel 050620|11:48:37 RSeq: 1 050620|11:48:37 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:37 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:37 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:37 Content-Length: 0 050620|11:48:37 050620|11:48:37 050620|11:48:37 Response=180; Ringing 050620|11:48:39 [0]Rx Msg from ip_sip_server:5060 050620|11:48:39 050620|11:48:39 SIP/2.0 180 Ringing 050620|11:48:39 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:39 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:39 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:39 Call-ID: 2133411222@ip_ata_1 050620|11:48:39 CSeq: 2 INVITE 050620|11:48:39 Require: 100rel 050620|11:48:39 RSeq: 1 050620|11:48:39 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:39 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:39 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:39 Content-Length: 0 050620|11:48:39 050620|11:48:39 050620|11:48:39 Response=180; Ringing 050620|11:48:41 [0]Rx Msg from ip_sip_server:5060 050620|11:48:41 050620|11:48:41 SIP/2.0 180 Ringing 050620|11:48:41 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:41 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:41 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:41 Call-ID: 2133411222@ip_ata_1 050620|11:48:41 CSeq: 2 INVITE 050620|11:48:41 Require: 100rel 050620|11:48:41 RSeq: 1 050620|11:48:41 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:41 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:41 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:41 Content-Length: 0 050620|11:48:41 050620|11:48:41 050620|11:48:41 Response=180; Ringing 050620|11:48:43 [0]Rx Msg from ip_sip_server:5060 050620|11:48:43 050620|11:48:43 SIP/2.0 504 Gateway Time-out 050620|11:48:43 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:43 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:43 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:43 Call-ID: 2133411222@ip_ata_1 050620|11:48:43 CSeq: 2 INVITE 050620|11:48:43 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:43 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:43 Content-Length: 0 050620|11:48:43 050620|11:48:43 050620|11:48:43 Response=504; Gateway Time-out 050620|11:48:43 [0:0]Tx Msg to ip_sip_server:5060 050620|11:48:43 050620|11:48:43 ACK sip:0332yyyy@domain.com;user=phone SIP/2.0 050620|11:48:43 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bK3504d15d5210ae37 050620|11:48:43 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:43 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:43 Call-ID: 2133411222@ip_ata_1 050620|11:48:43 CSeq: 2 ACK 050620|11:48:43 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:43 Proxy-Authorization: Digest username="332xxxx",realm="domain.com",nonce="tttt",uri="sip:03 32yyyy(a)domain.com",response="rrrr" 050620|11:48:43 Content-Length: 0 050620|11:48:43 050620|11:48:43 050620|11:48:43 change next hop[0]: c = 4 050620|11:48:43 All PriObProxy down 050620|11:48:43 No AltObProxy 050620|11:48:43 INVITE failed 050620|11:48:43 SCC:ev=7[0:0] 3 0 050620|11:48:43 [0]StartTone 2 050620|11:48:43 [0]Rx Msg from ip_sip_server:5060 050620|11:48:43 050620|11:48:43 SIP/2.0 504 Gateway Time-out 050620|11:48:43 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:43 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:43 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:43 Call-ID: 2133411222@ip_ata_1 050620|11:48:43 CSeq: 2 INVITE 050620|11:48:43 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:43 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:43 Content-Length: 0 050620|11:48:43 050620|11:48:43 050620|11:48:45 [0]Rx Msg from ip_sip_server:5060 050620|11:48:45 050620|11:48:45 SIP/2.0 504 Gateway Time-out 050620|11:48:45 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:45 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:45 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:45 Call-ID: 2133411222@ip_ata_1 050620|11:48:45 CSeq: 2 INVITE 050620|11:48:45 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:45 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:45 Content-Length: 0 050620|11:48:45 050620|11:48:45 050620|11:48:49 [0]Rx Msg from ip_sip_server:5060 050620|11:48:49 050620|11:48:49 SIP/2.0 504 Gateway Time-out 050620|11:48:49 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:49 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:49 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:49 Call-ID: 2133411222@ip_ata_1 050620|11:48:49 CSeq: 2 INVITE 050620|11:48:49 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:49 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:49 Content-Length: 0 050620|11:48:49 050620|11:48:49 050620|11:48:53 [0]Rx Msg from ip_sip_server:5060 050620|11:48:53 050620|11:48:53 SIP/2.0 504 Gateway Time-out 050620|11:48:53 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:53 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:53 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:53 Call-ID: 2133411222@ip_ata_1 050620|11:48:53 CSeq: 2 INVITE 050620|11:48:53 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:53 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:53 Content-Length: 0 050620|11:48:53 050620|11:48:53 050620|11:48:54 [0]StopTone 050620|11:48:54 SCC Cmd[0:0]RELEASE 050620|11:48:54 CCBS expired 050620|11:48:54 >>> SIP Soft Reset [0:0] <<< 050620|11:48:54 [0]MPT mode 0 ======================================================= Call from 3327xxxx to 3327yyyy ( log from the called ATA): ======================================================= 050620|11:48:29 INVITE sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp SIP/2.0 050620|11:48:29 Max-Forwards: 10 050620|11:48:29 Record-Route: <sip:0332yyyy@ip_sip_server;ftag=2064987569;lr=on> 050620|11:48:29 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone> 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Contact: <sip:332xxxx@ip_ata_1_nat:5060;user=phone;transport=udp> 050620|11:48:29 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Expires: 300 050620|11:48:29 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:29 Supported: 100rel,replaces 050620|11:48:29 Content-Length: 235 050620|11:48:29 Content-Type: application/sdp 050620|11:48:29 050620|11:48:29 v=0 050620|11:48:29 o=332xxxx 57793355 57793355 IN IP4 ip_ata_1 050620|11:48:29 s=ATA186 Call 050620|11:48:29 c=IN IP4 81.180.201.50 050620|11:48:29 t=0 0 050620|11:48:29 m=audio 36540 RTP/AVP 8 0 101 050620|11:48:29 a=rtpmap:8 PCMA/8000/1 050620|11:48:29 a=rtpmap:0 PCMU/8000/1 050620|11:48:29 a=rtpmap:101 telephone-event/8000 050620|11:48:29 a=fmtp:101 0-15 050620|11:48:29 050620|11:48:29 AVT: payload=101 050620|11:48:29 [1:0]Tx Msg to ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 100 Trying 050620|11:48:29 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 SCC:ev=21[1:0] 0 0 050620|11:48:29 SCC:ev=5[1:0] 13 0 050620|11:48:29 [1:0]RINGING 050620|11:48:29 SCC Cmd[1:0]ALERT 050620|11:48:29 [1]R2INV Retry 0 050620|11:48:29 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 180 Ringing 050620|11:48:29 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Require: 100rel 050620|11:48:29 RSeq: 1 050620|11:48:29 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:29 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 [1]R2INV Retry 1 050620|11:48:29 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 180 Ringing 050620|11:48:29 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Require: 100rel 050620|11:48:29 RSeq: 1 050620|11:48:29 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:29 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:30 [1]R2INV Retry 2 050620|11:48:30 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:30 050620|11:48:30 SIP/2.0 180 Ringing 050620|11:48:30 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:30 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:30 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:30 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:30 Call-ID: 2133411222@ip_ata_1 050620|11:48:30 CSeq: 2 INVITE 050620|11:48:30 Require: 100rel 050620|11:48:30 RSeq: 1 050620|11:48:30 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:30 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:30 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:30 Content-Length: 0 050620|11:48:30 050620|11:48:30 050620|11:48:32 [1]R2INV Retry 3 050620|11:48:32 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:32 050620|11:48:32 SIP/2.0 180 Ringing 050620|11:48:32 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:32 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:32 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:32 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:32 Call-ID: 2133411222@ip_ata_1 050620|11:48:32 CSeq: 2 INVITE 050620|11:48:32 Require: 100rel 050620|11:48:32 RSeq: 1 050620|11:48:32 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:32 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:32 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:32 Content-Length: 0 050620|11:48:32 050620|11:48:32 050620|11:48:34 [1]R2INV Retry 4 050620|11:48:34 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:34 050620|11:48:34 SIP/2.0 180 Ringing 050620|11:48:34 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:34 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:34 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:34 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:34 Call-ID: 2133411222@ip_ata_1 050620|11:48:34 CSeq: 2 INVITE 050620|11:48:34 Require: 100rel 050620|11:48:34 RSeq: 1 050620|11:48:34 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:34 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:34 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:34 Content-Length: 0 050620|11:48:34 050620|11:48:34 050620|11:48:36 [1]R2INV Retry 5 050620|11:48:36 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:36 050620|11:48:36 SIP/2.0 180 Ringing 050620|11:48:36 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:36 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:36 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:36 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:36 Call-ID: 2133411222@ip_ata_1 050620|11:48:36 CSeq: 2 INVITE 050620|11:48:36 Require: 100rel 050620|11:48:36 RSeq: 1 050620|11:48:36 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:36 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:36 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:36 Content-Length: 0 050620|11:48:36 050620|11:48:36 050620|11:48:38 [1]R2INV Retry 6 050620|11:48:38 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:38 050620|11:48:38 SIP/2.0 180 Ringing 050620|11:48:38 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:38 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:39 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:39 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:39 Call-ID: 2133411222@ip_ata_1 050620|11:48:39 CSeq: 2 INVITE 050620|11:48:39 Require: 100rel 050620|11:48:39 RSeq: 1 050620|11:48:39 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:39 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:39 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:39 Content-Length: 0 050620|11:48:39 050620|11:48:39 050620|11:48:41 [1]R2INV Retry 7 050620|11:48:41 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:41 050620|11:48:41 SIP/2.0 180 Ringing 050620|11:48:41 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:41 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:41 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:41 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:41 Call-ID: 2133411222@ip_ata_1 050620|11:48:41 CSeq: 2 INVITE 050620|11:48:41 Require: 100rel 050620|11:48:41 RSeq: 1 050620|11:48:41 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:41 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:41 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:41 Content-Length: 0 050620|11:48:41 050620|11:48:41 050620|11:48:43 R2INV TO; NoAltProxy 050620|11:48:43 [1:0]Provisional Response Timeout 050620|11:48:43 [1:0]Tx Msg to ip_sip_server:5060 050620|11:48:43 050620|11:48:43 SIP/2.0 504 Gateway Time-out 050620|11:48:43 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:43 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:43 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:43 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:43 Call-ID: 2133411222@ip_ata_1 050620|11:48:43 CSeq: 2 INVITE 050620|11:48:43 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:43 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:43 Content-Length: 0 050620|11:48:43 050620|11:48:43 050620|11:48:43 reiniting next hop 050620|11:48:43 Refresh DNS for sip.domain.com aaaaaaaa 0 0 0 0 050620|11:48:43 Trying 1st IP ADDR aaaaaaaa 050620|11:48:43 [1]Rx Msg from ip_sip_server:5060 050620|11:48:43 050620|11:48:43 ACK sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp SIP/2.0 050620|11:48:43 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:43 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:43 Call-ID: 2133411222@ip_ata_1 050620|11:48:43 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:43 CSeq: 2 ACK 050620|11:48:43 User-Agent: Sip EXpress router 050620|11:48:43 Content-Length: 0 050620|11:48:43 050620|11:48:43 050620|11:48:43 RING: got ACK 050620|11:48:43 9852:00;2,2,2,0, 050620|11:48:44 Recv[0]: 60 2906 464960 0 94 0 0 0 050620|11:48:44 Tx[0]: 60 2999 479840 050620|11:48:50 [1]REGISTER Retry 0 050620|11:48:50 [1:0]Tx Msg to ip_sip_server:5060 050620|11:48:50 050620|11:48:50 REGISTER sip:domain.com SIP/2.0 050620|11:48:50 Via: SIP/2.0/UDP ip_ata_2:5060;branch=z9hG4bKb417bd8a1c3112b7 050620|11:48:50 From: <sip:332yyyy@domain.com;user=phone>;tag=2612509671 050620|11:48:50 To: <sip:332yyyy@domain.com;user=phone> 050620|11:48:50 Call-ID: 2749858814@ip_ata_2 050620|11:48:50 CSeq: 355 REGISTER 050620|11:48:50 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp>;expires=3600 050620|11:48:50 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:50 Authorization: Digest username="332yyyy",realm="domain.com",nonce="42b678c1c05ddd4c6ff925bea0c9bb923c5ec1ee",uri="sip:domain.c om",response="asdasdd" 050620|11:48:50 Content-Length: 0 050620|11:48:50 050620|11:48:50 050620|11:48:50 [1]Rx Msg from ip_sip_server:5060 050620|11:48:50 050620|11:48:50 SIP/2.0 401 Unauthorized 050620|11:48:50 Via: SIP/2.0/UDP ip_ata_2:5060;branch=z9hG4bKb417bd8a1c3112b7 050620|11:48:50 From: <sip:332yyyy@domain.com;user=phone>;tag=2612509671 050620|11:48:50 To: <sip:332yyyy@domain.com;user=phone>;tag=fee886471600e72424ce85e67d30b332.919f 050620|11:48:50 Call-ID: 2749858814@ip_ata_2 050620|11:48:50 CSeq: 355 REGISTER 050620|11:48:50 WWW-Authenticate: Digest realm="domain.com", nonce="rrrr", stale=true 050620|11:48:50 Server: Sip EXpress router 050620|11:48:50 Content-Length: 0 050620|11:48:50 050620|11:48:50 050620|11:48:50 [1]Reg Resp 401; Unauthorized 050620|11:48:50 [1]REGISTER Retry 0 050620|11:48:50 [1:0]Tx Msg to ip_sip_server:5060 050620|11:48:50 050620|11:48:50 REGISTER sip:domain.com SIP/2.0 050620|11:48:50 Via: SIP/2.0/UDP ip_ata_2:5060;branch=z9hG4bK99467a1aab648880 050620|11:48:50 From: <sip:332yyyy@domain.com;user=phone>;tag=2612509671 050620|11:48:50 To: <sip:332yyyy@domain.com;user=phone> 050620|11:48:50 Call-ID: 2749858814@ip_ata_2 050620|11:48:50 CSeq: 356 REGISTER 050620|11:48:50 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp>;expires=3600 050620|11:48:50 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:50 Authorization: Digest username="332yyyy",realm="domain.com",nonce="rrrr",uri="sip:domain.c om",response="sdfsdfsdffd" 050620|11:48:50 Content-Length: 0 050620|11:48:50 050620|11:48:50 050620|11:48:50 [1]Rx Msg from ip_sip_server:5060 050620|11:48:50 050620|11:48:50 SIP/2.0 200 OK 050620|11:48:50 Via: SIP/2.0/UDP ip_ata_2:5060;branch=z9hG4bK99467a1aab648880 050620|11:48:50 From: <sip:332yyyy@domain.com;user=phone>;tag=2612509671 050620|11:48:50 To: <sip:332yyyy@domain.com;user=phone>;tag=fee886471600e72424ce85e67d30b332.6927 050620|11:48:50 Call-ID: 2749858814@ip_ata_2 050620|11:48:50 CSeq: 356 REGISTER 050620|11:48:50 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp>;q=0.00;expires=3600 050620|11:48:50 Server: Sip EXpress router 050620|11:48:50 Content-Length: 0 050620|11:48:50 050620|11:48:50 050620|11:48:50 [1]Reg Resp 200; OK 050620|11:48:50 [1]Reg OK (3600) =========================================================== __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

3 3

[Serusers] "ser-0.9.0.GettingStarted.1.1" Setup Problem !!
by Freeman 23 Jun '05

23 Jun '05

Hi, I have some problem when i tried to setup "ser-0.9.0.GettingStarted.1.1.tar.gz" in Redhat 9, i get some error detail as below : how can i setup in RedHat 9 ? ========== make_and_install_output.txt ========= Makefile.rules:81: action.d: file or folder not found. Makefile.rules:81: crc.d: file or folder not found. Makefile.rules:81: daemonize.d: file or folder not found. Makefile.rules:81: data_lump.d: file or folder not found. Makefile.rules:81: data_lump_rpl.d: file or folder not found. Makefile.rules:81: dprint.d: file or folder not found. Makefile.rules:81: dset.d: file or folder not found. Makefile.rules:81: error.d: file or folder not found. Makefile.rules:81: fifo_server.d: file or folder not found. Makefile.rules:81: flags.d: file or folder not found. Makefile.rules:81: forward.d: file or folder not found. Makefile.rules:81: hash_func.d: file or folder not found. Makefile.rules:81: ip_addr.d: file or folder not found. Makefile.rules:81: main.d: file or folder not found. Makefile.rules:81: md5.d: file or folder not found. Makefile.rules:81: md5utils.d: file or folder not found. Makefile.rules:81: modparam.d: file or folder not found. Makefile.rules:81: msg_translator.d: file or folder not found. Makefile.rules:81: pass_fd.d: file or folder not found. Makefile.rules:81: proxy.d: file or folder not found. Makefile.rules:81: qvalue.d: file or folder not found. Makefile.rules:81: re.d: file or folder not found. Makefile.rules:81: receive.d: file or folder not found. Makefile.rules:81: resolve.d: file or folder not found. Makefile.rules:81: route.d: file or folder not found. Makefile.rules:81: route_struct.d: file or folder not found. Makefile.rules:81: script_cb.d: file or folder not found. Makefile.rules:81: socket_info.d: file or folder not found. Makefile.rules:81: sr_module.d: file or folder not found. Makefile.rules:81: stats.d: file or folder not found. Makefile.rules:81: tcp_main.d: file or folder not found. Makefile.rules:81: tcp_read.d: file or folder not found. Makefile.rules:81: timer.d: file or folder not found. Makefile.rules:81: tsend.d: file or folder not found. Makefile.rules:81: udp_server.d: file or folder not found. Makefile.rules:81: unixsock_server.d: file or folder not found. Makefile.rules:81: usr_avp.d: file or folder not found. Makefile.rules:81: mem/f_malloc.d: file or folder not found. Makefile.rules:81: mem/mem.d: file or folder not found. Makefile.rules:81: mem/memtest.d: file or folder not found. Makefile.rules:81: mem/q_malloc.d: file or folder not found. Makefile.rules:81: mem/shm_mem.d: file or folder not found. Makefile.rules:81: mem/vq_malloc.d: file or folder not found. Makefile.rules:81: parser/hf.d: file or folder not found. Makefile.rules:81: parser/msg_parser.d: file or folder not found. Makefile.rules:81: parser/parse_allow.d: file or folder not found. Makefile.rules:81: parser/parse_content.d: file or folder not found. Makefile.rules:81: parser/parse_cseq.d: file or folder not found. Makefile.rules:81: parser/parse_disposition.d: file or folder not found. Makefile.rules:81: parser/parse_diversion.d: file or folder not found. Makefile.rules:81: parser/parse_event.d: file or folder not found. Makefile.rules:81: parser/parse_expires.d: file or folder not found. Makefile.rules:81: parser/parse_fline.d: file or folder not found. Makefile.rules:81: parser/parse_from.d: file or folder not found. Makefile.rules:81: parser/parse_hname2.d: file or folder not found. Makefile.rules:81: parser/parse_hostport.d: file or folder not found. Makefile.rules:81: parser/parse_methods.d: file or folder not found. Makefile.rules:81: parser/parse_nameaddr.d: file or folder not found. Makefile.rules:81: parser/parse_param.d: file or folder not found. Makefile.rules:81: parser/parser_f.d: file or folder not found. Makefile.rules:81: parser/parse_rpid.d: file or folder not found. Makefile.rules:81: parser/parse_rr.d: file or folder not found. Makefile.rules:81: parser/parse_sipifmatch.d: file or folder not found. Makefile.rules:81: parser/parse_to.d: file or folder not found. Makefile.rules:81: parser/parse_uri.d: file or folder not found. Makefile.rules:81: parser/parse_via.d: file or folder not found. Makefile.rules:81: parser/digest/digest.d: file or folder not found. Makefile.rules:81: parser/digest/digest_parser.d: file or folder not found. Makefile.rules:81: parser/digest/param_parser.d: file or folder not found. Makefile.rules:81: parser/contact/contact.d: file or folder not found. Makefile.rules:81: parser/contact/parse_contact.d: file or folder not found. Makefile.rules:81: db/db.d: file or folder not found. Makefile.rules:81: db/db_fifo.d: file or folder not found. Makefile.rules:81: lex.yy.d: file or folder not found. Makefile.rules:81: cfg.tab.d: file or folder not found. bison -d -b cfg cfg.y cfg.y contains 1 shift/reduce conflict. flex cfg.lex gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c action.c -o action.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c crc.c -o crc.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c daemonize.c -o daemonize.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c data_lump.c -o data_lump.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c data_lump_rpl.c -o data_lump_rpl.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c dprint.c -o dprint.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c dset.c -o dset.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c error.c -o error.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c fifo_server.c -o fifo_server.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c flags.c -o flags.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c forward.c -o forward.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c hash_func.c -o hash_func.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c ip_addr.c -o ip_addr.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c main.c -o main.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c md5.c -o md5.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c md5utils.c -o md5utils.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c modparam.c -o modparam.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c msg_translator.c -o msg_translator.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c pass_fd.c -o pass_fd.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c proxy.c -o proxy.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c qvalue.c -o qvalue.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c re.c -o re.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c receive.c -o receive.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c resolve.c -o resolve.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c route.c -o route.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c route_struct.c -o route_struct.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c script_cb.c -o script_cb.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c socket_info.c -o socket_info.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c sr_module.c -o sr_module.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c stats.c -o stats.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c tcp_main.c -o tcp_main.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c tcp_read.c -o tcp_read.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c timer.c -o timer.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c tsend.c -o tsend.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c udp_server.c -o udp_server.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c unixsock_server.c -o unixsock_server.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c usr_avp.c -o usr_avp.o In file included from usr_avp.c:45: mem/shm_mem.h: In function `_shm_malloc': mem/shm_mem.h:135: too many arguments to function `fm_malloc' mem/shm_mem.h: In function `_shm_realloc': mem/shm_mem.h:146: too many arguments to function `fm_realloc' usr_avp.c: In function `destroy_avp': usr_avp.c:313: too many arguments to function `fm_free' usr_avp.c: In function `destroy_avp_list_unsafe': usr_avp.c:328: too many arguments to function `fm_free' usr_avp.c: In function `destroy_avp_list': usr_avp.c:343: too many arguments to function `fm_free' usr_avp.c: In function `add_avp_galias': usr_avp.c:420: too many arguments to function `fm_malloc' usr_avp.c:426: too many arguments to function `fm_malloc' usr_avp.c:437: too many arguments to function `fm_malloc' usr_avp.c:459: too many arguments to function `fm_free' usr_avp.c:461: too many arguments to function `fm_free' make: *** [usr_avp.o] Error 1 ................... ========== make_and_install_output.txt ========= Tks a Lot ! Freemane 有即時通知，收艂

1 0

[Serusers] Weird register issue
by Anthony Dean 23 Jun '05

23 Jun '05

Weird issue, only with registers. Sometimes when I start ser (and openser, same issue with both) my test phone (behind nat) will not register allways getting 401 unauthorized back. But wait a minute and it will register (without resetting ser or the phone). Is this a built in feature or a bug (it only happens when I reset ser/openser after 60 seconds the server and phone behave normally). Got me stumped as the configs on the phone and ser don't change. REGISTER sip:XX.XX.XX.XX SIP/2.0. Via: SIP/2.0/UDP 192.168.10.25:5060. From: <sip:2959770500@XX.XX.XX.XX;user=phone>;tag=1886287424. To: <sip:2959770500@XX.XX.XX.XX;user=phone>. Call-ID: 147690828(a)192.168.10.25. CSeq: 1 REGISTER. Contact: Nathan Gross <sip:2959770500@192.168.10.25:5060;user=phone;transport=udp>;expires=60. User-Agent: Cisco-CP7905/1.02-040406A. Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER. Content-Length: 0. SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.10.25:5060;rport=62686;received=YY.YY.YY.YY. From: <sip:2959770500@XX.XX.XX.XX;user=phone>;tag=1886287424. To: <sip:2959770500@XX.XX.XX.XX;user=phone>;tag=264d52ef5b2f408fc315ea210b15c82e .5681. Call-ID: 147690828(a)192.168.10.25. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="XX.XX.XX.XX", nonce="42b9e43447292eaaa08c9b86dcc1e23344ac3c6a", qop="auth". Content-Length: 0. Thank You Anthony Dean Network Design & Support Tristar Communications This correspondence is considered confidential and any reproduction for the purpose of public disclosure is forbidden without written permission by the author signed above. If you are not the intended recipient, please immediately notify the sender and delete any copies. If you are responding to this e-mail, please append or attach all previous correspondence as a point of reference.

2 1

[Serusers] desc_time_order not found in registrar
by TCCHAN 23 Jun '05

23 Jun '05

Dear All, I am using ser 0.9.0 and found that the line modparam("registrar","desc_time_order",1) gives a error: set_mod_param_regex: parameter <desc_time_order> not found in module <registrar>. I look at the documentation, there is such a param is registrar. What's happening here? Thank for any help available. Regards, TC Chan

2 1

[Serusers] Load balancing with Mediaproxy
by peiyin 23 Jun '05

23 Jun '05

Hi everyone, I'm currently stuck at load balancing 2 ser with mediaproxy. I've setup the 2 DNS record in my master db.abc with ............ ser1.abc.net.my. IN A yy.yy.yy ser2.abc.net.my. IN A zz.zz.zz _mediaproxy._tcp.abc.net.my. ......... ser1.abc.net.my. _mediaproxy._tcp.abc.net.my. ......... ser2.abc.net.my. Both domains are pingable. What should be my next step? my present sip phones are all pointed directly to ser1.xx.xx.xx and ser1 is connected to remote db A. Do i still need to setup a virtual IP with heartbeat? What will be my sip setting at my sip phone? Also, what will be in my /etc/profile (export SIP_DOMAIN=??) Thanks in advance. peiyin

3 7

[Serusers] SER with Radius Authentication
by Rafael J. Risco G.V. 23 Jun '05

23 Jun '05

Hi, I´ve configured freeradius and SER according to the Radius HOW TO document, Accounting works very well but now I am doing some tests trying to do user authentication however all the authentication requests coming to the freeradius fails and X-lite sipphone is receiving an Unauthorized message from SER, please some advice, thanks rafael PS: config files... in /usr/local/etc/raddb/users : --------- test Auth-Type := Digest, User-Password == "test" Reply-Message = "Hello, test with digest" 6609876 Auth-Type := Digest User-Password := "9876", Digest-Response = "lalalalala", Reply-Message = "Hello, ibm1" 6604321 Auth-Type := Digest User-Password := "4321", Digest-Response = "lalalalala", Reply-Message = "Hello, ibm2" --------- Some relevant data in ser.cfg: ... modparam("group_radius", "use_domain", 0) .... if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "1"); break; }; if (!save("location")) { sl_reply_error(); }; break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc if (radius_is_user_in("Request-URI", "voicemail")) { log(1, "requested user is in voicemail group"); setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # sl_send_reply("404", "Not Found"); log(1,"unable to locate user"); route(4); break; }; }; # End of "if(uri==myself)" .... ------------------RADIUSD -X Output ---------------------------: rad_recv: Access-Request packet from host 127.0.0.1:33187, id=79, length=311 User-Name = "6604321(a)10.0.1.22" Digest-Attributes = 0x0a0936363034333231 Digest-Attributes = 0x010b31302e302e312e3232 Digest-Attributes = 0x022a34323366333163373062336631643261643330383833633238303434316632663133643136613830 Digest-Attributes = 0x040f7369703a31302e302e312e3232 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303162 Digest-Attributes = 0x08224433343132424232394131453131443939334232303035304241373836433642 Digest-Response = "a6a7812ac0331324f977453c228da2ed" Service-Type = IAPP-Register Sip-URI-User = "6604321" Cisco-AVPair = "call-id=D3412ADB9A1E11D993B20050BA786C6B(a)10.0.1.22" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 8 modcall[authorize]: module "preprocess" returns ok for request 8 modcall[authorize]: module "chap" returns noop for request 8 modcall[authorize]: module "mschap" returns noop for request 8 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "6604321" Digest-Realm = "10.0.1.22" Digest-Nonce = "423f31c70b3f1d2ad30883c280441f2f13d16a80" Digest-URI = "sip:10.0.1.22" Digest-Method = "REGISTER" Digest-QOP = "auth" Digest-Nonce-Count = "0000001b" Digest-CNonce = "D3412BB29A1E11D993B20050BA786C6B" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 8 rlm_realm: Looking up realm "10.0.1.22" for User-Name = "6604321(a)10.0.1.22" rlm_realm: No such realm "10.0.1.22" modcall[authorize]: module "suffix" returns noop for request 8 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 8 users: Matched DEFAULT at 152 modcall[authorize]: module "files" returns ok for request 8 modcall: group authorize returns ok for request 8 rad_check_password: Found Auth-Type DIGEST auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 8 rlm_digest: Configuration item "User-Password" is required for authentication. modcall[authenticate]: module "digest" returns invalid for request 8 modcall: group authenticate returns invalid for request 8 auth: Failed to validate the user. Delaying request 8 for 1 seconds Finished request 8 Going to the next request --- Walking the entire request list --- Waking up in 1 seconds... rad_recv: Access-Request packet from host 127.0.0.1:33188, id=80, length=311 User-Name = "6609876(a)10.0.1.22" Digest-Attributes = 0x0a0936363039383736 Digest-Attributes = 0x010b31302e302e312e3232 Digest-Attributes = 0x022a34323366333163373062336631643261643330383833633238303434316632663133643136613830 Digest-Attributes = 0x040f7369703a31302e302e312e3232 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303163 Digest-Attributes = 0x08224433343132424235394131453131443939334232303035304241373836433642 Digest-Response = "50fa695654b20e2eec54a1003fe15d9f" Service-Type = IAPP-Register Sip-URI-User = "6609876" Cisco-AVPair = "call-id=D3412ADE9A1E11D993B20050BA786C6B(a)10.0.1.22" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 9 modcall[authorize]: module "preprocess" returns ok for request 9 modcall[authorize]: module "chap" returns noop for request 9 modcall[authorize]: module "mschap" returns noop for request 9 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "6609876" Digest-Realm = "10.0.1.22" Digest-Nonce = "423f31c70b3f1d2ad30883c280441f2f13d16a80" Digest-URI = "sip:10.0.1.22" Digest-Method = "REGISTER" Digest-QOP = "auth" Digest-Nonce-Count = "0000001c" Digest-CNonce = "D3412BB59A1E11D993B20050BA786C6B" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 9 rlm_realm: Looking up realm "10.0.1.22" for User-Name = "6609876(a)10.0.1.22" rlm_realm: No such realm "10.0.1.22" modcall[authorize]: module "suffix" returns noop for request 9 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 9 users: Matched DEFAULT at 152 modcall[authorize]: module "files" returns ok for request 9 modcall: group authorize returns ok for request 9 rad_check_password: Found Auth-Type DIGEST auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 9 rlm_digest: Configuration item "User-Password" is required for authentication. modcall[authenticate]: module "digest" returns invalid for request 9 modcall: group authenticate returns invalid for request 9 auth: Failed to validate the user. Delaying request 9 for 1 seconds Finished request 9 Going to the next request --- Walking the entire request list --- Waking up in 1 seconds... --- Walking the entire request list --- Sending Access-Reject of id 79 to 127.0.0.1:33187 Waking up in 1 seconds... rad_recv: Access-Request packet from host 127.0.0.1:33189, id=81, length=311 User-Name = "6609876(a)10.0.1.22" Digest-Attributes = 0x0a0936363039383736 Digest-Attributes = 0x010b31302e302e312e3232 Digest-Attributes = 0x022a34323366333163373062336631643261643330383833633238303434316632663133643136613830 Digest-Attributes = 0x040f7369703a31302e302e312e3232 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303163 Digest-Attributes = 0x08224433343132424236394131453131443939334232303035304241373836433642 Digest-Response = "e4f68760f2b3eed0ad45942b32542c92" Service-Type = IAPP-Register Sip-URI-User = "6609876" Cisco-AVPair = "call-id=D3412ADE9A1E11D993B20050BA786C6B(a)10.0.1.22" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 10 modcall[authorize]: module "preprocess" returns ok for request 10 modcall[authorize]: module "chap" returns noop for request 10 modcall[authorize]: module "mschap" returns noop for request 10 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "6609876" Digest-Realm = "10.0.1.22" Digest-Nonce = "423f31c70b3f1d2ad30883c280441f2f13d16a80" Digest-URI = "sip:10.0.1.22" Digest-Method = "REGISTER" Digest-QOP = "auth" Digest-Nonce-Count = "0000001c" Digest-CNonce = "D3412BB69A1E11D993B20050BA786C6B" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 10 rlm_realm: Looking up realm "10.0.1.22" for User-Name = "6609876(a)10.0.1.22" rlm_realm: No such realm "10.0.1.22" modcall[authorize]: module "suffix" returns noop for request 10 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 10 users: Matched DEFAULT at 152 modcall[authorize]: module "files" returns ok for request 10 modcall: group authorize returns ok for request 10 rad_check_password: Found Auth-Type DIGEST auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 10 rlm_digest: Configuration item "User-Password" is required for authentication. modcall[authenticate]: module "digest" returns invalid for request 10 modcall: group authenticate returns invalid for request 10 auth: Failed to validate the user. Delaying request 10 for 1 seconds Finished request 10 Going to the next request Sending Access-Reject of id 80 to 127.0.0.1:33188 Waking up in 1 seconds... --- Walking the entire request list --- Waking up in 1 seconds... --- Walking the entire request list --- Sending Access-Reject of id 81 to 127.0.0.1:33189 Waking up in 2 seconds... --- Walking the entire request list --- Cleaning up request 8 ID 79 with timestamp 423f309b Waking up in 1 seconds... --- Walking the entire request list --- Cleaning up request 9 ID 80 with timestamp 423f309c Waking up in 1 seconds... --- Walking the entire request list --- Cleaning up request 10 ID 81 with timestamp 423f309d Nothing to do. Sleeping until we see a request. -- rrgv

4 5

[Serusers] SER Server behind NAT and Record Route
by Matt morris 23 Jun '05

23 Jun '05

Hello List, Here's a problem I am currently having. My SER server is behind a router and its ip is an internal ip address. Hence, SER has to listen on its internal ip (would it be possible to configure it to listen on the WAN ip?). The call establishment is fine but for some reason when the user agent (I have tried XLite and another hw ATA) hangs up it will send the BYE to the server's interal ip instead of the its WAN ip. Thus the server never receives the BYE and the other side will still be connected. I am guessing it is because the Record Route header has the internal ip instead of the WAN one. How could I rewrite the ip in the record router header to the server's WAN ip? Thank you all in advance _________________________________________________________________ Scan and help eliminate destructive viruses from your inbound and outbound e-mail and attachments. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

2 1

[Serusers] RTPProxy Problem !
by Freeman 23 Jun '05

23 Jun '05

Hi, I installed "Ser-0.8.14" + "nathelper" + "rtpproxy" in RedHat 9.0, and add "./rtpproxy" in startup script, when i tried both sipsoftphone in the internet behind NAT that can connect but "no voice" and auto hang-up, but when both sipsoftphone in local network that no problem ! Which port range i need release for rtpproxy in the firewall ? my ser.cfg config as below : # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no */ #log_stderror=yes #debug=4 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/dbtext.so" #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" #loadmodule "/usr/local/lib/ser/modules/mysql.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) #modparam("auth_db", "db_url", "db:/var/dbtext") #modparam("auth_db", "user_column", "user") #modparam("auth_db", "domain_column", "domain") #modparam("auth_db", "password_column", "password") #modparam("auth_db", "calculate_ha1", 1) #modparam("auth_db", "password_column_2", "ha1_2") modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock","/var/run/rtpproxy.sock") # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic #define NAT_UAC_TEST_C_1918 0x01 /* * test for occurences of RFC1918 addresses in Contact * header field */ #define NAT_UAC_TEST_RCVD 0x02 /* * test if source address of signaling is different from * address advertised in Via */ #define NAT_UAC_TEST_V_1918 0x04 /* * test for occurences of RFC1918 addresses in SDP body */ #define NAT_UAC_TEST_S_1918 0x08 /* * test for occurences of RFC1918 addresses top Via */ route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (nat_uac_test("3")) { append_hf("Alex-hint: NAThelper\r\n"); fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "REGISTER" || ! search("^Record-Route:")) { #if (www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "1"); # }; if (method == "INVITE") { append_hf("Alex-hint: SDP rewritten\r\n"); fix_nated_sdp("3"); # Add direction=active to SDP }; log("LOG: Someone trying to register from private IP, rewriting\n"); force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (isflagset(6)) { force_rtp_proxy(); append_hf("NAT: ...\r\n"); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? append_hf("NAT: tes\r\n"); if (isflagset(6) && status == "(200)") { fix_nated_contact(); fix_nated_sdp("3"); force_rtp_proxy(); append_hf("NAT: test\r\n"); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); fix_nated_sdp("3"); force_rtp_proxy(); append_hf("NAT: test1\r\n"); }; } Please help ! Tks a Lot ! Freeman 伈鴃立即知

3 9

[Serusers] Problesm with Vovida and Mediaproxy
by Alberto Cruz 22 Jun '05

22 Jun '05

Hi guys: Does anybody knows if there is a problem using SER 0.9.2, Mediaproxy 1.3.1 and Vovida B2BUA Vocal 1.5.0? I'm having no audio when I try to place a call even if the call is established. I'm attaching my ser.cfg and a ngrep debug for reference The only weird thing that I can notice is that the B2BUA is using the localhost IP as the Via Address even when the invites was sent to the Public IP address. Did I make somtehing wrong at the ser.cfg file? Regards Alberto Cruz # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -ddddd) fork=yes log_stderror=no /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes # (cmd line: -E) debug=7 */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=200.94.75.6 port=5060 alias=200.94.75.6 alias=sipsrv.conecta-te.com fifo_mode=0666 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri_radius.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/domain.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("domain", "db_mode", 1) modparam("usrloc|domain", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("group", "use_domain", 0) # -- auth params -- # Uncomment if you are using auth module #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) #modparam("auth_db", "password_column", "password") ### Radius Configuration ### #modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") #modparam("auth_radius", "service_type", 15) #modparam("group_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") #modparam("group_radius", "use_domain", 0) # -- nathelper params -- modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) # -- mediprox params -- modparam("mediaproxy", "natping_interval", 30) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-clients") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) # ------------------------- request routing logic ------------------- # main routing logic route{ #---------------------------------------------- # initial sanity checks -- messages with # max_forwards==0, or excessively long requests #---------------------------------------------- if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483","Too Many Hops"); }; break; }; if (msg:len > max_len ) { if (method!="ACK") { sl_send_reply("513", "Message too big"); }; break; }; #---------------------------------------------------- # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol #---------------------------------------------------- if (method=="INVITE" && client_nat_test("3")){ record_route_preset("200.94.75.6:5060;nat=yes"); }else if (!method=="REGISTER"){ record_route(); }; #--------------------------------------------- # we tear down the call before entering to loose route #--------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); append_hf("P-hint: finishing call\r\n"); }; #------------------------------------------------- # subsequent messages withing a dialog should take the # path determined by record-routing #------------------------------------------------- if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); # The following lines are added due media proxy if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") && !search("^Route:.*;nat=yes")) { # Mark as NAT'ed setflag(6); use_media_proxy(); }; }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here route(1); break; }; #---------------------------------------- # processing the most common sip messages # for all the calls #---------------------------------------- if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(5); route(1); break; }; if (uri==myself) { #--------------------------------- #all the calls from the sip uacs should #be handled by the b2bua for authorization #and accounting #--------------------------------- if (!(src_ip==200.94.75.6 || src_ip==200.94.75.1)and(method=="INVITE" || method=="BYE" || method=="CANCEL")) { log("Hey this request is NOT from B2BUA\n"); log("Forwarding to B2BUA\n"); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; forward(200.94.75.6,5065); break; }; if (method=="ACK") { route(6); break; } else if (method=="CANCEL" || method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; route(4); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { #------------------------------- # Default SIP message handler # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP #------------------------------ t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); append_hf("P-hint: finishing call no-relay\r\n"); }; sl_reply_error(); }; } route[2] { #--------------------------- # "REGISTER" message handler #--------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (is_from_local() || uri==myself) { if (!radius_www_authorize("")) { www_challenge("", "0"); break; } else if (!check_to()){ sl_send_reply("403", "Username!=To not allowed"); break; }; if (!save("location")) { sl_reply_error(); }; } else { append_hf("P-hint: outbound alias\r\n"); sl_send_reply("403", "This domain is not served here"); }; } route[3] { #-------------------------- #"CANCEL" and "INVITE" message handler #-------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE") { if (!(is_from_local() || uri==myself || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; } else if (method=="CANCEL") { route(1); break; }; route(4); if (!lookup("location")) { if ((uri=~"sip:01[1-9][0-9]+@.*") || (uri=~"sip:00[1-9][0-9]+@.*")) { rewritehost("200.94.75.1"); }; } else { sl_send_reply("404", "User not found"); break; }; route(5); route(1); } route[4] { #--------------- #we need to check for the alias #--------------- lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; } route[5] { #------------------------------------- #RTP Proxy handler #------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); append_hf("P-hint: are you nated?\r\n"); }; } route[6] { #-------------------------- #ACK handler #-------------------------- route(4); lookup("location"); route(1); } failure_route[1] { end_media_session(); append_hf("P-hint: finishing call on failure\r\n"); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|(2[0-9][0-9])")) { if (client_nat_test("1")) { fix_nated_contact(); }; if (!search("^Content-Length:\ +0")) { use_media_proxy(); append_hf("P-hint: use nat for call\r\n"); }; }; } interface: any filter: ip and ( port 5060 ) match: sip ##### U 2005/06/22 03:33:01.181845 200.67.33.247:61973 -> 200.94.75.6:5060 INVITE sip:018183324166@sipsrv.conecta-te.com SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-a8bb0d6a. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 101 INVITE. Max-Forwards: 70. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 314. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 2188429 2188429 IN IP4 172.31.253.127. s=-. c=IN IP4 172.31.253.127. t=0 0. m=audio 16470 RTP/AVP 18 0 4 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/06/22 03:33:01.184060 200.94.75.6:5060 -> 200.94.75.6:5065 INVITE sip:018183324166@sipsrv.conecta-te.com SIP/2.0. Record-Route: <sip:018183324166@200.94.75.6:5060;nat=yes;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=0. Via: SIP/2.0/UDP 172.31.253.127:5060;rport=61973;received=200.67.33.247;branch=z9hG4bK-a8bb0d6a. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 101 INVITE. Max-Forwards: 16. Contact: 8183651723 <sip:8183651723@200.67.33.247:61973>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 314. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 2188429 2188429 IN IP4 172.31.253.127. s=-. c=IN IP4 172.31.253.127. t=0 0. m=audio 16470 RTP/AVP 18 0 4 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/06/22 03:33:01.195680 200.94.75.6:5065 -> 200.94.75.6:5060 INVITE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1:5065. To: <sip:018183324166@sipsrv.conecta-te.com>. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 101 INVITE. Max-Forwards: 16. Expires: 240. Contact: <sip:8183651723@127.0.0.1:5065>. Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER. Supported: x-sipura. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Type: application/sdp. Content-Length: 314. . v=0. o=- 2188429 2188429 IN IP4 172.31.253.127. s=-. c=IN IP4 172.31.253.127. t=0 0. m=audio 16470 RTP/AVP 18 0 4 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=fmtp:101 0-15. a=ptime:30. # U 2005/06/22 03:33:01.201069 200.94.75.6:5060 -> 200.94.75.6:5065 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. To: <sip:018183324166@sipsrv.conecta-te.com>. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 101 INVITE. Server: Sip EXpress router (0.9.2 (i386/linux)). Content-Length: 0. Warning: 392 200.94.75.6:5060 "Noisy feedback tells: pid=22827 req_src_ip=200.94.75.6 req_src_port=5065 in_uri=sip:018183324166@200.94.75.6:5060 out_uri=sip:018183324166@200.94.75.1:5060 via_cnt==1". . # U 2005/06/22 03:33:01.201175 200.94.75.6:5060 -> 200.94.75.1:5060 INVITE sip:018183324166@200.94.75.1:5060 SIP/2.0. Record-Route: <sip:018183324166@200.94.75.6:5060;nat=yes;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK6512.acf311c7.0. Via: SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. To: <sip:018183324166@sipsrv.conecta-te.com>. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 101 INVITE. Max-Forwards: 15. Expires: 240. Contact: <sip:8183651723@200.94.75.6:5065>. Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER. Supported: x-sipura. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Type: application/sdp. Content-Length: 311. P-hint: are you nated?. . v=0. o=- 2188429 2188429 IN IP4 172.31.253.127. s=-. c=IN IP4 200.94.75.6. t=0 0. m=audio 35024 RTP/AVP 18 0 4 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=fmtp:101 0-15. a=ptime:30. # U 2005/06/22 03:33:01.216824 200.94.75.1:5060 -> 200.94.75.6:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK6512.acf311c7.0,SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:22 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 2005/06/22 03:33:06.105293 200.94.75.1:5060 -> 200.94.75.6:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK6512.acf311c7.0,SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:22 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 210. . v=0. o=CiscoSystemsSIP-GW-UserAgent 133 5493 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.75.1. t=0 0. m=audio 18168 RTP/AVP 18. c=IN IP4 200.94.75.1. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. # U 2005/06/22 03:33:06.108419 200.94.75.6:5060 -> 200.94.75.6:5065 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:22 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 210. P-hint: use nat for call. . v=0. o=CiscoSystemsSIP-GW-UserAgent 133 5493 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.75.6. t=0 0. m=audio 35024 RTP/AVP 18. c=IN IP4 200.94.75.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. # U 2005/06/22 03:33:11.985671 200.94.75.1:5060 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK6512.acf311c7.0,SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:22 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:018183324166@200.94.75.1:5060>. Record-Route: <sip:018183324166@200.94.75.6:5060;nat=yes;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Type: application/sdp. Content-Length: 210. . v=0. o=CiscoSystemsSIP-GW-UserAgent 133 5493 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.75.1. t=0 0. m=audio 18168 RTP/AVP 18. c=IN IP4 200.94.75.1. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. # U 2005/06/22 03:33:11.988704 200.94.75.6:5060 -> 200.94.75.6:5065 SIP/2.0 200 OK. Via: SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:22 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:018183324166@200.94.75.1:5060>. Record-Route: <sip:018183324166@200.94.75.6:5060;nat=yes;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Type: application/sdp. Content-Length: 210. P-hint: use nat for call. . v=0. o=CiscoSystemsSIP-GW-UserAgent 133 5493 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.75.6. t=0 0. m=audio 35024 RTP/AVP 18. c=IN IP4 200.94.75.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. # U 2005/06/22 03:33:12.082614 200.67.33.247:61973 -> 200.94.75.6:5060 ACK sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-c5ebdced. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 101 ACK. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:12.085346 200.94.75.6:5060 -> 127.0.0.1:5065 ACK sip:018183324166@127.0.0.1:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-c5ebdced. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 101 ACK. Max-Forwards: 16. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: rr-enforced. . # U 2005/06/22 03:33:12.087437 200.94.75.6:5065 -> 200.94.75.6:5060 ACK sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1:5065. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 101 ACK. Max-Forwards: 70. Route: <sip:018183324166@200.94.75.1:5060>. Content-Length: 0. . # U 2005/06/22 03:33:12.090626 200.94.75.6:5060 -> 200.94.75.1:5060 ACK sip:018183324166@200.94.75.1:5060 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=0. Via: SIP/2.0/UDP 127.0.0.1:5065;received=200.94.75.6. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 101 ACK. Max-Forwards: 16. Content-Length: 0. P-hint: rr-enforced. . # U 2005/06/22 03:33:20.813952 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:20.816802 200.94.75.6:5060 -> 127.0.0.1:5065 BYE sip:018183324166@127.0.0.1:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK47f1.7c391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 16. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: finishing call. P-hint: rr-enforced. . # U 2005/06/22 03:33:20.818924 200.94.75.6:5065 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6:5060;branch=z9hG4bK47f1.7c391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:20.820445 200.94.75.6:5065 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060;lr=on SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1:5065. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@200.94.75.1:5060>. Content-Length: 0. . # U 2005/06/22 03:33:20.823726 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:20.826479 200.94.75.6:5060 -> 200.94.75.1:5060 BYE sip:018183324166@200.94.75.1:5060 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK3512.5c7c5523.0. Via: SIP/2.0/UDP 127.0.0.1:5065;received=200.94.75.6. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 102 BYE. Max-Forwards: 16. Content-Length: 0. P-hint: finishing call. P-hint: rr-enforced. . # U 2005/06/22 03:33:20.833064 200.94.75.1:5060 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK3512.5c7c5523.0,SIP/2.0/UDP 127.0.0.1:5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:41 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. Content-Length: 0. CSeq: 102 BYE. . # U 2005/06/22 03:33:20.834554 200.94.75.6:5060 -> 200.94.75.6:5065 SIP/2.0 200 OK. Via: SIP/2.0/UDP 127.0.0.1:5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:41 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. Content-Length: 0. CSeq: 102 BYE. . # U 2005/06/22 03:33:21.314404 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:21.317081 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:22.312811 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:22.315462 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:24.315246 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:24.317882 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . ##### U 2005/06/22 03:33:28.313369 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:28.316301 200.94.75.6:5060 -> 127.0.0.1:5065 BYE sip:018183324166@127.0.0.1:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK47f1.8c391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 16. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: finishing call. P-hint: rr-enforced. . # U 2005/06/22 03:33:28.317393 200.94.75.6:5065 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6:5060;branch=z9hG4bK47f1.8c391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:28.320084 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:32.313366 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:32.316045 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:36.311492 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:36.314360 200.94.75.6:5060 -> 127.0.0.1:5065 BYE sip:018183324166@127.0.0.1:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK47f1.9c391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 16. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: finishing call. P-hint: rr-enforced. . # U 2005/06/22 03:33:36.315488 200.94.75.6:5065 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6:5060;branch=z9hG4bK47f1.9c391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:36.318179 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:40.312612 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:40.315279 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:44.312361 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:44.315262 200.94.75.6:5060 -> 127.0.0.1:5065 BYE sip:018183324166@127.0.0.1:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK47f1.ac391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 16. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: finishing call. P-hint: rr-enforced. . # U 2005/06/22 03:33:44.316366 200.94.75.6:5065 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6:5060;branch=z9hG4bK47f1.ac391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:44.319075 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:48.312358 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:48.315043 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:52.312107 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:52.315065 200.94.75.6:5060 -> 127.0.0.1:5065 BYE sip:018183324166@127.0.0.1:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK47f1.bc391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 16. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: finishing call. P-hint: rr-enforced. . # U 2005/06/22 03:33:52.316179 200.94.75.6:5065 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6:5060;branch=z9hG4bK47f1.bc391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:52.318886 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . ############exit 89 received, 0 dropped

1 0

[Serusers] Re: [Sems] sems module
by harry gaillac 22 Jun '05

22 Jun '05

Stefan, May be you read my emails about asterisk+ser+mwi. i sent questions to find a sip media server for ser but asterisk don't seem to be the right way for ser (just pstn gateway). how many devoloppers work on sems project ? I don't understand ivr plugin this one can provide IVR UMS/mwi and more but sems conf file provide one file *.py for this module !? Based on IVR plugin is it possible to write some dedicated modules : UMS+mwi module MOH module IVR module Harry --- Stefan Sayer <sayer(a)cs.tu-berlin.de> a écrit : > harry gaillac wrote: > > hello, > > > > > > If i add this code will it play music as in xmms > > playlist for example. > > > > what i expect from moh module is to look in the > moh > > directory for one or more wav file (-this > directory is > > defined in sems.cfg). > > > > if a least one file is found moh play the same > file > > (boucle) . > > > > If more than one file is found all playlist is > palyed > > (boucle) > > > > i expect from moh module it provide something like > > xmms > > if you want to listen the same music. > > > > I agree i know what the moh should do but > translate my > > > > need into C++ is a problem for me. > it is way easier to write a short python script: > import ivr > import os > > ... > > def pushFiles(): > print "pushing files in the media queue..." > for file in os.listdir("/path/to/mymusic"): > fullname = os.path.join(directory, > file) > print "enqueueing " + fullname > ivr.enqueueMediaFile(fullname, false) > > ... > > This could of course be optimized; it will open all > files when the media > queue is empty...just to get you started. > > see e.g. http://effbot.org/librarybook/os-path.htm > or > http://docs.python.org/lib/os-file-dir.html > > Stefan > > > > > > Thanks for help > > > > Harry > > > > > > --- Stefan Sayer <sayer(a)cs.tu-berlin.de> a écrit : > > > > > >>pardon: > >>Stefan Sayer wrote: > >> > >>>and some python documentation ( > >> > >>www.python.org/docs ). > >>www.python.org/doc/ > >> > >>>... > >> > >>>If you prefer c++ you can simply extend the > >> > >>announcement module > >> > >>>// somewhere in onSessionStart > >>>while (!getSession()->sess_stopped.get()) { > >>> getSession()->rtp_str.play("file1.wav"); > >>> getSession()->rtp_str.play("file2.wav"); > >>> getSession()->rtp_str.play("file3.wav"); > >>>} > >> > >>vector<string> files; > >>files.push_back("file1.wav"); > >>files.push_back("file2.wav"); > >>files.push_back("file3.wav"); > >> > >>vector<string>::iterator it = files.begin(); > >>while (!getSession()->sess_stopped.get()) { > >> getSession()->rtp_str.play(*it); > >> it++; > >> if (it == files.end()) > >> it = files.begin(); > >>} > >> > >> > >>>Stefan > >>> > >>> > >>> > >>>_______________________________________________ > >> > >>Sems mailing list > >>sems(a)lists.iptel.org > >>http://lists.iptel.org/cgi-bin/mailman/listinfo/sems > >> > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 1

[Serusers] 5060 Firewalled
by Daniel Poulsen 22 Jun '05

22 Jun '05

Hello- I have a client in a building where the IT staff refuse to open port 5060 for him citing 'security concerns'. Is there a way to get SER to listen on multiple ports? or do I have to start another instance of SER with a different ser.cfg? Thanks for your input. Dan

2 1

RE: [Serusers] 5060 Firewalled
by Giudice, Salvatore 22 Jun '05

22 Jun '05

Where is this SER going to be deployed? at the customer location or in your rack? If you wanted the same installation to listen on two different ports, you could install a second instance or configure port forwarding. All this is sort of useless though and creates something else to support. Maybe they will be willing to buy an SBC like an ingate or a jasomi and deploy it at the customer location instead of opening the ominous scary port 5060... Either way, good luck. _____ From: Daniel Poulsen [mailto:dpoulsen@gmail.com] Sent: Wednesday, June 22, 2005 12:20 PM To: serusers(a)lists.iptel.org Subject: [Serusers] 5060 Firewalled Hello- I have a client in a building where the IT staff refuse to open port 5060 for him citing 'security concerns'. Is there a way to get SER to listen on multiple ports? or do I have to start another instance of SER with a different ser.cfg? Thanks for your input. Dan

1 0

RE: [Serusers] pa cannot work as a B2BUA?
by Simith Nambiar 22 Jun '05

22 Jun '05

-----Original Message----- From: Jamey Hicks [mailto:jamey.hicks@hp.com] Sent: Monday, June 13, 2005 11:23 PM To: mma Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] pa cannot work as a B2BUA? mma wrote: > hi, > > i read in pa module that it cannot subsribe to presence of users > using SIP, > so how can i do to manage the presence of my contact list > PA still does not have support to SUBSCRIBE to contacts. What you can do is set up PA so that it only handles presence for users who register against and publish presence information to your SER server, and to forward subscribe requests belonging to other registrars or UAs. > i added pa module to SER and a had this errors in response to a > SUBSCRIBE request > > parse_hfs(): Error while parsing headers > I would have to see the message to be able to tell what is wrong. Which UA was generating the subscription? Which version of SER? -Jamey Hi Jamey, I have a similar Error , when using MSN Messenger Ver 4.6, and Ser-0.9.0 , Please let me know if you have any hints to look for problems ! Cheers. Simith _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ______________________________________________________________________ This email has been scanned via a message filtering facility. If you have any issues with your email, please contact the IT Dept.

5 8

[serusers] pa module
by michael p 22 Jun '05

22 Jun '05

excuse me i have this errors too : 12(942) parse_hfs(): Error while parsing headers 12(942) handle_subscription(): Error while parsing message header 12(942) ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established 12(942) send_reply(): Error while sending 400 Bad Request someone can help me? thanks M.

3 3

RE: [Serusers] append_to_reply and save behaviour
by Fermín Galán Márquez 22 Jun '05

22 Jun '05

Ok. Thanks! I thought that the response was sent when the route block ends (that is, when the 'break' action is reached). Best regards, ------ Fermín Agora Systems, S. A. -----Mensaje original----- De: Samuel Osorio Calvo [mailto:samuel.osorio@nl.thalesgroup.com] Enviado el: miércoles, 22 de junio de 2005 13:15 Para: fermin.galan(a)agora-2000.com; serdev(a)lists.iptel.org; serusers(a)lists.iptel.org CC: sd-ims(a)agws.agora-2000.com Asunto: Re: [Serusers] append_to_reply and save behaviour The reason is that the function save sends the response and all further processing can not affect this response (which is already travelling back). I don't know if it can be called bug, feature, or just design... ;) SER config file executes the commands sequentially (besides route blocks) and you have to be careful with the order. Samuel. Unclassified. >>> Fermín Galán Márquez <fermin.galan(a)agora-2000.com> 06/22/05 12:47PM >>> Hello, I've found and estrange behaviour in the 'append_to_reply' action when used in conjunction with 'save' action during REGISTER processing. In particular, if I use in ser.cfg: ... append_to_reply("Service-Route: <sip:orig@scscf.domain1.com;lr>\r\n"); save("domain1"); ... Then the Service-Route header is added to the "200 OK". But if changing the relative order of the actions: ... save("domain1"); append_to_reply("Service-Route: <sip:orig@scscf.domain1.com;lr>\r\n"); ... Then the 'append_to_reply' is ignored: no header is added to the response. I think is a weird behaviour... I would like to know if this is a bug or a feature (I have read the textops documentation at http://www.iptel.org/ser/doc/modules/html/textops.html#AEN104 and nothing about problems with 'save' are mentioned). Thank you in advance! Best regards, ------ Fermín Agora Systems, S. A. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] stuck while using fork=no
by Chia Huey Lim 22 Jun '05

22 Jun '05

Hi. I am using ser 0.9.0. The moment I change from fork = yes to fork = no to enter debug mode And I attempt to "serctl restart" I was stuck in [root@localhost ser-0.9.0]# serctl start Starting SER : The moment I change back to fork = yes, I can start SER again. Any further adjustment I need to do when I switch from fork = yes to fork = no? Thanks in advance. Regards, Chia

2 1

[Serusers] SEMS with multiple SERs
by Andreas Granig 22 Jun '05

22 Jun '05

Hi all, I gave the SEMS-conference module a try and it looks quite promising. But one thing comes in my mind: We've multiple SERs balanced via DNS-SRV, so a call to a conference room can end up in any of the SEMS instances where SER is running, which results in different conference rooms for one single number. To solve this issue, the communication between SER and SEMS has to be made networking enabled (assuming a simple case where one SEMS-server serves many SERs; for multiple SEMS-Instances maybe a common DB or a dispatcher has to be used to find the correct SEMS-Server). Some possibilites come in my mind: 1. Adding a t_write_udp/tcp() function to SER's tm-module and add a UDP/TCP-Listener to SEMS. 2. Enhance the new xmlrpc-module of SER to allow sending XMLRPC-requests from inside the SER-config and add an XMLRPC-server to SEMS. Is there already anybody working on this? Or any comments on the approaches? Andy

2 2

[Serusers] Re: [Sems] SEMS crashes, fifo errors
by Llanos Serna García-Conde 22 Jun '05

22 Jun '05

1 0

[Serusers] ASTERISK+SER+MWI
by harry gaillac 22 Jun '05

22 Jun '05

Hello, I wish to use asterisk as voicemail system with ser My phones are registered on SER and asterisk provide conference ivr and voicemail system. I want to use Asterisk Realtime Architecture to store voicemessages and configuration . Michael Shuler has written a patch for mwi. http://bugs.digium.com/bug_view_page.php?bug_id=0002980 I need help to set up voicemail storage and how to send mwi to clients registered on ser Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

4 7

Re: [Serusers] SER act as a 3cpp server
by Samuel Osorio Calvo 22 Jun '05

22 Jun '05

I don't think SER can act as a 3pcc because it does not have a full SIP stack and it is not able to act as a b2bua(which is what a 3pcc mainly is). Moreover, since it is not a call statefull proxy is hardly impossible to add several functionalities for a SIP entityy to work as a 3pcc. my 2 cents, Samuel. Unclassified. >>> Iqbal <iqbal(a)gigo.co.uk> 06/22/05 02:54PM >>> I havent looked at the code below, but th MWI works if the phone is registered in asterisk, if the phone is registered in SER, then MWI in asterisk needs to send a message back via ser to the phone to make its light flash. As for the call being in asterisk, what I mean is if the rtp stream is in asterisk, then when you press keys asterisk can detect these, and based on keystrokes be able to do what you wish. Doin this in SER I dont think is possible since it works on SIP messages as opposed to keystrokes (which come as part of the RTP stream, having said that I do believe some IP phones can be set to send these outside using SIP INFO (I havent tested this)), and hence when you use hold/transfer and all the rest they should send a MESSAGE which SER can recognise, else it will just be buttons you are pressing which do little, if you have transferred the cvall to asterisk, asterisk should recognise and act on it. Iqbal harry gaillac wrote: >What do you mean "if the call is in asterisk" >what about MWI_Version_11.tar.gz > >some examples help me > >Regards >harry >--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > > > >>if the call is in asterisk, then yes you should be >>able to do this, >>since asterisk should be able to detect your >>keystrokes, and based on >>that direct you to whatever >> >>Iqbal >> >>harry gaillac wrote: >> >> >> >>>Here is my problem >>> >>>I use 2 polycom ip phones (see posted mail) >>>asterisk+ser . >>> >>>When i press hold key i wish caller listen music >>> >>> >>via >> >> >>>asterisk MOH >>>Polycom told me you need a 3pcc server >>>Does asterisk is able to act as a 3pcc server ? >>> >>>caller controller callee MOH >>> >>>Regards >>>--- Quang Minh PHAN <quang-minh.phan(a)wengo.fr> a >>> >>> >>écrit >> >> >>>: >>> >>> >>> >>> >>> >>>>You can find a shell script (examples/ctd.sh) >>>> >>>> >>which >> >> >>>>implement Click To Dial >>>>example by using SER FIFO. However, you shouldn't >>>>use SER to implement a >>>>3PCC server as it's not designed for that purpose. >>>> >>>>Minh >>>> >>>>-----Message d'origine----- >>>>De : harry gaillac [mailto:gaillacharry@yahoo.fr] >>>>Envoyé : mercredi 22 juin 2005 12:25 >>>>À : Quang Minh PHAN >>>>Objet : RE: [Serusers] SER act as a 3cpp server >>>> >>>>NO, >>>> >>>>I mean 3pcc . >>>> >>>>Harry >>>> >>>> >>>> >>>> >>>> >>>> >>http://www.iptel.org/info/players/ietf/callprocessing/3pcc/draft-ietf-sippin >> >> >>> >>> >>> >>> >>>>g-3pcc-03.txt >>>>--- Quang Minh PHAN <quang-minh.phan(a)wengo.fr> a >>>>écrit >>>>: >>>> >>>> >>>> >>>> >>>> >>>>>Hi, >>>>> >>>>> Did you mean 3GPP? In that case, the answer is >>>>> >>>>> >>>>> >>>>> >>>>no. >>>> >>>> >>>> >>>> >>>>>Regards, >>>>> >>>>>Minh >>>>> >>>>>-----Message d'origine----- >>>>>De : serusers-bounces(a)lists.iptel.org >>>>>[mailto:serusers-bounces@lists.iptel.org] De la >>>>>part de harry gaillac >>>>>Envoyé : mercredi 22 juin 2005 11:41 >>>>>À : serusers(a)lists.iptel.org >>>>>Objet : [Serusers] SER act as a 3cpp server >>>>> >>>>>Hello, >>>>> >>>>>Does SER is able to cat as a 3cpp server ? >>>>> >>>>>Regards >>>>>harry >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>___________________________________________________________________________ >> >> >>> >>> >>> >>> >>>>>Appel audio GRATUIT partout dans le monde avec le >>>>>nouveau Yahoo! Messenger >>>>>Téléchargez cette version sur >>>>>http://fr.messenger.yahoo.com >>>>> >>>>>_______________________________________________ >>>>>Serusers mailing list >>>>>serusers(a)lists.iptel.org >>>>>http://lists.iptel.org/mailman/listinfo/serusers >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>___________________________________________________________________________ >> >> >>> >>> >>> >>> >>>>Appel audio GRATUIT partout dans le monde avec le >>>>nouveau Yahoo! Messenger >>>>Téléchargez cette version sur >>>>http://fr.messenger.yahoo.com >>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>>serusers(a)lists.iptel.org >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>> >>>> >>>> >>> >>> >>> >>> >>> >>> >>> >>___________________________________________________________________________ >> >> >> >>>Appel audio GRATUIT partout dans le monde avec le >>> >>> >>nouveau Yahoo! Messenger >> >> >>>Téléchargez cette version sur >>> >>> >>http://fr.messenger.yahoo.com >> >> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>>. >>> >>> >>> >>> >>> > > > > > > > >___________________________________________________________________________ >Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger >Téléchargez cette version sur http://fr.messenger.yahoo.com > >. > > > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] (no subject)
by lucape＠inwind.it 22 Jun '05

22 Jun '05

can you help me to configure lcs2005 with asterisk... I use SER to resolve the problem that there is for communication protocol...LCS uses tcp, Asterik UDP. Someone, knows how to do the configuration beetwen LCS and SER , SER and Asterisk? the function of asterisk is SIP-PSTN Gateway for the LCS PC-phone communication thanks ____________________________________________________________ Navighi a 4 MEGA e i primi 3 mesi sono GRATIS. Scegli Libero Adsl Flat senza limiti su http://www.libero.it

1 0

[Serusers] SER act as a 3cpp server
by harry gaillac 22 Jun '05

22 Jun '05

Hello, Does SER is able to cat as a 3cpp server ? Regards harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

3 5

[Serusers] variables in ser.cfg
by Jack Wei 22 Jun '05

22 Jun '05

is there anyway to set temp variables in ser.cfg? if so, how to set/access? Jack Wei __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

3 2

[Serusers] Does SER may act as a 3pcc server not 3cpp !
by harry gaillac 22 Jun '05

22 Jun '05

Hello, Does SER is able to cat as a 3pcc server ? http://www.iptel.org/info/players/ietf/callprocessing/3pcc/draft-ietf-sippi… Regards harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 1

[Users] failover and presence/IM service
by jimmy way 22 Jun '05

22 Jun '05

Hi all, 1st question is does openser support Failover and load balance of openser server. That means we can use two or more openser server to handle the SIP request/response and use hot backup. How to do that 2nd question is that I find openser support both Jabber and SIMPLE. It look like Jabber is stable and support and many people from open source world, many plus GOOGLE. But SIMPLE is still unstable. I want to use Jabber for providing presence/IM service, plus file transfer service.(I want to using reSIProcate stack from sipfoundry as client's stack) Does openser support that all? And which is better, Jabber or SIMPLE? Regards, Jimway __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

2 3

Re: [Serusers] append_to_reply and save behaviour
by Samuel Osorio Calvo 22 Jun '05

22 Jun '05

The reason is that the function save sends the response and all further processing can not affect this response (which is already travelling back). I don't know if it can be called bug, feature, or just design... ;) SER config file executes the commands sequentially (besides route blocks) and you have to be careful with the order. Samuel. Unclassified. >>> Fermín Galán Márquez <fermin.galan(a)agora-2000.com> 06/22/05 12:47PM >>> Hello, I've found and estrange behaviour in the 'append_to_reply' action when used in conjunction with 'save' action during REGISTER processing. In particular, if I use in ser.cfg: ... append_to_reply("Service-Route: <sip:orig@scscf.domain1.com;lr>\r\n"); save("domain1"); ... Then the Service-Route header is added to the "200 OK". But if changing the relative order of the actions: ... save("domain1"); append_to_reply("Service-Route: <sip:orig@scscf.domain1.com;lr>\r\n"); ... Then the 'append_to_reply' is ignored: no header is added to the response. I think is a weird behaviour... I would like to know if this is a bug or a feature (I have read the textops documentation at http://www.iptel.org/ser/doc/modules/html/textops.html#AEN104 and nothing about problems with 'save' are mentioned). Thank you in advance! Best regards, ------ Fermín Agora Systems, S. A. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] append_to_reply and save behaviour
by Fermín Galán Márquez 22 Jun '05

22 Jun '05

Hello, I've found and estrange behaviour in the 'append_to_reply' action when used in conjunction with 'save' action during REGISTER processing. In particular, if I use in ser.cfg: ... append_to_reply("Service-Route: <sip:orig@scscf.domain1.com;lr>\r\n"); save("domain1"); ... Then the Service-Route header is added to the "200 OK". But if changing the relative order of the actions: ... save("domain1"); append_to_reply("Service-Route: <sip:orig@scscf.domain1.com;lr>\r\n"); ... Then the 'append_to_reply' is ignored: no header is added to the response. I think is a weird behaviour... I would like to know if this is a bug or a feature (I have read the textops documentation at http://www.iptel.org/ser/doc/modules/html/textops.html#AEN104 and nothing about problems with 'save' are mentioned). Thank you in advance! Best regards, ------ Fermín Agora Systems, S. A.

1 0

Re: [Serusers] Portal for forking call to preferredenddevice-sequential ringing
by Aisling O'Driscoll 22 Jun '05

22 Jun '05

Ok, just to recap - cos Im getting a little bit confused ;) I have two choices(?) 1. Somehow invoke sipsak to configure permanent addresses with a particular q value. 2. Develop a FIFO method to change q value. Am I correct in thinking directly modifying the usrloc table in the database is out of the question because the changes cant be updated except by SER itself in which case a reboot would required - Correct? Also lcr module (load_contacts() etc) isnt suitable for per user configurable sequential forking? Have others tried to implement similar functionality or is it usually a generic site wide sequential forking policy? Many thanks for the opinions and help so far. Aisling. > >---- Original Message ---- >From: jh(a)tutpro.com >To: greger(a)teigre.com >Subject: Re: [Serusers] Portal for forking call to >preferredenddevice-sequential ringing >Date: Tue, 21 Jun 2005 21:41:54 +0300 > >>Greger V. Teigre writes: >> >> > :-D Yes, that is true. But I didn't know that you could change >q-value in >> > FIFO? >> >>if that is not possible, then you can always use sipsak to install >>"permanent" registrations with a given q value. sipsak has an >>additional advantage over fifo in that you can apply permissions >check >>to sipsak registration, but not to fifo registration. >> >>-- juha >> >>-------------------Legal >Disclaimer--------------------------------------- >> >>The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. >> >> -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

5 4

[Serusers] SEMS crashes, fifo errors
by Llanos Serna García-Conde 22 Jun '05

22 Jun '05

1 0

[serusers] calling problem
by michael p 22 Jun '05

22 Jun '05

hi i have a problem : i have 2 same softphone , 1 can call the other and the other cannot call the previous. it's the same configuration for both in the softphone. i did test about this that can't call: i change the softphone, i change the laptop, i use the user that this can call use and change with mine... in basis ser.cfg what do the difference between 2 users ? nothing i guess i'm using a stun to pass over nat and it works in one direction (from one user to the other) it didn't work someone has an idea thanks M.

1 0

[serusers] pa module
by michael p 22 Jun '05

22 Jun '05

hi i'm wroting you cause i saw that you wrote a lot about pa module and you finally succeed to install it. congratulations! i 'm still trying and only one person answered me... please can you help me? my problems and questions are : 1. did you find more docs about pa module cause the README is not very long.? 2. how install it? what the difference between handle_subscription ("registrar") and publish ? 3. how i have to use the pa module and mysql ? i saw in one of your message your ser.cfg but i don't know where you found all the informations to write the lines about pa modules... 4. pa module works with all the softphone (i'm using a softphone from Xten (eyebeam))? how do they know to communicate with this module of SER? how can i do test? please help me it's very important thanks a lot Michael

2 1

Re: [Serusers] Portal for forking call to preferredenddevice-sequential ringing
by Samuel Osorio Calvo 22 Jun '05

22 Jun '05

>> 2. Develop a FIFO method to change q value. > >Yes. Except that I don't know if anybody verified that q value cannot be >changed with today's FIFO. I just asked the question... I think the current usrloc fifo commands provide enough flexibility to modify the q value, I don't know now the exact spelling but womething like: serctl ul add location user@domain user@ip {expires} {q} {replicate} I recall that there where a few values representing expiration, replicate, q value and maybe something else. From the source code, it seems that if the contact already exists, serctl ul add will update the record. If you don't have the user's record values you can use serctl ul show. I'm not 100% sure, but if you use sipsack, you should be carefull to not modifying other parameters just as received, User-Agent...am I right? Samuel. Unclassified. >>> "Greger V. Teigre" <greger(a)teigre.com> 06/22/05 06:42AM >>> Aisling O'Driscoll wrote: > Ok, just to recap - cos Im getting a little bit confused ;) > > I have two choices(?) > > 1. Somehow invoke sipsak to configure permanent addresses with a > particular q value. > 2. Develop a FIFO method to change q value. Yes. Except that I don't know if anybody verified that q value cannot be changed with today's FIFO. I just asked the question... > Am I correct in thinking directly modifying the usrloc table in the > database is out of the question because the changes cant be updated > except by SER itself in which case a reboot would required - Correct? Correct. "SER itself" is here either serctl, sipsak or FIFO. > Also lcr module (load_contacts() etc) isnt suitable for per user > configurable sequential forking? I don't know about that. There is definitely a q-value based functionality there. > Have others tried to implement similar functionality or is it usually > a generic site wide sequential forking policy? I think using CPL could be an option. Have you looked at CPL and the cpl module? g-) > Many thanks for the opinions and help so far. > Aisling. > >> >> ---- Original Message ---- >> From: jh(a)tutpro.com >> To: greger(a)teigre.com >> Subject: Re: [Serusers] Portal for forking call to >> preferredenddevice-sequential ringing >> Date: Tue, 21 Jun 2005 21:41:54 +0300 >> >>> Greger V. Teigre writes: >>> >>>> :-D Yes, that is true. But I didn't know that you could change >> q-value in >>>> FIFO? >>> >>> if that is not possible, then you can always use sipsak to install >>> "permanent" registrations with a given q value. sipsak has an >>> additional advantage over fifo in that you can apply permissions >> check >>> to sipsak registration, but not to fifo registration. >>> >>> -- juha >>> >>> -------------------Legal >> Disclaimer--------------------------------------- >>> >>> The above electronic mail transmission is confidential and intended >> only for the person to whom it is addressed. Its contents may be >> protected by legal and/or professional privilege. Should it be >> received by you in error please contact the sender at the above >> quoted email address. Any unauthorised form of reproduction of this >> message is strictly prohibited. The Institute does not guarantee the >> security of any information electronically transmitted and is not >> liable if the information contained in this communication is not a >> proper and complete record of the message as transmitted by the >> sender nor for any delay in its receipt. >>> >>> > > > > -------------------Legal > Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and intended > only for the person to whom it is addressed. Its contents may be > protected by legal and/or professional privilege. Should it be > received by you in error please contact the sender at the above > quoted email address. Any unauthorised form of reproduction of this > message is strictly prohibited. The Institute does not guarantee the > security of any information electronically transmitted and is not > liable if the information contained in this communication is not a > proper and complete record of the message as transmitted by the > sender nor for any delay in its receipt.

1 0

[Serusers] Tr: FW: Polycom soundpoint IP
by harry gaillac 22 Jun '05

22 Jun '05

Remarque : message transféré en pièce jointe. Hi all, I asked to polycom how to place on hold with music caller when i press hold key. I use SER and asterisk to provide MOH. >From Polycom: We don't support this because when in BroadSoft and Sylantro environments, the servers act as 3pcc's so the phone doesn't need to do anything for the system to get MOH (i.e., we don't need 7155 in these environments. Does Ser is able to act as a 3pcc to provide MOH when hold key is pressed ? http://www.iptel.org/info/players/ietf/callprocessing/3pcc/draft-ietf-sippi… ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] ===== MediaProxy Setup Problem =====
by Freeman 22 Jun '05

22 Jun '05

Hi, I have some problem when i setup the mediaproxy for ser, OS : Redhat 9 Ser : Ser 0.8.14 MediaProxy : MediaProxy 1.3.1 python : python-2.2.2-26 python-optik : python-optik-1.4-2 I tried uppack the mediaproxy to "/usr/local/" and copied the start script from "/usr/local/mediaproxy/boot/mediaproxy.other" to "/etc/init.d/mediaproxy", when i run the start script "/etc/init.d mediaproxy start", i get some error as below : Starting SER MediaProxy server: mediaproxyTraceback (most recent call last): File "/usr/local/mediaproxy/mediaproxy.py", Line 21, in ? from optparse import OptionParser ImportError: No module named optparse Traceback (most recent call last): File "/usr/local/mediaproxy/proxydispatcher.py", line 35, in ? from optparse import OptionParser ImportError: No module named optparse . How can i fix the problem ! How can i config ser.cfg for MediaProxy ? Please help ! Tks a Lot ! Freeman _______________________________________ 想即時收到新 email 通知？下載 Yahoo! Messenger http://messenger.yahoo.com.hk

1 0

[Serusers] ASTERISK+SER+MWI
by harry gaillac 22 Jun '05

22 Jun '05

Hello, I wish to use asterisk as voicemail system with ser My phones are registered on SER and asterisk provide conference ivr and voicemail system. I want to use Asterisk Realtime Architecture to store voicemessages and configuration . Michael Shuler has written a patch for mwi. http://bugs.digium.com/bug_view_page.php?bug_id=0002980 I need help to set up voicemail storage and how to send mwi to clients registered on ser Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

Re: [Serusers] 30s calls
by Greger V. Teigre 22 Jun '05

22 Jun '05

Jerlique Ban wrote: >> In order for the call to not disconnect, the final ACK from >> caller (the ACK to the OK sent by the callee) must be >> received by the callee (Asterisk). So, your problem may be >> caused by your ser.cfg not handling ACKs properly. I have > > The ACK is most certainly being received. I have confimed this by > watching the 'sip debug' in asterisk. I neglected to mention in my > first email that another phone through a different router (with NAT) > work without issue. Its only this one phone. > > What I do see at the time of losing audio, is another INVITE. So, did the OK go through? And did Asterisk send the 100 Trying first? Probably notm because when the UA receices a 100 Trying, it should not resend the INVITE. You could try adding: sl_send_reply("100", "Trying"); Pretty early in your INVITE handling and before forwarding to asterisk. This should at least cause the UA to keep shut for awhile. > If I register the phone direct to Asterisk, it works without issue, > its only having issues through SER. > > I've got it working now, but I'm not sure what I did.... > >> If not, have a look at the >> reference configs at http://onsip.org/ to see how ACKs are handled. > > My config is based on one of their samples :) Excellent! ;-) > PS: I've added a bunch of log() statements to the file, and can see > the output on my terminal, however how do I record the src ip, or URI > with the log function? Use xlog (0.9.x). I have this right after all the "not supported" and integrity checks: # Log all messages xlog("L_INFO","%is sends %rm, r-uri:%ru, from:%fu, to:%tu, status:%rs\n"); g-)

1 0

[Serusers] how does a b2bua help billing
by Iqbal 22 Jun '05

22 Jun '05

Hi I am curious as to how a b2bua does the billing side of things, I know it sits in the middle and the media goes through it, but how does it keep count of how long the session has been open for, does it keep a continously running count of time after a INVITE or use some other non-evasive method. If it keeps a continous count (I presume in memory), how does is initiate a disconnect, does it compare the time used with that left (which is determined at call start) Any links , pointers, Iqbal

3 4

[Serusers] ser.cfg scripting error
by Jack Wei 21 Jun '05

21 Jun '05

my setup is 2 ser servers & 1 mysql server. i want to look up a user in table "location" to see if one of the two ser servers registered the user, so i've decided to follow the example in the admin guide and use exec_dset(). however, i'm getting an error output. anyone knows what's wrong with the following? if ( !exec_dset( 'QUERY="select contact from location where username=\"$SIP_OUSER\""; DEST_URI=`mysql -user -pheslo -h10.1.201.107 -Bs -e"$QUERY" ser`; if [ -z "$DEST_URI" ] then exit 1 fi; echo "$DEST_URI"; echo > /dev/null;' ) ) { xlog("L_INFO", "\n\n%ru is not online (in db)\n"); sl_send_reply( "404", "User does not exist" ); break; }; the error output is: sh: -c: line 4: syntax error: unexpected end of file 12(14932) ERROR:exec_str: no uri from QUERY="select contact from location where username=\"$SIP_OUSER\""; DEST_URI=`mysql -user -pheslo -h10.1.201.107 -Bs -e"$QUERY" ser`; if [ -z "$DEST_URI" ] then exit 1 fi; echo "$DEST_URI"; echo > /dev/null; sip:12222220002@10.1.201.191 12(14932) i'm not sure why 'sip:12222220002@10.1.201.191' is there, but i think that might be part of the problem. when i tried the following: if ( !exec_dset( 'echo sip:12222220002@10.1.201.65; echo > /dev/null' ) ) it works fine. Jack Wei __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

3 2

Re: [Serusers] Portal for forking call to preferred enddevice-sequential ringing
by Greger V. Teigre 21 Jun '05

21 Jun '05

Samuel Osorio Calvo wrote: > In fact, SER has a model for usrloc changes outside itself: FIFO > interface. Using the ul command of this interface it is possible to > add/update/remove users' information during runtime without having to > restart SER. In fact, these commands update SER's cache and this > information is immediately used by SER. It will be propagated to the > permament DB storage depending on the mode selected. :-D Yes, that is true. But I didn't know that you could change q-value in FIFO? Never really thought about that, it's logical, though :-) Kind of double work to build something that uses FIFO for 0.9.x, when head (hopefully) will get the xmlrpc. It looks like Jan plans to remove the fifo code when replacing with xmlrpc. > I would suggest to add a chapter of the FIFO/UNIX/TCP/XMLRPC > interface to the famous Getting Started Document ;) Good idea :-) However, it's sort of a moving target so far, it's quite an effort to keep the document updated, and we would rather suffer in coverage/depth than suffer from outdated info... g-) > Samuel. > > > Unclassified. >>>> "Greger V. Teigre" <greger(a)teigre.com> 06/21/05 04:53PM >>> > You get into even worse problems as SER does not really have a model > for > usrloc changes outside itself. Thus, dependent on your db mode > (memory-only, > flush, write-through etc) SER will potentially overwrite your > externally set > q-value before stopping. If you use write-through, you probably don't > get > that problem, and yes, you could probably theoretically restart SER to > reload the usrloc with new q-values (never validated that, though). > However, with a lot of locations, it's not really feasible due to > SER's > start-up loading of locations (and being unavailable while doing so). > Also, > you will probably restart in the middle of a transaction and calls > may fail (unless you use the restart safe parameter). > g-) > > Aisling wrote: >> Thank you for the replys. I will look up the post you suggested for >> further info on the lcr module. One thing I just want to clarify: >> >> " DB-based changes is the way to go - Updating the usrloc table >> outside >> SER does not work as SER loads usrloc info into memory."...... >> >> Does this mean I must modify the database and then restart SER, >> thereby >> disrupting the service for other users? >> >> Thanks, >> Aisling. >> >> -----Original Message----- >> From: Greger V. Teigre [mailto:greger@teigre.com] >> Sent: 21 June 2005 13:33 >> To: Samuel Osorio Calvo; ashling.odriscoll(a)cit.ie; serusers(a)lists.iptel.org >> Subject: Re: [Serusers] Portal for forking call to preferred end >> device-sequential ringing >> >> I believe the lcr module will do q-value based sequential forking. >> There >> was a thread on that a while back. Search for q-value and Juha. LCR >> can >> be >> found in head and as a backported 0.9.x module. >> DB-based changes is the way to go! Doing dynamic ser.cfg changes is >> not >> feasible as long you have to restart ser (and thus reload contacts, >> which >> may take a long time). Updating the usrloc table outside SER does >> not work >> as SER loads usrloc info into memory. >> g-) >> >> Samuel Osorio Calvo wrote: >>> I guess a sequential forking ordered with the q value of the contact >>> header field value would do what you are asking for. The user has >>> just to configure the appropriate value in his/her UA to the >>> preferred ringing sequence. This method is fully SIP compliant but >>> what I am not sure is if SER does sequential forking in the q order >>> (it wasn't a few months ago but some modifications might have added >>> it...someone can tell you better than me). >>> If you still want users to modify SER's database with a web >>> interface, you might try to modify the q value in the usrloc >>> database, of course if SER does q-ordered sequential forking, but I >>> just think it adds lots of complexity to a feature which SER is >>> supposed to provide together with minum user configuration in the >>> endpoints. >>> >>> Hope it helps, >>> >>> Samuel. >>> >>> >>> Unclassified. >>>>>> "Aisling" <ashling.odriscoll(a)cit.ie> 06/21/05 12:54PM >>> >>> Hello all, >>> >>> I'm hoping someone will offer an opinion as to how I should approach >>> the >>> following and if I am thinking along the correct lines: >>> >>> I am creating a web application where a user can dictate which >>> device they want a call delivered to. So if I have a user with sip >>> url "sip:2000@server" and they have registered with the SER server >>> from a pda, pc and laptop, SER will currently parallel fork a call >>> to all these >>> destinations causing all softphones to ring (correct?). However I >>> want a >>> user to choose from a drop down menu in their browser (which I'm >>> developing with servlets and JSP's) their preferred phone e.g. pda >>> and then SER will fork the call to the softphone on the pda first. >>> >>> So basically my current plan is to retrieve the information from the >>> user about their preferred phone (which will be associated with a >>> particular IP address) and then dynamically modify the ser.cfg for a >>> sequential forking rule for that user. I would appreciate opinions >>> as to >>> whether I am approaching this correctly or if there's an alternative >>> method for such functionality (perhaps such as forking the call to >>> the device which last registered or something)? >>> >>> Many Thanks, >>> Aisling. >>> >>> >>> >>> >>> -------------------Legal >>> Disclaimer--------------------------------------- >>> >>> The above electronic mail transmission is confidential and intended >>> only for the person to whom it is addressed. Its contents may be >>> protected by legal and/or professional privilege. Should it be >>> received by you in error please contact the sender at the above >>> quoted email address. Any unauthorised form of reproduction of this >>> message is strictly prohibited. The Institute does not guarantee the >>> security of any information electronically transmitted and is not >>> liable if the information contained in this communication is not a >>> proper and complete record of the message as transmitted by the >>> sender nor for any delay in its receipt. >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> -------------------Legal >> Disclaimer--------------------------------------- >> >> The above electronic mail transmission is confidential and intended >> only >> for the person to whom it is addressed. Its contents may be protected >> by >> legal and/or professional privilege. Should it be received by you in >> error please contact the sender at the above quoted email address. >> Any unauthorised form of reproduction of this message is strictly >> prohibited. The Institute does not guarantee the security of any >> information electronically transmitted and is not liable if the >> information contained in this communication is not a proper and >> complete >> record of the message as transmitted by the sender nor for any delay >> in >> its receipt. >> >> >> -------------------Legal >> Disclaimer--------------------------------------- >> >> The above electronic mail transmission is confidential and intended >> only for the person to whom it is addressed. Its contents may be >> protected by legal and/or professional privilege. Should it be >> received by you in error please contact the sender at the above >> quoted email address. Any unauthorised form of reproduction of this >> message is strictly prohibited. The Institute does not guarantee the >> security of any information electronically transmitted and is not >> liable if the information contained in this communication is not a >> proper and complete record of the message as transmitted by the >> sender nor for any delay in its receipt.

3 2

[Serusers] Tr: Re: [Sems] sems module
by harry gaillac 21 Jun '05

21 Jun '05

Remarque : message transféré en pièce jointe. ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] Problem with Voice in SER0.9.0
by Bruno Oliveira 21 Jun '05

21 Jun '05

hi when I make one call enters two kphoner i don't have voice but the message functions. anyone know why ? thanks for your time -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

2 3

Re: [Serusers] Portal for forking call to preferred end device-sequential ringing
by Samuel Osorio Calvo 21 Jun '05

21 Jun '05

In fact, SER has a model for usrloc changes outside itself: FIFO interface. Using the ul command of this interface it is possible to add/update/remove users' information during runtime without having to restart SER. In fact, these commands update SER's cache and this information is immediately used by SER. It will be propagated to the permament DB storage depending on the mode selected. I would suggest to add a chapter of the FIFO/UNIX/TCP/XMLRPC interface to the famous Getting Started Document ;) Samuel. Unclassified. >>> "Greger V. Teigre" <greger(a)teigre.com> 06/21/05 04:53PM >>> You get into even worse problems as SER does not really have a model for usrloc changes outside itself. Thus, dependent on your db mode (memory-only, flush, write-through etc) SER will potentially overwrite your externally set q-value before stopping. If you use write-through, you probably don't get that problem, and yes, you could probably theoretically restart SER to reload the usrloc with new q-values (never validated that, though). However, with a lot of locations, it's not really feasible due to SER's start-up loading of locations (and being unavailable while doing so). Also, you will probably restart in the middle of a transaction and calls may fail (unless you use the restart safe parameter). g-) Aisling wrote: > Thank you for the replys. I will look up the post you suggested for > further info on the lcr module. One thing I just want to clarify: > > " DB-based changes is the way to go - Updating the usrloc table > outside > SER does not work as SER loads usrloc info into memory."...... > > Does this mean I must modify the database and then restart SER, > thereby > disrupting the service for other users? > > Thanks, > Aisling. > > -----Original Message----- > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: 21 June 2005 13:33 > To: Samuel Osorio Calvo; ashling.odriscoll(a)cit.ie; serusers(a)lists.iptel.org > Subject: Re: [Serusers] Portal for forking call to preferred end > device-sequential ringing > > I believe the lcr module will do q-value based sequential forking. > There > was a thread on that a while back. Search for q-value and Juha. LCR > can > be > found in head and as a backported 0.9.x module. > DB-based changes is the way to go! Doing dynamic ser.cfg changes is > not > feasible as long you have to restart ser (and thus reload contacts, > which > may take a long time). Updating the usrloc table outside SER does not > work > as SER loads usrloc info into memory. > g-) > > Samuel Osorio Calvo wrote: >> I guess a sequential forking ordered with the q value of the contact >> header field value would do what you are asking for. The user has >> just to configure the appropriate value in his/her UA to the >> preferred ringing sequence. This method is fully SIP compliant but >> what I am not sure is if SER does sequential forking in the q order >> (it wasn't a few months ago but some modifications might have added >> it...someone can tell you better than me). >> If you still want users to modify SER's database with a web >> interface, you might try to modify the q value in the usrloc >> database, of course if SER does q-ordered sequential forking, but I >> just think it adds lots of complexity to a feature which SER is >> supposed to provide together with minum user configuration in the >> endpoints. >> >> Hope it helps, >> >> Samuel. >> >> >> Unclassified. >>>>> "Aisling" <ashling.odriscoll(a)cit.ie> 06/21/05 12:54PM >>> >> Hello all, >> >> I'm hoping someone will offer an opinion as to how I should approach >> the >> following and if I am thinking along the correct lines: >> >> I am creating a web application where a user can dictate which device >> they want a call delivered to. So if I have a user with sip url >> "sip:2000@server" and they have registered with the SER server from a >> pda, pc and laptop, SER will currently parallel fork a call to all >> these >> destinations causing all softphones to ring (correct?). However I >> want a >> user to choose from a drop down menu in their browser (which I'm >> developing with servlets and JSP's) their preferred phone e.g. pda >> and then SER will fork the call to the softphone on the pda first. >> >> So basically my current plan is to retrieve the information from the >> user about their preferred phone (which will be associated with a >> particular IP address) and then dynamically modify the ser.cfg for a >> sequential forking rule for that user. I would appreciate opinions as >> to >> whether I am approaching this correctly or if there's an alternative >> method for such functionality (perhaps such as forking the call to >> the device which last registered or something)? >> >> Many Thanks, >> Aisling. >> >> >> >> >> -------------------Legal >> Disclaimer--------------------------------------- >> >> The above electronic mail transmission is confidential and intended >> only for the person to whom it is addressed. Its contents may be >> protected by legal and/or professional privilege. Should it be >> received by you in error please contact the sender at the above >> quoted email address. Any unauthorised form of reproduction of this >> message is strictly prohibited. The Institute does not guarantee the >> security of any information electronically transmitted and is not >> liable if the information contained in this communication is not a >> proper and complete record of the message as transmitted by the >> sender nor for any delay in its receipt. >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > > > -------------------Legal > Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and intended > only > for the person to whom it is addressed. Its contents may be protected > by > legal and/or professional privilege. Should it be received by you in > error please contact the sender at the above quoted email address. Any > unauthorised form of reproduction of this message is strictly > prohibited. The Institute does not guarantee the security of any > information electronically transmitted and is not liable if the > information contained in this communication is not a proper and > complete > record of the message as transmitted by the sender nor for any delay > in > its receipt. > > > -------------------Legal > Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and intended > only for the person to whom it is addressed. Its contents may be > protected by legal and/or professional privilege. Should it be > received by you in error please contact the sender at the above > quoted email address. Any unauthorised form of reproduction of this > message is strictly prohibited. The Institute does not guarantee the > security of any information electronically transmitted and is not > liable if the information contained in this communication is not a > proper and complete record of the message as transmitted by the > sender nor for any delay in its receipt.

1 0

[Users] avpop request and lcr ideas
by Kristin Galway 21 Jun '05

21 Jun '05

Thanks for creating this branch away from the seemingly stagnant SER project. I have in the past seen that requests for help or features have fallen on empty replies as evidenced while searching the mailing lists. I hope that this will be different here. I hope that be able to contribute back to this project once I become more versed in the inner workings as I see a very strong need for the following: Route lookups based on OCN/LERG data and carrier cost tables. (I already have the queries to do the but not the means to implement yet) This function would set say a defined number of flags to indicate outbound carriers based on cost. Having a module to do this would allow LCR lookups with fallover capabilities but based on OCN's and carrier data. Something else I would like to see is a way of using avops to set a fr_inv_timeout value. For instance: if (avp_db_load("$ruri/username", "s:voicemail")) { if (avp_check("s:voicemail", "eq/y/i")) { xlog("L_N", "\033[1m\033[36m ROUTE[0] - s:voicemail = y \033[0m\n"); # recipient has voicemail enabled. set flag(31) setflag(31); # test for a user set timeout if (avp_db_load("$ruri/username", i:vmtimeout")) { avp_write("$vmtimeout", s:callee_fr_inv_timer"); }; }; }; Thanks! Kristin ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

2 1

[Serusers] Re: [Serdev] rel_0_9_0 release critical bugs/missing things
by Greger V. Teigre 21 Jun '05

21 Jun '05

As far I can see (using a filter on Matches versions: 0.9.3), there are no issues. The same search, but using HEAD, gives me 22 issues. Jan stated that everything he is aware of has been registered at http://bugs.sip-router.org/. I would like to make a call for submissions on http://bugs.sip-router.org/ of lacking documentation that you either have identified and started on (ref. earlier calls for documentation) or just identified (so that others may pick up the documentation effort). Let's try to make 0.9.x documentation as complete as possible now leading up to the release!! g-) Klaus Darilion wrote: > maybe on the bugtracker? > > Juha Heinanen wrote: >> is there somewhere a list of rel_0_9_0 release critical bugs/missing >> things? it would be high time to release the next stable version >> and if the lists exists, perhaps people could help to fix the loose >> ends. -- juha >> >> _______________________________________________ >> Serdev mailing list >> serdev(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serdev >> >> > > _______________________________________________ > Serdev mailing list > serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev

6 12

Re: [Serusers] Portal for forking call to preferred end device -sequential ringing
by Samuel Osorio Calvo 21 Jun '05

21 Jun '05

I guess a sequential forking ordered with the q value of the contact header field value would do what you are asking for. The user has just to configure the appropriate value in his/her UA to the preferred ringing sequence. This method is fully SIP compliant but what I am not sure is if SER does sequential forking in the q order (it wasn't a few months ago but some modifications might have added it...someone can tell you better than me). If you still want users to modify SER's database with a web interface, you might try to modify the q value in the usrloc database, of course if SER does q-ordered sequential forking, but I just think it adds lots of complexity to a feature which SER is supposed to provide together with minum user configuration in the endpoints. Hope it helps, Samuel. Unclassified. >>> "Aisling" <ashling.odriscoll(a)cit.ie> 06/21/05 12:54PM >>> Hello all, I'm hoping someone will offer an opinion as to how I should approach the following and if I am thinking along the correct lines: I am creating a web application where a user can dictate which device they want a call delivered to. So if I have a user with sip url "sip:2000@server" and they have registered with the SER server from a pda, pc and laptop, SER will currently parallel fork a call to all these destinations causing all softphones to ring (correct?). However I want a user to choose from a drop down menu in their browser (which I'm developing with servlets and JSP's) their preferred phone e.g. pda and then SER will fork the call to the softphone on the pda first. So basically my current plan is to retrieve the information from the user about their preferred phone (which will be associated with a particular IP address) and then dynamically modify the ser.cfg for a sequential forking rule for that user. I would appreciate opinions as to whether I am approaching this correctly or if there's an alternative method for such functionality (perhaps such as forking the call to the device which last registered or something)? Many Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

4 4

[Serusers] iptel is hiring
by Jiri Kuthan 21 Jun '05

21 Jun '05

iptel is hiring developers who are familiar with programming in C, C++, and understand TCP/IP, SIP and ideally SER+SEMS. iptel is present in two locations: Berlin, Germany and Prague, Czech Republic. In some cases, we may consider remote work as well. Independent of the location, we are having nice offices with scenic views over the city and rooms crowded with seazoned SER developers :) I will be happy to take inquiries and applications off-line. Please include CV with work experience. Thank you! -jiri PS -- I am continously travelling, so don't be surprised if I cannot respond in real time. -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[serusers] pa module
by michael p 21 Jun '05

21 Jun '05

hi all i have some problems to include the pa module in SER. in the README it's a very quick config. i don't understand where exactly i hvae to put the function "handle_subscription("registrar");" ? i don't understand how it works tha pa module to create the tables for pa module ? (i don't find the script modules/pa/pa.sql ---- how can i find it and how i have to run it? ) thanks for the help Mike

1 0

Re: [Serusers] Is there an AVP to alter "expires=" in REGISTERmessages?
by Samuel Osorio Calvo 21 Jun '05

21 Jun '05

I totally agree on the solution, luck with the vendor.... Samuel. Unclassified. >>> Java Rockx <javarockx(a)gmail.com> 06/21/05 01:40PM >>> Thanks for the feedback all. I've been considering what to do with this issue, and I decided that it is better to (but more painful) to have the vendor fix their firmware than it is for me to alter SER's functionality in a way as to move it away from RFC compliance. The save_extended_expires suggestion would be an easy fix, but the right thing to do is fix the problem rather than the symptom. Regards, Paul On 6/21/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > > I was taking a more detailled look at the code and it seems that the > Contact header will not be added into the reply if you use > save_noreply+sl_send_reply because the Contact buffer is added inside the > registrar module....but I recall that it worked. I suggest you to give it a > try and see the answer with ethereal|ngrep|* maybe it is taken after when > constructing the answer from the sipmsg. > Anyhow it would be easy to add a save_extended_expires command that makes > what you need but, as I said, people can start using these SIP patches to > build even more non-compliant proxies with SER.... > > Samuel. > > > Unclassified. > >>> "Greger V. Teigre" <greger(a)teigre.com> 06/21/05 07:19AM >>> > You're right of course :-) > Samuel's approach is more correct, but I'm not sure you will be able to > generate the correct reply using save_noreply and sl_reply (with Contact) ?! > Does save_noreply generate the reply? > g-) > > ---- Original Message ---- > From: Java Rockx > To: Greger V. Teigre > Cc: serusers > Sent: Monday, June 20, 2005 02:21 PM > Subject: Re: [Serusers] Is there an AVP to alter "expires=" in > REGISTERmessages? > > > Greger, > > > > That would be possible, but the way SER generates the 200OK reply, it > > would contain the altered "expires=" value. By doing so, the ATA > > would take that new value and use it, so if the ATA sent the REGISTER > > with an expires=300 and SER replied with a 200OK and a contact header > > containing expires=360, the ATA would not send another REGISTER for > > 360 seconds. > > > > I was fooling around with sip_msg.c and reply.c were I padded the > > expires with 60 seconds before saving to usrloc and then removing the > > pad before sending the reply and it works, but I would just feel > > better by not altering the source code. > > > > I guess I'd just feel more comfortable if SER had some sort of AVP or > > modparam for the registrar module that did this automagically. > > > > Regards, > > Paul > > > > > > On 6/20/05, Greger V. Teigre <greger(a)teigre.com> wrote: > > It was supposed to be short, but maybe too short. I meant doing the > > subst on the REGISTER message before save. > > g-) > > ---- Original Message ---- > > From: Greger V. Teigre > > To: Java Rockx ; serusers > > Sent: Monday, June 20, 2005 09:01 AM > > Subject: Re: [Serusers] Is there an AVP to alter "expires=" in > > REGISTERmessages? > > > >> Hi Paul, > >> Would it be possible to "do it upside down" ? ;-) I.e. do a > >> subst("/^Expires:(.*)[0-9]$/ Expires: yourvalue+60/gi"); > >> g-) > >> > >> ---- Original Message ---- > >> From: Java Rockx > >> To: serusers > >> Sent: Sunday, June 19, 2005 04:28 PM > >> Subject: [Serusers] Is there an AVP to alter "expires=" in REGISTER > >> messages? > >> > >>> Hi All. > >>> > >>> We've got some ATAs that don't quite re-REGISTER in a timely fashion > >>> and therefore SER expires the usrloc record a few seconds before the > >>> ATA sends another REGISTER message. > >>> > >>> Is there an AVP or something to add, for example, 60 seconds to the > >>> "expires=" value in the REGISTER message prior to calling save()? By > >>> doing so, these "broken" ATAs would re-REGISTER before SER removes > >>> the usrloc record from cache. > >>> > >>> Regards, > >>> Paul > >>> > >>> > >>> > >>> > >>> _______________________________________________ > >>> Serusers mailing list > >>> serusers(a)lists.iptel.org > >>> http://lists.iptel.org/mailman/listinfo/serusers > >> > >> > >> > >> _______________________________________________ > >> Serusers mailing list > >> serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Users] Re: no solution yet :( that's why i've decided to write to you
by Daniel-Constantin Mierla 21 Jun '05

21 Jun '05

You have to create a dialing plan for openser also. In your script, after the block: if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; (Just four your information, you don't need "if (uri==myself)" because will be true all the time) Back to dialing plan, you have to add some rules like: if(uri=~"^sip:05[0-9]+@") { # call to mobile rewritehost("192.168.1.35 <mailto:20105353490056@192.168.1.35%29>"); prefix("201"); route(1); break; } Daniel On 06/20/05 11:00, Ozan Blotter wrote: > Dear Team, > > I know this is annoying to post nearly same things but still no answer > :( Is it really so hard or can anybody tell me why this is not working. > > > Basically the system is as : > > ATA 1 : 12345 (registered and can call 54321, no problem) This is > Zyxel P2002. > ATA 2 : 54321 (registered and can call 12345, no problem) This is > Zyxel P2002 too. > ANALOG FXO : 201 (registered as 201, for test purposes i only > registered one port of it) This is Eusso 7104-S 4 Port Analog FXO and > i can add 3 more ports as available. > > With Brekeke's Ondo SIP Proxy, i created a simple dial-plan and it > worked, even i can add more prefixes no problem. But in SER i cannot > :( I'm using OpenSER. Normally it's like when i want to call a GSM, > proxy simply adds a prefix 201 which is same with registered sip > number for regarding analog port and forwards it (looks like > sip:20105353490056@192.168.1.35) > <mailto:20105353490056@192.168.1.35%29> to Gateway and Gateway simply > strips 201 from it and dials 05353490056, yes it connects. But what > this same does not work on SER, any ideas because it's very boring to > play all day and night ? > > Thanks.... > > This is my current SER.CFG as it comes with OPENSER. Can you please > help me to add lines where or/and add remove somethings... > > [root@localhost openser]# cat ser.cfg > # > # $Id: openser.cfg,v 1.1.1.1 2005/06/13 16:47:30 bogdan_iancu Exp $ > # > # simple quick-start config script > # > > # ----------- global configuration parameters ------------------------ > > #debug=3 # debug level (cmd line: -dddddddddd) > #fork=yes > #log_stderror=no # (cmd line: -E) > > /* Uncomment these lines to enter debugging mode > fork=no > log_stderror=yes > */ > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > #port=5060 > #children=4 > fifo="/tmp/openser_fifo" > > # ------------------ module loading ---------------------------------- > > # Uncomment this if you want to use SQL database > #loadmodule "/usr/local/lib/ser/modules/mysql.so" > > loadmodule "/usr/lib/openser/modules/sl.so" > loadmodule "/usr/lib/openser/modules/tm.so" > loadmodule "/usr/lib/openser/modules/rr.so" > loadmodule "/usr/lib/openser/modules/maxfwd.so" > loadmodule "/usr/lib/openser/modules/usrloc.so" > loadmodule "/usr/lib/openser/modules/registrar.so" > loadmodule "/usr/lib/openser/modules/textops.so" > > # Uncomment this if you want digest authentication > # mysql.so must be loaded ! > #loadmodule "/usr/lib/openser/modules/auth.so" > #loadmodule "/usr/lib/openser/modules/auth_db.so" > > # ----------------- setting module-specific parameters --------------- > > # -- usrloc params -- > > modparam("usrloc", "db_mode", 0) > > # Uncomment this if you want to use SQL database > # for persistent storage and comment the previous line > #modparam("usrloc", "db_mode", 2) > > # -- auth params -- > # Uncomment if you are using auth module > # > #modparam("auth_db", "calculate_ha1", yes) > # > # If you set "calculate_ha1" parameter to yes (which true in this config), > # uncomment also the following parameter) > # > #modparam("auth_db", "password_column", "password") > > # -- rr params -- > # add value to ;lr param to make some broken UAs happy > modparam("rr", "enable_full_lr", 1) > > # ------------------------- request routing logic ------------------- > > # main routing logic > > route{ > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > break; > }; > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > break; > }; > > > # we record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > if (!method=="REGISTER") record_route(); > > # subsequent messages withing a dialog should take the > # path determined by record-routing > if (loose_route()) { > # mark routing logic in request > append_hf("P-hint: rr-enforced\r\n"); > route(1); > break; > }; > > if (!uri==myself) { > # mark routing logic in request > append_hf("P-hint: outbound\r\n"); > route(1); > break; > }; > > # if the request is for other domain use UsrLoc > # (in case, it does not work, use the following command > # with proper names and addresses in it) > if (uri==myself) { > > if (method=="REGISTER") { > > # Uncomment this if you want to use digest authentication > # if (!www_authorize("openser.org", "subscriber")) { > # www_challenge("openser.org", "0"); > # break; > # }; > > save("location"); > break; > }; > > lookup("aliases"); > if (!uri==myself) { > append_hf("P-hint: outbound alias\r\n"); > route(1); > break; > }; > > # native SIP destinations are handled using our USRLOC DB > if (!lookup("location")) { > sl_send_reply("404", "Not Found"); > break; > }; > }; > append_hf("P-hint: usrloc applied\r\n"); > route(1); > } > > route[1] > { > # send it out now; use stateful forwarding as it works reliably > # even for UDP2TCP > if (!t_relay()) { > sl_reply_error(); > }; > } > > [root@localhost openser]# > >------------------------------------------------------------------------ > >_______________________________________________ >Team mailing list >Team(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/team > >

2 3

Re: [Serusers] Is there an AVP to alter "expires=" in REGISTERmessages?
by Java Rockx 21 Jun '05

21 Jun '05

Thanks for the feedback all. I've been considering what to do with this issue, and I decided that it is better to (but more painful) to have the vendor fix their firmware than it is for me to alter SER's functionality in a way as to move it away from RFC compliance. The save_extended_expires suggestion would be an easy fix, but the right thing to do is fix the problem rather than the symptom. Regards, Paul On 6/21/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > > I was taking a more detailled look at the code and it seems that the > Contact header will not be added into the reply if you use > save_noreply+sl_send_reply because the Contact buffer is added inside the > registrar module....but I recall that it worked. I suggest you to give it a > try and see the answer with ethereal|ngrep|* maybe it is taken after when > constructing the answer from the sipmsg. > Anyhow it would be easy to add a save_extended_expires command that makes > what you need but, as I said, people can start using these SIP patches to > build even more non-compliant proxies with SER.... > > Samuel. > > > Unclassified. > >>> "Greger V. Teigre" <greger(a)teigre.com> 06/21/05 07:19AM >>> > You're right of course :-) > Samuel's approach is more correct, but I'm not sure you will be able to > generate the correct reply using save_noreply and sl_reply (with Contact) ?! > Does save_noreply generate the reply? > g-) > > ---- Original Message ---- > From: Java Rockx > To: Greger V. Teigre > Cc: serusers > Sent: Monday, June 20, 2005 02:21 PM > Subject: Re: [Serusers] Is there an AVP to alter "expires=" in > REGISTERmessages? > > > Greger, > > > > That would be possible, but the way SER generates the 200OK reply, it > > would contain the altered "expires=" value. By doing so, the ATA > > would take that new value and use it, so if the ATA sent the REGISTER > > with an expires=300 and SER replied with a 200OK and a contact header > > containing expires=360, the ATA would not send another REGISTER for > > 360 seconds. > > > > I was fooling around with sip_msg.c and reply.c were I padded the > > expires with 60 seconds before saving to usrloc and then removing the > > pad before sending the reply and it works, but I would just feel > > better by not altering the source code. > > > > I guess I'd just feel more comfortable if SER had some sort of AVP or > > modparam for the registrar module that did this automagically. > > > > Regards, > > Paul > > > > > > On 6/20/05, Greger V. Teigre <greger(a)teigre.com> wrote: > > It was supposed to be short, but maybe too short. I meant doing the > > subst on the REGISTER message before save. > > g-) > > ---- Original Message ---- > > From: Greger V. Teigre > > To: Java Rockx ; serusers > > Sent: Monday, June 20, 2005 09:01 AM > > Subject: Re: [Serusers] Is there an AVP to alter "expires=" in > > REGISTERmessages? > > > >> Hi Paul, > >> Would it be possible to "do it upside down" ? ;-) I.e. do a > >> subst("/^Expires:(.*)[0-9]$/ Expires: yourvalue+60/gi"); > >> g-) > >> > >> ---- Original Message ---- > >> From: Java Rockx > >> To: serusers > >> Sent: Sunday, June 19, 2005 04:28 PM > >> Subject: [Serusers] Is there an AVP to alter "expires=" in REGISTER > >> messages? > >> > >>> Hi All. > >>> > >>> We've got some ATAs that don't quite re-REGISTER in a timely fashion > >>> and therefore SER expires the usrloc record a few seconds before the > >>> ATA sends another REGISTER message. > >>> > >>> Is there an AVP or something to add, for example, 60 seconds to the > >>> "expires=" value in the REGISTER message prior to calling save()? By > >>> doing so, these "broken" ATAs would re-REGISTER before SER removes > >>> the usrloc record from cache. > >>> > >>> Regards, > >>> Paul > >>> > >>> > >>> > >>> > >>> _______________________________________________ > >>> Serusers mailing list > >>> serusers(a)lists.iptel.org > >>> http://lists.iptel.org/mailman/listinfo/serusers > >> > >> > >> > >> _______________________________________________ > >> Serusers mailing list > >> serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Serusers] new voicamail.so
by Llanos Serna García-Conde 21 Jun '05

21 Jun '05

1 0

[Users] OpenSER Web Site hosting
by Daniel-Constantin Mierla 21 Jun '05

21 Jun '05

Hello everybody, currently the openser.org web site is hosted by Voice-System server. It has to be moved on a separate hosting server, so that developers and contributors can have full access to it. Are you aware of a good hosting service? It has to be a Linux/Unix server, allow ssh access, provide a good bandwidth and the price to be as much as possible. I have opened a topic in openser forum, too, -- you can use what you like more to reply. Daniel

4 6

[Serusers] Portal for forking call to preferred end device - sequential ringing
by Aisling 21 Jun '05

21 Jun '05

Hello all, I'm hoping someone will offer an opinion as to how I should approach the following and if I am thinking along the correct lines: I am creating a web application where a user can dictate which device they want a call delivered to. So if I have a user with sip url "sip:2000@server" and they have registered with the SER server from a pda, pc and laptop, SER will currently parallel fork a call to all these destinations causing all softphones to ring (correct?). However I want a user to choose from a drop down menu in their browser (which I'm developing with servlets and JSP's) their preferred phone e.g. pda and then SER will fork the call to the softphone on the pda first. So basically my current plan is to retrieve the information from the user about their preferred phone (which will be associated with a particular IP address) and then dynamically modify the ser.cfg for a sequential forking rule for that user. I would appreciate opinions as to whether I am approaching this correctly or if there's an alternative method for such functionality (perhaps such as forking the call to the device which last registered or something)? Many Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

Re: [Serusers] Re: [Serdev] rel_0_9_0 release critical bugs/missingthings
by Samuel Osorio Calvo 21 Jun '05

21 Jun '05

Undocummented code = non existing code (for almomst everyone except the developer and few guys reading the devel list, updating the working copy of the CVS, and checking the new code).....I know it is unpleasant, but two lines with an overview is not asking too much.....isn't it? my contribution, Samuel. Unclassified. >>> Jan Janak <jan(a)iptel.org> 06/21/05 11:58AM >>> On 21-06-2005 09:20, Juha Heinanen wrote: > Greger V. Teigre writes: > > > Let's try to make 0.9.x documentation as complete as possible now > > leading up to the release!! > > nathelper README doesn't describe all nat_uac_test modes. tm module > README seems to lack description of several exported functions. > > these both are example of bad iptel.org policy. no new features should > be allowed to be added/changed in the code unless also the documentation > is updated. Are you suggesting that commits should be rejected if they do not update documentation as well ? What is better, having undocumented code or having no code ? Jan. _______________________________________________ Serdev mailing list serdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev

1 0

Re: [Serusers] Is there an AVP to alter "expires=" in REGISTERmessages?
by Samuel Osorio Calvo 21 Jun '05

21 Jun '05

I was taking a more detailled look at the code and it seems that the Contact header will not be added into the reply if you use save_noreply+sl_send_reply because the Contact buffer is added inside the registrar module....but I recall that it worked. I suggest you to give it a try and see the answer with ethereal|ngrep|* maybe it is taken after when constructing the answer from the sipmsg. Anyhow it would be easy to add a save_extended_expires command that makes what you need but, as I said, people can start using these SIP patches to build even more non-compliant proxies with SER.... Samuel. Unclassified. >>> "Greger V. Teigre" <greger(a)teigre.com> 06/21/05 07:19AM >>> You're right of course :-) Samuel's approach is more correct, but I'm not sure you will be able to generate the correct reply using save_noreply and sl_reply (with Contact) ?! Does save_noreply generate the reply? g-) ---- Original Message ---- From: Java Rockx To: Greger V. Teigre Cc: serusers Sent: Monday, June 20, 2005 02:21 PM Subject: Re: [Serusers] Is there an AVP to alter "expires=" in REGISTERmessages? > Greger, > > That would be possible, but the way SER generates the 200OK reply, it > would contain the altered "expires=" value. By doing so, the ATA > would take that new value and use it, so if the ATA sent the REGISTER > with an expires=300 and SER replied with a 200OK and a contact header > containing expires=360, the ATA would not send another REGISTER for > 360 seconds. > > I was fooling around with sip_msg.c and reply.c were I padded the > expires with 60 seconds before saving to usrloc and then removing the > pad before sending the reply and it works, but I would just feel > better by not altering the source code. > > I guess I'd just feel more comfortable if SER had some sort of AVP or > modparam for the registrar module that did this automagically. > > Regards, > Paul > > > On 6/20/05, Greger V. Teigre <greger(a)teigre.com> wrote: > It was supposed to be short, but maybe too short. I meant doing the > subst on the REGISTER message before save. > g-) > ---- Original Message ---- > From: Greger V. Teigre > To: Java Rockx ; serusers > Sent: Monday, June 20, 2005 09:01 AM > Subject: Re: [Serusers] Is there an AVP to alter "expires=" in > REGISTERmessages? > >> Hi Paul, >> Would it be possible to "do it upside down" ? ;-) I.e. do a >> subst("/^Expires:(.*)[0-9]$/ Expires: yourvalue+60/gi"); >> g-) >> >> ---- Original Message ---- >> From: Java Rockx >> To: serusers >> Sent: Sunday, June 19, 2005 04:28 PM >> Subject: [Serusers] Is there an AVP to alter "expires=" in REGISTER >> messages? >> >>> Hi All. >>> >>> We've got some ATAs that don't quite re-REGISTER in a timely fashion >>> and therefore SER expires the usrloc record a few seconds before the >>> ATA sends another REGISTER message. >>> >>> Is there an AVP or something to add, for example, 60 seconds to the >>> "expires=" value in the REGISTER message prior to calling save()? By >>> doing so, these "broken" ATAs would re-REGISTER before SER removes >>> the usrloc record from cache. >>> >>> Regards, >>> Paul >>> >>> >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] openser/ser - avoiding forks
by Andrei Pelinescu-Onciul 21 Jun '05

21 Jun '05

I think that a fork is bad for everybody. Whatever way you put it, it will still involve extra overhead at least for each code merge. So this mail is an attempt to come to a compromise that would be acceptable for everybody. My first proposal would be to have another branch on ser cvs (berlios). Something like openser_0.x.y (where the version number should reflect somehow that it's newer than 0.9 but not unstable). Bogdan and Daniel would mantain this branch and they would be able to release from it as often as they want. It would be some kind of stable on steroids :-) This would have the main advantage that code merges/diffs/fixes propagation would be much easier and much faster. Administrating all the stuff will be also easier (like common bug tracking system a.s.o). It would be also easier for users, it will be much more easier to keep track of the changes and to choose an appropiate version. There must be however some rules. Something like: backporting/forwadporting from unstable/stable should be ok, the same for minor patches (e.g. xlog colors), but not major code changes, not present in unstable and without the code maintainer approval (to maintain future compatibility). Adding new stuff (new modules) should also be ok, but it would be preferable that these module come from either unstable or experimental. The mailing list for development should remain serdev. For users it could be a different one (if the changes or the mail traffic gets too big). As far as integrating new contributions faster in unstable, I agree that this is a problem, but I think we can sort it out. The experimental repository is a beginning and if there are better ideeas on how to speed things up while maintaing sanity, please speak. Maybe some kind of reviewed patches (e.g: someone sends a patch that adds a new feature in core, Bogdan reviews it decides it's ok and blesses it => I will integrate much faster something blessed by other ser developer). If you don't agree with my proposal, please say what you don't like and what you would like changed so that it would be acceptable for you. I'm sure that we can reach a compromise and hopefully improve ser development pace in the process. Andrei

6 10

[Serusers] How to config the ser?where can I find the instruction?
by Alan bus 21 Jun '05

21 Jun '05

I am new to using SER.I installed ser-0.8.14_src.tar.gz on Linux-redhat-2.4.18-14 with mysql-standard-5.0.6-beta-linux-i686.tar.gz.Mysql can run successfully.The following is what I have done:(the main directory:/usr/local/src/ser-0.8.14) step 1:make all step 2:ser_mysql.sh create The following are the commands and the error messages: [root@redhat scripts]# ser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': blrc.com ./ser_mysql.sh: line 1: gen_ha1: command not found HA1 calculation failed what's the matter?What should I do the next?Could you help me? Here are the ser.cfg: # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("iptel.org", "subscriber")) { www_challenge("iptel.org", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } thx! _________________________________________________________________ 享用世界上最大的电子邮件系统― MSN Hotmail。 http://www.hotmail.com

1 0

[Serusers] avp_pushto() + append_branch() not working
by Java Rockx 21 Jun '05

21 Jun '05

Hi All. I just need a sanity check regarding avp_pushto() for serial forking. Someone please correct me if I'm wrong here. When processing an INVITE message which results in a 486 Busy status my failure_route[] gets invoked. In the failure route I use avp_pushto() to load the next leg of the call in to the $ruri and then call append_branch() and lookup(location) prior to t_relay()ing the message. Using ngrep I do see the R-URI being changed to the value used in the avp_pushto() call, but the IP address that the INVITE is then sent to is the same as the IP address of the SIP router, which I cannot understand because I do call lookup("location") before t_relay() In otherwords, if my sip router, sip.mycompany.com<http://sip.mycompany.com>has an IP of 71.72.73.74 <http://71.72.73.74> and I use avp_pushto() to set the R-URI to 1001_at_sip.mycompany.com, lookup(location) does return TRUE, but ngrep still shows the INVITE leaving the SIP router and going to 71.72.73.74<http://71.72.73.74>(ie, itself) rather than the IP address in usrloc. Is there anything obvious that I'm not doing that would cause lookup(location) to return TRUE, but not change the R-URI accordingly? Regards, Paul

2 1

[Serusers] loading db during initialization
by Jack Wei 21 Jun '05

21 Jun '05

is there anyway to stop ser from loading users from db during startup? Jack Wei ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

2 1

[Serusers] voice sniffing
by Iqbal 21 Jun '05

21 Jun '05

Hi I have managed to get voice calls sniffed, by using a external sniffer. Does anyone know how I could bind a username/caller/caller tag to each wav file. I am not sure this is possible because the voip stream doesnt really have this info, but since my users are making calls by sending invites etc to ser, could I use ser to get the ID's and then call the sniffer to tag the names....or has anyone come across a really good voip sniffer Iqbal

2 1

[Serusers] Is there an AVP to alter "expires=" in REGISTER messages?
by Java Rockx 21 Jun '05

21 Jun '05

Hi All. We've got some ATAs that don't quite re-REGISTER in a timely fashion and therefore SER expires the usrloc record a few seconds before the ATA sends another REGISTER message. Is there an AVP or something to add, for example, 60 seconds to the "expires=" value in the REGISTER message prior to calling save()? By doing so, these "broken" ATAs would re-REGISTER before SER removes the usrloc record from cache. Regards, Paul

2 4

[Serusers] UAC module: multiple PSTN gateways
by Cameron Beattie 21 Jun '05

21 Jun '05

Is it possible to specify multiple authentication credentials within the UAC module? What I want to do is access multiple PSTN gateways that each require different credentials. The basic purpose of this is to allow least cost routing. Example Provider 1 offers cheap calling to +44 (UK) Provider 2 offers cheap calling to +1 (US) When user dials a 44 destination ser should relay to Provider 1 gateway, which will require authentication using Credential 1. When user dials a 1 destination ser should relay to Provider 2 gateway, which will require authentication using Credential 2. Any advice would be greatfully received. Regards Cameron

4 5

[Serusers] Using Sipsak to register
by Cameron Beattie 21 Jun '05

21 Jun '05

I am struggling to understand the syntax of sipsak. I want to use it to register against a proxy. I enter the command: sipsak -U -C sip:1234@192.168.0.5 -a password -s sip:1234@192.168.0.5 I get the message error: REGISTER Contact uri doesn't not contain sip:, *, or is not empty" If I enter the command sipsak -U -s sip:1234@192.168.0.5 -a password then the register message gets sent but with a contact of sip:1234@127.0.0.1 Could anyone help me with the correct syntax to use? Regards Cameron

3 2

[Serusers] voice messages in /tmp
by Llanos Serna García-Conde 20 Jun '05

20 Jun '05

2 1

[Serusers] SER + Prepaid = Old discussion
by Ricardo Poppi 20 Jun '05

20 Jun '05

Hi all. This list had a long discussion for the last two months about how to use SER with prepaid applications, since it does not participate into the media path. I´ll try to put things together in a easy way - at least for me... - for trying to solve some doubts about how this feature can work with SIP without creating new problems. And, of course, I´m assuming that we WILL NOT use b2bua, but try to make this work in a proxied environment. As mentioned before, one way to do so, is using the session timer (RFC-4028) definitions. The main reason those definitions were created, is for using when a BYE message don´t reach one of the UAs involved into a SIP dialog, that without those definitions would be virtualy connected "forever" since it did not receive any SIP signaling to disconect. With session timer working, even if a BYE message never reaches the other point, the maximun time that the "not-reached" UA would stay connected is the time setted into the session-timer parameter. Ok. But let´s use it to work with the prepaid environment. We implement a logic into our sip proxy (SER), that record-route all signaling messages between all UAs of our SIP network. Then we put our proxy to analize all re-invite messages that goes into our SIP dialogs and, if the customer credit (in seconds) is below that the time into the session-timer parameter, it rewrites the "seconds-credit" into the message session-timer position, decrements the last session time into the customer seconds credits table, and counts on the UAs to disconecting the call, since it - the proxy - will drop/block the next re-invite(s) to this especific dialog. In this aproach, we need to check the first invite as well, because if the customer credit is below that the value of the first/default session-timer, the SIP message needs to be rewriten too. Problems on this aproach: - If a re-invite from the caling user - the one that will be billed - never reachs the proxy, the UA will disconect the call sending a BYE, and, if the BYE never reachs the proxy either, the seconds of the last session won´t be billed, correct? - Thinking about security issues into a environment that just one UA support session-timers, It would be very easy for a malicious UA - the one that supports session-timers - when it runs out of credit, to send its last re-invite - the never answered one - withouting disconecting the media path. In this case, the proxy will "think" that the call is off, but it would be not true. Does anyone work with a true proxied prepaid environment using session timers? There is any aproach, other than using RFC-4028 - session timers, for doing this? Any clue will be welcome. Thanks in advance for the list. Regards, Ricardo Poppi

3 2

Re: [Serusers] SER + Prepaid = Old discussion
by Samuel Osorio Calvo 20 Jun '05

20 Jun '05

I don't think that session timers have the accuracy required for a billing system. If you take a look at the specification, minimum values are 90 seconds and recommended values are 30 minutes. If you reduce these values to bill every second a user is accessing the service, you could end up in interoperability problems. Moreover, any traversed proxy may use Min-SE header to increase the minimum value to few minutes, or hours, and then your user is gonna feel really happy having minutes for free... I really think the only reliable way is to use a b2bua because it is the only entity aware of the media path and able of dropping the call "instantaneously". I don't even think a statefull proxy will work since it is not in the media path and it is not supposed to initiate BYEs... I would suggest to choose between b2bua or flat fares. I think the latter suits more the IP world: almost everywhere you pay X per month for internet access, doesn't it apply better to VoIP?? interconnecting with PSTN is another topic and I think something like skype would suit better to IP than the classic X minutes->Y money. Samuel. Unclassified. >>> Ricardo Poppi <rpoppi77(a)giro.com.br> 06/20/05 06:11PM >>> Hi all. This list had a long discussion for the last two months about how to use SER with prepaid applications, since it does not participate into the media path. I ll try to put things together in a easy way - at least for me... - for trying to solve some doubts about how this feature can work with SIP without creating new problems. And, of course, I m assuming that we WILL NOT use b2bua, but try to make this work in a proxied environment. As mentioned before, one way to do so, is using the session timer (RFC-4028) definitions. The main reason those definitions were created, is for using when a BYE message don t reach one of the UAs involved into a SIP dialog, that without those definitions would be virtualy connected "forever" since it did not receive any SIP signaling to disconect. With session timer working, even if a BYE message never reaches the other point, the maximun time that the "not-reached" UA would stay connected is the time setted into the session-timer parameter. Ok. But let s use it to work with the prepaid environment. We implement a logic into our sip proxy (SER), that record-route all signaling messages between all UAs of our SIP network. Then we put our proxy to analize all re-invite messages that goes into our SIP dialogs and, if the customer credit (in seconds) is below that the time into the session-timer parameter, it rewrites the "seconds-credit" into the message session-timer position, decrements the last session time into the customer seconds credits table, and counts on the UAs to disconecting the call, since it - the proxy - will drop/block the next re-invite(s) to this especific dialog. In this aproach, we need to check the first invite as well, because if the customer credit is below that the value of the first/default session-timer, the SIP message needs to be rewriten too. Problems on this aproach: - If a re-invite from the caling user - the one that will be billed - never reachs the proxy, the UA will disconect the call sending a BYE, and, if the BYE never reachs the proxy either, the seconds of the last session won t be billed, correct? - Thinking about security issues into a environment that just one UA support session-timers, It would be very easy for a malicious UA - the one that supports session-timers - when it runs out of credit, to send its last re-invite - the never answered one - withouting disconecting the media path. In this case, the proxy will "think" that the call is off, but it would be not true. Does anyone work with a true proxied prepaid environment using session timers? There is any aproach, other than using RFC-4028 - session timers, for doing this? Any clue will be welcome. Thanks in advance for the list. Regards, Ricardo Poppi _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Users] RE: [Serusers] OpenSER release
by Giudice, Salvatore 20 Jun '05

20 Jun '05

I am not an advocate for either ser or openser, but I would like to comment. Is openser going to be equipped with a forum/ticket system where people can document bugs, feature requests, etc (non-configuration issues)? This is just my observation and you may not agree, but I believe this project could be much better maintained if it used a more structured ticketing style system to manage development issues instead of the current mailing lists. In my experience, mailing lists like this foster a terrible user experience where many development issues can go on without response. Ideally, if there was a mailing list to address user issues and ticketing system like the one Digium uses to manage Asterisk, I think everyone would benefit by being better informed and ser would ultimately be a better product for it. How many people out there feel that their issues have fallen through the cracks in the past couple years? -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: Wednesday, June 15, 2005 4:28 AM To: Andrei Pelinescu-Onciul Cc: SER developer mailing list; serusers; users(a)openser.org; devel(a)openser.org Subject: Re: [Serusers] OpenSER release On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: >On Jun 14, 2005 at 22:48, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > >[...] > > >>It is your opinion, but I repeat myself, that the SER code maintained by >>us will go further -- I don't think that someone can claim that we >>didn't do the job for our code (the only discrepancy is some last-minute >>adds in xlog (to print avps) - will be committed on unstable very soon >>with the new color patch). The cvs was created just to ease the >>maintainance. The patches would be a nightmare. >> >> > >Maybe I've misunderstood you: is this only a parallel "stabilized" >version + some features or is it a full fork (do you intend to fork >unstable also)? > > It is fork for the code that we changed (acc module, usrloc module ...), in the future may be other that they do not find the path in SER. We will maintain and upgrade our part of code from SER continuously. >I have no problem with another stable version, what worries me is >fragmenting the development for unstable (which is the place where major >changes are made). > > I see no fragmenting there -- the situation is the same for SER as it was before. For example, there is no fragment for acc module, it will be maintained by who did it till now, adding what he considers necessary there. But we came to meet a lot of requests of why the acc patch is not included in the CVS (it was fully backward compatible and had new features requested by many SER users) and we want to promote _more open_ approach to contributions to all parts of code. The acc patch was sent on November 1, 2004. No real response (neither negative, nor positive) from maintainer to the submission since then ... are you aware of a good reason?!?! ... should we wait just about (or more) half an year for each contribution?!? I will not do that anymore!!! Daniel > >Andrei > > > _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers

11 23

RE: [Serusers] double INVITE in a single call
by Giudice, Salvatore 20 Jun '05

20 Jun '05

Multiple aliases will fork the call in parallel. -----Original Message----- From: Jesus Rodriguez [mailto:jesusr@voztele.com] Sent: Monday, June 20, 2005 10:05 AM To: Nerea Munguira Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] double INVITE in a single call On Mon, 20 Jun 2005, Nerea Munguira wrote: Hola Nerea, > I have several calls that have two branches, one in one way and another in > the other way, I mean, for the same call ID, user 1 calls user 2, and also, > user 2 calls user 1. > > I wonder if this is normal, or if I have a something wrong. This only occur > sometimes, so it isn?t a routing problem. > > I have this problem in different scenarios, so it has to do with ser > configuration, not with the user routing scheme. Also, I have this problem > with calls between sip users or calls between sip user and PSTN. > > For the same call, I?ve got two logs: > > User 1 -> User 2 INVITE fromtag: id1 totag: id2 CALLID1 > User 2 -> User 1 INVITE fromtag: id2 totag: id1 CALLID1 > > Where id1 and id2 are the same identifier. What you see is possibly a re-INVITE because a codec or other SDP parameter negotiation. Saludos JesusR. ------------------------------- Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 ------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Users] sems module
by harry gaillac 20 Jun '05

20 Jun '05

Hello , I've heard that nathepler could provide MOH with rttproxy. I should try to use annoucement module. I want a dedicated directory to put .wav files My problem would be to write a boucle "while <condition> do <action> . In other words play file(1), file(2), ... file(3), file(1) ... I need help to start this boucle ! Is it a good idea ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 1

Re: [Serusers] How-to write a custom module
by Samuel Osorio Calvo 20 Jun '05

20 Jun '05

In fact there's the print module which is there just to give an example of a starting point for a new module......or is there anyone using print?? ;) Samuel. Unclassified. >>> Andreas Granig <andreas.granig(a)inode.info> 06/20/05 03:17PM >>> Steve Blair wrote: > I'd like to write a new module for SER. I see a number of other people > have asked this question but I don't see a how-to guide mentioned in > any of their messages. Does such a guide exist? If so can someone > provide a pointer to it? IMHO the "group" module is a good starting point because of it's simplicity.. Andy _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] Is there an AVP to alter "expires=" inREGISTERmessages?
by Samuel Osorio Calvo 20 Jun '05

20 Jun '05

I think that the UA should send the re-registration message before. I would try to push UAs developers to upgrade the firmware instead of adding more stuff to SER just to correct end points which are not compliant. Said that, since we are not in a perfect world, you might try something like (just until the ATA firmware is corrected, of course ;) ): save_noreply(); //modify Expires in reply sl_send_reply; I don't know exactly how to modify the Expires.....maybe something like subst textopts. Or remove the header and add a header to the reply (there was a function to do that but I don't remember the exact syntax or how it exactly works). I can not help you more than that....and I don't know for sure if it would work, just give it a try. Samuel Unclassified. >>> Java Rockx <javarockx(a)gmail.com> 06/20/05 02:21PM >>> Greger, That would be possible, but the way SER generates the 200OK reply, it would contain the altered "expires=" value. By doing so, the ATA would take that new value and use it, so if the ATA sent the REGISTER with an expires=300 and SER replied with a 200OK and a contact header containing expires=360, the ATA would not send another REGISTER for 360 seconds. I was fooling around with sip_msg.c and reply.c were I padded the expires with 60 seconds before saving to usrloc and then removing the pad before sending the reply and it works, but I would just feel better by not altering the source code. I guess I'd just feel more comfortable if SER had some sort of AVP or modparam for the registrar module that did this automagically. Regards, Paul On 6/20/05, Greger V. Teigre <greger(a)teigre.com> wrote: > > It was supposed to be short, but maybe too short. I meant doing the subst > on the REGISTER message before save. > g-) > ---- Original Message ---- > From: Greger V. Teigre > To: Java Rockx ; serusers > Sent: Monday, June 20, 2005 09:01 AM > Subject: Re: [Serusers] Is there an AVP to alter "expires=" in > REGISTERmessages? > > > Hi Paul, > > Would it be possible to "do it upside down" ? ;-) I.e. do a > > subst("/^Expires:(.*)[0-9]$/ Expires: yourvalue+60/gi"); > > g-) > > > > ---- Original Message ---- > > From: Java Rockx > > To: serusers > > Sent: Sunday, June 19, 2005 04:28 PM > > Subject: [Serusers] Is there an AVP to alter "expires=" in REGISTER > > messages? > > > >> Hi All. > >> > >> We've got some ATAs that don't quite re-REGISTER in a timely fashion > >> and therefore SER expires the usrloc record a few seconds before the > >> ATA sends another REGISTER message. > >> > >> Is there an AVP or something to add, for example, 60 seconds to the > >> "expires=" value in the REGISTER message prior to calling save()? By > >> doing so, these "broken" ATAs would re-REGISTER before SER removes > >> the usrloc record from cache. > >> > >> Regards, > >> Paul > >> > >> > >> > >> > >> _______________________________________________ > >> Serusers mailing list > >> serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] double INVITE in a single call
by Nerea Munguira 20 Jun '05

20 Jun '05

I have several calls that have two branches, one in one way and another in the other way, I mean, for the same call ID, user 1 calls user 2, and also, user 2 calls user 1. I wonder if this is normal, or if I have a something wrong. This only occur sometimes, so it isn´t a routing problem. I have this problem in different scenarios, so it has to do with ser configuration, not with the user routing scheme. Also, I have this problem with calls between sip users or calls between sip user and PSTN. For the same call, I´ve got two logs: User 1 -> User 2 INVITE fromtag: id1 totag: id2 CALLID1 User 2 -> User 1 INVITE fromtag: id2 totag: id1 CALLID1 Where id1 and id2 are the same identifier. Can anybody help me? thanks a lot!!

2 1

[Serusers] How-to write a custom module
by Steve Blair 20 Jun '05

20 Jun '05

Hello: I'd like to write a new module for SER. I see a number of other people have asked this question but I don't see a how-to guide mentioned in any of their messages. Does such a guide exist? If so can someone provide a pointer to it? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

3 2

[Serusers] How-to write a custom module
by Steve Blair 20 Jun '05

20 Jun '05

Hello: I'd like to write a new module for SER. I see a number of other people have asked this question but I don't see a how-to guide mentioned in any of their messages. Does such a guide exist? If so can someone provide a pointer to it? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

1 0

Re: Re: [Serusers] OpenSER release
by Max Malloy 20 Jun '05

20 Jun '05

2 1

[Serusers] help on mediaproxy with ser no sound
by peiyin 20 Jun '05

20 Jun '05

Hi all, I'm having trouble configuring mediaproxy to work on my SER server. One phone is behind NAT and the other is using a public ip. Both are registered in SER. Calls from phone A (nat) to phone B(public) are successful whereby phone B will rings.However, no sound comes out from both side, also, when i hang up phone B, phone A doesn't produce any engaged tone. Calls from phone B to phone A also successful but i don't get any audio on either end. I'm storing all information in database. My config is as below (taken from example and modified): route{ if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too many hops"); }; break; }; if (msg:len >= max_len) { if (method!="ACK") { sl_send_reply("513", "Message too big"); }; break; }; if(method=="INVITE" || method=="BYE") { setflag(1); #accounting } if (method=="REGISTER") { if (is_from_local()) { # Mark as NAT'ed if (client_nat_test("3")) { setflag(6); force_rport(); fix_contact(); }; if (!www_authorize("my_ser_ip", "subscriber")) { www_challenge("my_ser_ip", "0"); break; } else if (!check_to()) { sl_send_reply("403", "Username!=To not allowed"); break; }; if (!save("location")) { sl_reply_error(); }; } else { sl_send_reply("403", "This domain is not served here"); }; break; }; if (method=="INVITE") { if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; # Force subsequent messages to pass trough this proxy if (method == "INVITE") { record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; if (method=="INVITE") { t_on_reply("1"); }; if (is_uri_host_local()) { if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; }

2 1

[Serusers] (no subject)
by Chia Huey Lim 20 Jun '05

20 Jun '05

Hi. I am trying to configure the SER routing script for the NAT environment. Is there any function to replace IP in the Connection Information (c=) to its Public IP? I have tried using fix_nated_sdp("3") , but it is still giving the internal IP. Thanks a lot!

1 0

[Serusers] Is there a more detailed radius howto?
by Ryan Pagquil 20 Jun '05

20 Jun '05

Hi, I'm currently working on making SER to authenticate with freeradius, is there any other howto that is detailed enough to guide a newbie? Thanks a lot, Ryan

1 0

[Serusers] Asterisk+mediaproxy+loose_route
by Ah khng 20 Jun '05

20 Jun '05

Hi, I'm using the mediaproxy configuration example from onsip.org and asterisk as PSTN gateway. However, when loose routing, stream was pass through the mediaproxy which i want asterisk to handles NAT for the RTP packet itself. I have made an inelegant change to check for uri and source ip of the asterisk, but how to implement when LCR is being used? If i'm wrong please correct me. # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { #xlog("L_INFO", "loose_route(): Looking up %rm URI %ru from %is\n"); #lookup("location"); xlog("L_INFO", "loose_route(): t_relay() %rm to URI %ru\n"); if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); if (!(src_ip==A.B.C.D || uri=~"^sip:.*@A.B.C.D")) { #A.B.C.D => asterisk ip use_media_proxy(); xlog("L_INFO", "loose_route(): use mediaproxy for %rm to URI %ru\n"); }; }; }; route(1); break; }; cheer seong

1 0

[Serusers] Daniel's Commitment to Community?
by Peter L P 19 Jun '05

19 Jun '05

I am joining this discussion for the first time, so please appologize me if I provide you with novice's questions. I find the idea of SER great. As a matter of fact, I think that all software should be free and I am very thankful for people who advocate it, like Daniel orginally seemed to do. We, as a community, should fully support free software! Now what I am beginning to be concerned about is whether Daniel's team has actually the same goals! But he keeps the VNT and VPS products insufficiently transparent; I cannot download it from the Internet! How credible is this then??? The avpops patch has not commited immediatelly either! And scaringly, as Bogdan's email revealed, the commercial company voice-system has branched some other source code which is using some other forking code -- aren't they sucking serusers community to create a new branch under their control? I am concerned, because just giving away three or five modules is not much software for free and in turn for that we are being offered untransparent VNT and VPS. What assurance can we obtained we are not beeing fooled and this is not actually a red harring or even a mousetrap??? PS --- What I would also like to request is better documentation!!! __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] getting contact info from db
by Jack Wei 19 Jun '05

19 Jun '05

how do i look up a user from the database, get the contact info from the location table, rewrite the uri with the contact info? Jack Wei __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] rtpproxy
by Glenn Dalgliesh 19 Jun '05

19 Jun '05

I must be totally out of it. I am trying to download rtpproxy for the CVS head and I am not having at luck. Has something changed with regard to it's location? cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co rtpproxy ? rtpproxy/rtpproxy cvs server: existing repository /cvsroot/ser/PortaSIP/rtpproxy does not match /cvsroot/ser/rtpproxy cvs server: ignoring module rtpproxy [root@ser software]#

2 1

[Serusers] open letter: ser, tls, openser
by Peter Griffiths 19 Jun '05

19 Jun '05

folks -- i usually do not participate in mailing list discussions but it seems that my tls contribution became hot topic in recent ser vs. openser discussions so i think i will make an exception here. i think that openser is a bad idea. please stop using the tls code as a hostage in this dispute, i did not write it to fork the project. iptel guys made no attempt to hide the code when i approached them although they probably have their own commercial implementation. for me ser is an open and free project because: 1. i could download ser freely and use it 2. i needed tls, i was not able to buy it from iptel so i tried to implement it 3. i sent it to the mailing lists, anyone can find it there and use it if they want 4. someone else picked up the code, improved it and committed to cvs (thank you for this, by the way) what can be more open than this ? maybe sources stored in a wiki where anyone can change anything. openser claims that it will be more open, but: 1. over 20 people have access to ser cvs on berlios, but only 4 have access to openser cvs on sourceforge, not even other developers. 2. you took existing code from ser cvs, added your own improvements, but you did not give other developers who work on ser any chance to say whether such changes should be included or not, although they are still listed in AUTHORS file, in module documentation and elsewhere. from the user point of view they will be responsible for bugs introduced by you and have no chance to influence it. 3. i did not see any discussion about this move, it is a bit strange that the people on the lists were not allowed to participate in the decision, especially if you claim that it is for the interest of the community. how do you know people want the project to be split when you did not ask ? 4. the complete cvs history is missing in openser repository, effectively hiding who contributed what. I also noticed that user miconda created project named ser on sourceforge, is this a preparation for the next step - full control ? did other ser developers know about this ? i think having two projects with the same name on two sites is confusing. from my perspective the new fork is only an attempt to get more power and control over the sources. it is about who will control what, not about freedom. try to resolve it without forking the project. if ser is bad then openser is not any better: - iptel.org advertises iptelorg.com on its webpage - there is an ad of voice-system on the main page of openser.org, there are voice-system copyright statements and readme states that the project is maintained by voice-systems, proving beyond reasonable doubts that openser promotes voice-systems just like ser promotes iptel.org - there were complaints about maintainers deciding what comes into modules - so in openser only selected people have access to cvs and they will probably form a committee and decide. - there were calls for more discussions, but the fork was not discussed at all there probably is a good reason why tls was put in experimental directory first and is not in the main tree yet, i do not see this as a problem. in fact i would appreciate if someone who knows in detail how tcp code in ser works could review the code before it is committed in the main tree. there are many places in the code i don't understand. i did only limited testing because tls support in end devices is virtually non-existent. use it at your own risk. i don't know about others, but i am on alert when people talk about "the needs of the community" and "take over when someone is very busy or unwilling to cooperate", i don't remember seeing users asking for anything like this. -- peter __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] Failure reason.
by Dmitry Isakbayev 19 Jun '05

19 Jun '05

Hello, I am using serial forking, and I need to keep track of failed calls to t_relay. While in the 'failure_route[x]' block, can the script query why the previous call to t_relay failed (user not registered, connection failure, etc)? Thank You, Dmitry

2 1

Re: [Serusers] serweb PHP errors
by zhaomin 19 Jun '05

19 Jun '05

when I register a new account .but show me a error blow: "Sorry, there was an error when sending mail. Please try again later." Why? where can I configure mail server ? Zhaomin ----- Original Message ----- From: "Karel Kozlik" <karel(a)iptel.org> To: "PJR" <pjr_lists(a)mail.ntua.gr> Cc: <serusers(a)iptel.org> Sent: Wednesday, June 15, 2005 10:29 PM Subject: Re: [Serusers] serweb PHP errors > Hi, > try set: > > error_reporting = E_ALL & ~E_NOTICE > > in php.ini. > Regards Karel > > PJR napsal(a): > > On Fedora core 3 with all needed packages installed and config.php > > configured properly I can't login to serweb user_intrface or admin > > > > When I first load the user or admin index.php I get many errors in > > apache error logs like the following: > > > > [client 192.168.25.2] PHP Notice: Undefined index: extrahtml in /var/www/html/phplib/oohforms.inc on line 72, referer: http://192.168.25.3/iptel/user/ > > [client 192.168.25.2] PHP Notice: Undefined property: isfile in /var/www/html/phplib/oohforms.inc on line 250, referer: http://192.168.25.3/iptel/user/ > > [client 192.168.25.2] PHP Notice: Undefined property: hidden in /var/www/html/phplib/oohforms.inc on line 253, referer: http://192.168.25.3/iptel/user/ > > .....snip > > [client 192.168.25.2] PHP Notice: Undefined index: frozen in /var/www/html/phplib/oohforms.inc on line 284, referer: http://192.168.25.3/iptel/user/ > > > > When I enter username/passwd and submit I get more errors in logs > > (Notice, index frozen etc.) and also A Bad Username or Password message > > and I return to login screen. There is nothing wrong with passwords, and > > I can connect to ser db with mysql client just fine. > > > > I use serweb_2004-07-27 and ser-0.8.14. > > I have the vars in php.ini according to the instractions. > > > > Please point me to any possible solution. > > > > Thanks. > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)iptel.org > > http://mail.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Analog Gateway & SER
by Ozan Blotter 19 Jun '05

19 Jun '05

is there anyone have good experience with analog fxo gateways, because while more basic sip proxies can work with them i still do not understand why SER does not work with... because i'm neither an operator nor have a cisco gateway... please contact me, thanks ozan -- >>> please do not use >>> l.s.d(a)dnaofdesign.com >>> above account is stand by >>> use same account as >>> cosmocid(a)ispro.net.tr

3 2

[Serusers] ser monitoring by serweb php script
by Rosario Pingaro 18 Jun '05

18 Jun '05

I'd like to try the monitoring script by serweb but I am unable to get the tables updated. I followed the installation readme file updating my cronjob conf file, I have a working configuration of serweb so the connection to the db is fine. But I don't get any updated. Searching the var/log/messages I see the cronjob executed, I don't see any error. So I don't know where to look to get more info on my problem. Is there some one of you that uses this script? Thanks Rosario

2 2

Re: [Serusers] rtpproxy and serctl with problem
by Klaus Darilion 17 Jun '05

17 Jun '05

It looks like you are trying to start ser, although it is already running. regards, klaus PS: Please Cc: to the list. Bruno Oliveira wrote: > hi > > I try, but no resolv. Print this in term: > uiara:~# netstat -anp|grep 5060 > > "tcp 0 0 200.236.162.3:5060 0.0.0.0:* > OUÇA 2816/ser > tcp 0 0 192.168.1.2:5060 0.0.0.0:* > OUÇA 2816/ser > udp 0 0 200.236.162.3:5060 0.0.0.0:* > 2816/ser > udp 0 0 192.168.1.2:5060 0.0.0.0:* > 2816/ser" > > it continues with exactly error. > > thanks for your time > > On 6/17/05, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > >>Bruno Oliveira wrote: >> >>>hi >>> >>>i have rtpproxy runing and serctl. where it is file config of rtpproxy >>>? and when I type this command "serctl start" show me in >>>/var/log/syslog this message: >>> >>>"Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: Maxfwd module- initializing >>>Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: INFO: udp_init: >>>SO_RCVBUF is initially 111616 >>>Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: INFO: udp_init: >>>SO_RCVBUF is finally 223232 >>>Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: ERROR: tcp_init: >>>bind(6, 0x813753c, 16) on 192.168.1.2: Address already in use" >> >>Looks like another process already has bind to TCP port 5060. Maybe ser >>is already running? Or maybe a SIP client? >> >>Try >>#> netstat -anp|grep 5060 >> >>regards, >>klaus >> > > >

1 0

[Serusers] problems with ACKs and BYEs
by Jorge Crichigno 17 Jun '05

17 Jun '05

Hi all, I have a problem with in the following scenario: Lucent GW SER IPPhone IP: 1.2.3.4 IP:5.6.7.8 IP:8.9.5.5 INVITE--> <-- 100 INVITE--> <--100 <--180 <--180 <--200 <--200 ACK-->(Request-Uri = sip:5.6.7.8:5060) <-- PROBLEM HERE The problem arises only with the Lucent Gateway, because it sends the ACK with the Request-Uri as showed above (Request-Uri has the IP and port of the SER). I also found the same problem with BYEs. I attach the ser.cfg at this mail. I think this is a bug of Lucent, because I expected that Request-Uri of the ACKs should equal to the contact uri of the 200 message sent by the ipphone. The same for BYEs. Could some one clarify me that? Thanks in advance !!!

2 2

RE: [Users] extcmd removed?
by Matt Schulte 17 Jun '05

17 Jun '05

Yes, your right. my bad. -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Friday, June 17, 2005 1:08 PM To: Matt Schulte Cc: users(a)openser.org Subject: Re: [Users] extcmd removed? Hi Matt, there are two module which allows you to run external application : exec ext extcmd was a module developed exclusively for a FOKUS project - some interface (via udp) messaging for a JAVA application... are you sure you are not referring to "exec" as functionality?? regards, bogdan Matt Schulte wrote: >All, >I was just curious why ext/cmd were removed? > >The website says: >"extcmd - the functionality is not required anymore" > >I've ran into several cases where we require running external commands, >while I reason they're not the best idea and can cause memory >leaks/crashes, there is no alternativer other than writing a module. Of >which no one in our company can do so :-) > >ideas? Thanks. > >_______________________________________________ >Users mailing list >Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users > > >

1 0

[Users] extcmd removed?
by Matt Schulte 17 Jun '05

17 Jun '05

All, I was just curious why ext/cmd were removed? The website says: "extcmd - the functionality is not required anymore" I've ran into several cases where we require running external commands, while I reason they're not the best idea and can cause memory leaks/crashes, there is no alternativer other than writing a module. Of which no one in our company can do so :-) ideas? Thanks.

3 2

RE: [Users] extcmd removed?
by Matt Schulte 17 Jun '05

17 Jun '05

We did, it f'd up in the long run. -----Original Message----- From: Juha Heinanen [mailto:jh@tutpro.com] Sent: Friday, June 17, 2005 1:10 PM To: Matt Schulte Cc: users(a)openser.org Subject: [Users] extcmd removed? Matt Schulte writes: > leaks/crashes, there is no alternativer other than writing a module. Of > which no one in our company can do so :-) hire someone from outside to write your module. -- juha

1 0

[Serusers] OpenSER release
by Max Malloy 17 Jun '05

17 Jun '05

2 1

[Serusers] second mediaproxy on another network, not working
by Rosario Pingaro 17 Jun '05

17 Jun '05

I have successfully installed ser and mediaproxy on my network, it is a /27. Now to load balance I installed a second mediaproxy server on a datacenter, so on another network. I configure dns and the mediaproxy.ini (on the new server) to work with the new ip address. I disbaled all firewall rules to test the proxy. But it didn't work, the call is established but no rtp flow between the ua and mediaproxy. This is the debug from the dispacher: Jun 17 18:10:32 voip proxydispatcher[26275]: command request 493086ee-17262f1a(a)10.254.0.42 10.254.0.42:16440:audio 213.178.216.33 voadsl.italycomnet.it local 213.178.208.130 remote Sipura/SPA2100-3.1.3 info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag: Jun 17 18:10:32 voip proxydispatcher[26275]: forwarding to mediaproxy on voicegw-mi.italycomnet.it:25060: got: '213.178.208.130 35006' Jun 17 18:10:32 voip proxydispatcher[26275]: command execution time: 289.48 ms Jun 17 18:10:32 voip proxydispatcher[26275]: command lookup 493086ee-17262f1a(a)10.254.0.42 213.178.208.130:13308:audio 213.178.208.130 voadsl.italycomnet.it local voadsl.italycomnet.it unknown Asterisk=20Italycom=20Mix info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:as58d31ab2 Jun 17 18:10:33 voip proxydispatcher[26275]: forwarding to mediaproxy on voicegw-mi.italycomnet.it:25060: got: '213.178.208.130 35006' Jun 17 18:10:33 voip proxydispatcher[26275]: command execution time: 183.21 ms Jun 17 18:10:40 voip proxydispatcher[26275]: command lookup 493086ee-17262f1a(a)10.254.0.42 213.178.208.130:13308:audio 213.178.208.130 voadsl.italycomnet.it local voadsl.italycomnet.it unknown Asterisk=20Italycom=20Mix info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:as58d31ab2 Jun 17 18:10:41 voip proxydispatcher[26275]: forwarding to mediaproxy on voicegw-mi.italycomnet.it:25060: got: '213.178.208.130 35006' Jun 17 18:10:41 voip proxydispatcher[26275]: command execution time: 175.07 ms Jun 17 18:10:46 voip proxydispatcher[26275]: command delete 493086ee-17262f1a(a)10.254.0.42 info= Jun 17 18:10:46 voip proxydispatcher[26275]: forwarding to mediaproxy on voicegw-mi.italycomnet.it:25060: got: '' Jun 17 18:10:46 voip proxydispatcher[26275]: command execution time: 191.27 ms this is the debug from the mediaproxy server: Jun 17 18:10:32 voicegw-mi mediaproxy[4213]: command request 493086ee-17262f1a(a)10.254.0.42 10.254.0.42:16440:audio 213.178.216.33 voadsl.italycomnet.it local 213.178.208.130 remote Sipura/SPA2100-3.1.3 info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:,dispatcher Jun 17 18:10:32 voicegw-mi mediaproxy[4213]: session 493086ee-17262f1a(a)10.254.0.42: started. listening on 213.178.208.130:35006 Jun 17 18:10:32 voicegw-mi mediaproxy[4213]: command execution time: 1.61 ms Jun 17 18:10:33 voicegw-mi mediaproxy[4213]: command lookup 493086ee-17262f1a(a)10.254.0.42 213.178.208.130:13308:audio 213.178.208.130 voadsl.italycomnet.it local voadsl.italycomnet.it unknown Asterisk=20Italycom=20Mix info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:as58d31ab2,dispatcher Jun 17 18:10:33 voicegw-mi mediaproxy[4213]: command execution time: 0.56 ms Jun 17 18:10:33 voicegw-mi mediaproxy[4213]: session 493086ee-17262f1a(a)10.254.0.42: called signed in from 213.178.208.130:13308 (RTP) (will return to 213.178.208.130:13308) Jun 17 18:10:33 voicegw-mi mediaproxy[4213]: warning: Received packet from a third party: 213.178.216.2:16440 Jun 17 18:10:40 voicegw-mi mediaproxy[4213]: command lookup 493086ee-17262f1a(a)10.254.0.42 213.178.208.130:13308:audio 213.178.208.130 voadsl.italycomnet.it local voadsl.italycomnet.it unknown Asterisk=20Italycom=20Mix info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:as58d31ab2,dispatcher Jun 17 18:10:40 voicegw-mi mediaproxy[4213]: command execution time: 0.18 ms Jun 17 18:10:46 voicegw-mi mediaproxy[4213]: command delete 493086ee-17262f1a(a)10.254.0.42 info=dispatcher Jun 17 18:10:46 voicegw-mi mediaproxy[4213]: session 493086ee-17262f1a(a)10.254.0.42: 0/646/0 packets, 0/38760/0 bytes (caller/called/relayed) Jun 17 18:10:46 voicegw-mi mediaproxy[4213]: session 493086ee-17262f1a(a)10.254.0.42: ended. Jun 17 18:10:46 voicegw-mi mediaproxy[4213]: command execution time: 0.42 ms Jun 17 18:11:52 voicegw-mi sshd(pam_unix)[4228]: session opened for user root by root(uid=0) Seems that is not able to find the right pubblic address of the ua.... Any help is appreciated. Thanks Rosario

2 1

[Users] TLS support
by Bogdan-Andrei Iancu 17 Jun '05

17 Jun '05

Hi folks, thanks to Peter and Cesc we have TLS support buit-in OpenSER. The implementation is only on CVS head (development branch). To getting do: - if you already have a local checkout of CVS head: cvs update -dP - no local checkout of CVS head: cvs -d:pserver:anonymous@cvs.sourceforge.net:/cvsroot/openser login cvs -z3 -d:pserver:anonymous@cvs.sourceforge.net:/cvsroot/openser co -P sip-server Full documentation about how to compile and use it (script example included) can be found in the documentation repository: http://openser.org/docs/tls.html The next step will be to convince as many people as possible to use it and to test it for eventual bugs/problems. I will be very helpful in moving towards a stable version. regards, Bogdan

1 0

[Serusers] 30s calls
by Jerlique Ban 17 Jun '05

17 Jun '05

Hello, I have configured SER and asterisk to allow me to make calls to the PSTN network, however on my voip phone (behind NAT) I am having issues with the voip tx audio dropping out after 30 seconds. Now I'm guessing it's a nat issue but even that doesn't really make sense!! Why, well because the only the TX of the voip phone drops out (ie the PSTN phone cannot hear what is said on the voip phone). The PSTN phone can still transmit audio to the voip phone (through the nat). Anyway in SER, I have set the natping_interval to 5 seconds, and this still doesn't resolve the issue. Strangely at the time that the audio disconnects Asterisk is sending my phone an INVITE message. Why would it do this mid call? I'm using SER0.9.0+Asterisk as my platform. Any pointers?? JB

2 1

[Serusers] avaya
by Iqbal 17 Jun '05

17 Jun '05

Hi has anyone worked with avaya, and linked it to ser, if so are there any obstacles or is it pretty straightforward Iqbal

1 0

[Serusers] free SER
by A C 17 Jun '05

17 Jun '05

Dear Bogdan, Dear Daniel, As a user of SER for number of years now I really appreciate your work and effort to liberate SER from the clutches of iptelorg with its microsoft like mentality. I am sure it was not easy for you to stand-up against those who have paid your salary for a number of years and have given you the knowledge and software with which you have built your own company. I also agree with you that the Fraunhofer as the maintainer of the iptel.org and berlios sites has in no way the right to refer on these sites to a spinn-off in which it is involved in -having invested a huge sum in this project including your salary in the last years is surely no excuse. I am also of your opinion that the benefits of the users should be the utmost goal of all who are involved in SER. Therefore my question is: will the next release of openSER also include the VNT and VPS products your company are offering? Providing these components to the community will surely help the SER users to offer better VoIP services -which as you have stressed should be our major goal. I would recommend to publish your products under the BSD license so that we could include them into our own products as well? We are building an NGN product that would benefit greatly from your components. iptelorg has actually asked us to pay for their source code. Those guys have surely not realized yet that SER does not belong to them but to the users and community. Best regards -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

1 0

[Serusers] ONsip.org poll announcement regarding SER/OpenSER
by Greger V. Teigre 17 Jun '05

17 Jun '05

I believe the community should be allowed to tell the core developers their opinion and I have thus created a poll at http://onsip.org/ (http://www.onsip.org/modules/xoopspoll/index.php?poll_id=5) I apologize that you will have to log in (or register if you have not yet done so) in order to vote. This is how Xoops (the content management system) avoids voting manipulation through (otherwise allowed) multiple votes from one user. Below are the statements that you can agree to or not. Please send me a private note if you have suggestions for other statements that will be informative. Regards, Greger ------------------------------------------------------------------------------------- What is your reaction to the OpenSER announcement? Overall, a separate OpenSER will benefit the users Overall, a separate OpenSER will not benefit the users I will go for the "old" SER I will go for the new OpenSER The conflict should be resolved into one project Splitting in two is a good thing

1 0

Re: [Users] new version openser how to deal with call a SIP uri with multi registar
by jimmy way 17 Jun '05

17 Jun '05

Hi bogdan, Need I write script for failure_route and onreply_route in ser.cfg to support parallel fork or I only lets openser to do that by itself. If need, how to write the script? By the way, how to support serial fork? Regards, Jimway --- Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > Hi Jimmy. > > see inline comments. As short note, what you are > describing in parallel > forking and it behaves exactly as you thing (see RFC > 3261). > > regards, > bogdan > > jimmy way wrote: > > >Hi all, > > > > The question is: > > 1. a user A registar from multi address, like > > UA1 registrar from address1 > > UA2 registrar from address2 > > UA3 registrar from address3 > > and so on > > so in db "location" there are multi address > in > >the list about user A > > 2. if set append_branch as on in registrar > module, > >when user B invite user A, Openser server will > append > >all these address in the invite packet. So all > these > >user will get the call. normally all UAs ringing, > then > > 3. if one of these UAs accept the call, how > about > >other UAs, will they stop ringing? > > > > > yes, the other pending branches will be > automatically cancelled. > > > 4. if one of these UAs deny(40x) the call, how > >about other UAs, will the still ringing? > > > > > yes - there are two way for a parallel forked call > to end: either one of > the branches picks up (2xx reply), either all of > them send negative > replies (>=300). > > > 5. if one of these UAs not connected, how about > >caller the callee do? > > > > > if one of the branches is not connected (but still > registered), an > internal timeout (408) will be generated for it. > > > > >tks. > >Jimway > > > > > > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

3 2

[Serusers] test rtpproxy
by Bruno Oliveira 17 Jun '05

17 Jun '05

hi. as I make test connection beetwen kphone ( sip client ) and me rtpproxy. thanks for your time. -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

1 0

Re: [Serusers] rtpproxy and serctl with problem
by Bruno Oliveira 17 Jun '05

17 Jun '05

hi I try, but no resolv. Print this in term: "uiara:~# netstat -anp|grep 5060 tcp 0 0 200.236.162.3:5060 0.0.0.0:* OUÇA 2816/ser tcp 0 0 192.168.1.2:5060 0.0.0.0:* OUÇA 2816/ser udp 0 0 200.236.162.3:5060 0.0.0.0:* 2816/ser udp 0 0 192.168.1.2:5060 0.0.0.0:* 2816/ser" it continues with exactly error. thanks for your time > On 6/17/05, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > > Bruno Oliveira wrote: > > > hi > > > > > > i have rtpproxy runing and serctl. where it is file config of rtpproxy > > > ? and when I type this command "serctl start" show me in > > > /var/log/syslog this message: > > > > > > "Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: Maxfwd module- initializing > > > Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: INFO: udp_init: > > > SO_RCVBUF is initially 111616 > > > Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: INFO: udp_init: > > > SO_RCVBUF is finally 223232 > > > Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: ERROR: tcp_init: > > > bind(6, 0x813753c, 16) on 192.168.1.2: Address already in use" > > > > Looks like another process already has bind to TCP port 5060. Maybe ser > > is already running? Or maybe a SIP client? > > > > Try > > #> netstat -anp|grep 5060 > > > > regards, > > klaus > > > > > -- > []s, > Bruno "Niggas" Oliveira > Belo Horizonte - MG > Msn: n1gg4s(a)gmail.com > Icq: 176314647 > > "Todo o nosso descontentamento por aquilo > que nos falta procede da nossa falta de > gratidão por aquilo que temos." > -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

1 0

[Serusers] SER does not forward ACK
by jero 17 Jun '05

17 Jun '05

I've a problem with SER forwarding ACK's when a call has stablished. I only need to make outbound call from sip-phone, every call has to be redirected to AS5300. These are the packets sended: SIP-PHONE -----------> SER -------------> AS5300 ------>PSTN _________________________________________________ |-----invite------->| | |<----trying--------| | | |-----invite------->| | |<----trying--------| | |<----ringing-------| |<----ringing-------| | | |<------ok----------| |<------ok----------| | |-------*ack*------>| | <--here is the problem | | | | |<------ok----------| |<------ok----------| | | |<------ok----------| |<------ok----------| | | | timeout As seen, the ack looses in SER and is not sended to AS5300, the call is stablished, but after 20 seconds is dropped by as5300 because of the timeout. I've read the messages on this forum about this, and it seems to be a misconfiguration problem, but i didn't find the solution. Here is mi ser.cfg and a more detailed trace of the packets. SIP-PHONE - 81.37.111.201 SER - 196.220.57.58 AS5300 - 196.220.57.51 # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "log_flag", 1 ) modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) # ------------------------- request routing logic ------------------- # main routing logic alias="196.220.57.58" route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # Lots deleted, such as processing registrations # we record-route all messages -- to make sure that # subsequent messages will go through our proxy. if(method=="INVITE"){ record_route(); } # loose-route processing if (loose_route()) { t_relay(); break; }; if(uri==myself){ if(method=="REGISTER"){ save("location"); break; } } # IF HOST IS JUST NUMBERS SEND TO route(1) if (uri =~ "^sip:[0-9]*@"){ log(1, "SER: matched ^sip:[0-9]*@ - just numbers - process in route(3)\n"); route(1); break; }; } route[1] { rewritehostport("196.220.57.51:5060"); setflag(1); t_relay(); break; } ########################################################################## Packet trace ########################################################################## No. Time Source Destination Protocol Info 5053 17.486988 81.37.111.201 196.220.57.58 SIP/SDP Request: INVITE sip:607628000@196.220.57.58, with session description Internet Protocol, Src Addr: 81.37.111.201 (81.37.111.201), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:607628000@196.220.57.58 SIP/2.0 Message Header From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58> Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-3e951-f479ae6-2caf Max-Forwards: 70 Supported: replaces User-Agent: FXS_GW (4asipfxs.108a) Contact: <sip:777910020@81.37.111.201:5060> Content-Type: application/SDP Content-Length: 286 Message body No. Time Source Destination Protocol Info 5054 17.491678 196.220.57.58 81.37.111.201 SIP Status: 100 trying -- your call is important to us Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Message Header From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58> Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-3e951-f479ae6-2caf;received=81.37.111.201 Server: Sip EXpress router (0.9.0 (i386/linux)) Content-Length: 0 Warning: 392 196.220.57.58:5060 "Noisy feedback tells: pid=1024 req_src_ip=81.37.111.201 req_src_port=5060 in_uri=sip:607628000@196.220.57.58 out_uri=sip:607628000@196.220.57.51:5060 via_cnt==1" No. Time Source Destination Protocol Info 5058 17.494454 196.220.57.58 196.220.57.51 SIP/SDP Request: INVITE sip:607628000@196.220.57.51:5060, with session description Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 196.220.57.51 (196.220.57.51) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:607628000@196.220.57.51:5060 SIP/2.0 Message Header Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58> Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 1 INVITE Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0 Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf Max-Forwards: 16 Supported: replaces User-Agent: FXS_GW (4asipfxs.108a) Contact: <sip:777910020@81.37.111.201:5060> Content-Type: application/SDP Content-Length: 286 Message body No. Time Source Destination Protocol Info 5063 17.508105 196.220.57.51 196.220.57.58 SIP Status: 100 Trying Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Content-Length: 0 No. Time Source Destination Protocol Info 8037 28.278422 196.220.57.51 196.220.57.58 SIP Status: 180 Ringing Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Length: 0 No. Time Source Destination Protocol Info 8038 28.279923 196.220.57.58 81.37.111.201 SIP Status: 180 Ringing Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Length: 0 No. Time Source Destination Protocol Info 9791 33.535713 196.220.57.51 196.220.57.58 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 9794 33.537703 196.220.57.58 81.37.111.201 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 9956 34.033762 196.220.57.51 196.220.57.58 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 9957 34.035211 196.220.57.58 81.37.111.201 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 9958 34.045266 81.37.111.201 196.220.57.58 SIP Request: ACK sip:607628000@196.220.57.51:5060 Internet Protocol, Src Addr: 81.37.111.201 (81.37.111.201), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:607628000@196.220.57.51:5060 SIP/2.0 Message Header From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 1 ACK Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-3e962-f47db5a-57d8 Max-Forwards: 70 User-Agent: FXS_GW (4asipfxs.108a) Contact: <sip:777910020@81.37.111.201:5060> Route: <sip:196.220.57.58;lr;ftag=c0a80129-13c4-3e951-f479ae6-6c8e> Content-Length: 0 No. Time Source Destination Protocol Info 10269 35.033720 196.220.57.51 196.220.57.58 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 10271 35.035346 196.220.57.58 81.37.111.201 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 10791 36.690602 196.220.57.51 81.37.111.201 RTCP Sender Report Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 19343 (19343), Dst Port: 16385 (16385) Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Reception report count: 1 Packet type: Sender Report (200) Length: 12 Sender SSRC: 459946035 Timestamp, MSW: 3327900282 Timestamp, LSW: 356227917 RTP timestamp: 0 Sender's packet count: 52 Sender's octet count: 3011 Source 1 Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Source count: 1 Packet type: Source description (202) Length: 19 Chunk 1, SSRC/CSRC 459946035 No. Time Source Destination Protocol Info 10894 37.033690 196.220.57.51 196.220.57.58 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 10895 37.035934 196.220.57.58 81.37.111.201 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 11726 40.222188 196.220.57.51 81.37.111.201 RTCP Sender Report Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 19343 (19343), Dst Port: 16385 (16385) Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Reception report count: 1 Packet type: Sender Report (200) Length: 12 Sender SSRC: 459946035 Timestamp, MSW: 3327900285 Timestamp, LSW: 2639679338 RTP timestamp: 0 Sender's packet count: 102 Sender's octet count: 5715 Source 1 Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Source count: 1 Packet type: Source description (202) Length: 19 Chunk 1, SSRC/CSRC 459946035 No. Time Source Destination Protocol Info 12971 44.392482 196.220.57.51 81.37.111.201 RTCP Sender Report Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 19343 (19343), Dst Port: 16385 (16385) Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Reception report count: 1 Packet type: Sender Report (200) Length: 12 Sender SSRC: 459946035 Timestamp, MSW: 3327900289 Timestamp, LSW: 3371471883 RTP timestamp: 0 Sender's packet count: 108 Sender's octet count: 6075 Source 1 Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Source count: 1 Packet type: Source description (202) Length: 19 Chunk 1, SSRC/CSRC 459946035 No. Time Source Destination Protocol Info 13187 45.033530 196.220.57.51 196.220.57.58 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK3e84.b6c26904.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 13188 45.034943 196.220.57.58 81.37.111.201 SIP/SDP Status: 200 OK, with session description Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e951-f479ae6-2caf From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Date: Thu, 16 Jun 2005 08:44:22 GMT Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:607628000@196.220.57.51:5060> Record-Route: <sip:196.220.57.58;ftag=c0a80129-13c4-3e951-f479ae6-6c8e;lr=on> Content-Type: application/sdp Content-Length: 229 Message body No. Time Source Destination Protocol Info 14829 50.802294 196.220.57.51 81.37.111.201 RTCP Sender Report Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 19343 (19343), Dst Port: 16385 (16385) Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Reception report count: 1 Packet type: Sender Report (200) Length: 12 Sender SSRC: 459946035 Timestamp, MSW: 3327900296 Timestamp, LSW: 837215452 RTP timestamp: 0 Sender's packet count: 203 Sender's octet count: 11330 Source 1 Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Source count: 1 Packet type: Source description (202) Length: 19 Chunk 1, SSRC/CSRC 459946035 No. Time Source Destination Protocol Info 15481 53.037113 196.220.57.51 81.37.111.201 RTCP Receiver Report Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 19343 (19343), Dst Port: 16385 (16385) Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Reception report count: 1 Packet type: Receiver Report (201) Length: 7 Sender SSRC: 459946035 Source 1 Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Source count: 1 Packet type: Source description (202) Length: 7 Chunk 1, SSRC/CSRC 459946035 Real-time Transport Control Protocol Stream setup by SDP (frame 5058) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Source count: 1 Packet type: Goodbye (203) Length: 1 Identifier: 459946035 No. Time Source Destination Protocol Info 17285 59.087291 81.37.111.201 196.220.57.58 SIP Request: BYE sip:607628000@196.220.57.51:5060 Internet Protocol, Src Addr: 81.37.111.201 (81.37.111.201), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:607628000@196.220.57.51:5060 SIP/2.0 Message Header From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 2 BYE Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-3e97b-f483cd0-e16 Max-Forwards: 70 Supported: replaces User-Agent: FXS_GW (4asipfxs.108a) Route: <sip:196.220.57.58;lr;ftag=c0a80129-13c4-3e951-f479ae6-6c8e> Content-Length: 0 No. Time Source Destination Protocol Info 17286 59.090474 196.220.57.58 196.220.57.51 SIP Request: BYE sip:607628000@196.220.57.51:5060 Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 196.220.57.51 (196.220.57.51) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:607628000@196.220.57.51:5060 SIP/2.0 Message Header From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 2 BYE Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK0e84.aed52364.0 Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e97b-f483cd0-e16 Max-Forwards: 16 Supported: replaces User-Agent: FXS_GW (4asipfxs.108a) Content-Length: 0 No. Time Source Destination Protocol Info 17289 59.094579 196.220.57.51 196.220.57.58 SIP Status: 481 Call Leg/Transaction Does Not Exist Internet Protocol, Src Addr: 196.220.57.51 (196.220.57.51), Dst Addr: 196.220.57.58 (196.220.57.58) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 481 Call Leg/Transaction Does Not Exist Message Header Via: SIP/2.0/UDP 196.220.57.58;branch=z9hG4bK0e84.aed52364.0,SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e97b-f483cd0-e16 From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 2 BYE Content-Length: 0 No. Time Source Destination Protocol Info 17290 59.096162 196.220.57.58 81.37.111.201 SIP Status: 481 Call Leg/Transaction Does Not Exist Internet Protocol, Src Addr: 196.220.57.58 (196.220.57.58), Dst Addr: 81.37.111.201 (81.37.111.201) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 481 Call Leg/Transaction Does Not Exist Message Header Via: SIP/2.0/UDP 192.168.1.41:5060;received=81.37.111.201;branch=z9hG4bK-3e97b-f483cd0-e16 From: "777910020"<sip:777910020@196.220.57.58>;tag=c0a80129-13c4-3e951-f479ae6-6c8e To: <sip:607628000@196.220.57.58>;tag=2A3EE1E8-F7B Call-ID: 569ecc-c0a80129-13c4-3e951-f479ae1-4390(a)196.220.57.58 CSeq: 2 BYE Content-Length: 0 ############################################################################## One more thing, when looking at ser debug, after receiving the ACK from SIP-PHONE a line shows the following message: DEBUG : sl_filter_ACK: to late to be a local ACK! Please any help will be appreciated. Thanks.

3 4

[Users] cvs tags and branches
by Klaus Darilion 17 Jun '05

17 Jun '05

Hi! Can someone please describe the CVS tags/branches and their relation with ser? What is the difference between MAIN and HEAD? thanks, klaus

2 3

[Serusers] freeware MCU
by Rodríguez García, Marcos 17 Jun '05

17 Jun '05

Hi, Any one now any free software that makes the functions o a MCU for voice/video multiconferences with SIP? Thanks, Marcos

1 0

[Users] new version openser how to deal with call a SIP uri with multi registar
by jimmy way 17 Jun '05

17 Jun '05

Hi all, The question is: 1. a user A registar from multi address, like UA1 registrar from address1 UA2 registrar from address2 UA3 registrar from address3 and so on so in db "location" there are multi address in the list about user A 2. if set append_branch as on in registrar module, when user B invite user A, Openser server will append all these address in the invite packet. So all these user will get the call. normally all UAs ringing, then 3. if one of these UAs accept the call, how about other UAs, will they stop ringing? 4. if one of these UAs deny(40x) the call, how about other UAs, will the still ringing? 5. if one of these UAs not connected, how about caller the callee do? tks. Jimway ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

2 1

[Users] mailing list archive does not work
by Klaus Darilion 17 Jun '05

17 Jun '05

regards, klaus

3 2

RE: [Serusers] pa cannot work as a B2BUA?
by Simith Nambiar 17 Jun '05

17 Jun '05

Hi mma, I use windows mesenger 4.6 , it has the Expires header ! ================================================================ SUBSCRIBE sip:pramukh@10.10.100.44 SIP/2.0 Via: SIP/2.0/UDP 10.10.100.242:8385 Max-Forwards: 70 From: "amith_nambiar" <sip:amith_nambiar@10.10.100.44>;tag=bdc8933dec424822be3b3d671f3660a6;epid=d 9d835a5ca To: <sip:pramukh@10.10.100.44> Call-ID: c454b0c6f13f46cc90d45be8f968d1d3(a)10.10.100.242 CSeq: 2 SUBSCRIBE Contact: <sip:10.10.100.242:8385> User-Agent: RTC/1.2 Expires: 0 Event: presence Content-Length: 0 ================================================================ I get a parse error in the function handle_subscription ! I'am trying to debug why it is giving me that error, using a debugger to step thru code, have found a problem, but i'am not 100 % sure! cheers. Simith -----Original Message----- From: mma [mailto:mmouhiha@softiphone.net] Sent: Thursday, June 16, 2005 3:46 PM To: Simith Nambiar; 'Jamey Hicks' Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] pa cannot work as a B2BUA? Hi I think the pb is that the SUBSCRIBE requests that sends Windows Messenger does not contain an "Expires" header as defined in RFC3265 ftp://ftp.rfc-editor.org/in-notes/rfc3265.txt Regards ----- Original Message ----- From: "Simith Nambiar" <simith(a)indtelesoft.com> To: "'Jamey Hicks'" <jamey.hicks(a)hp.com>; "mma" <mmouhiha(a)softiphone.net> Cc: <serusers(a)lists.iptel.org> Sent: Tuesday, June 14, 2005 6:14 AM Subject: RE: [Serusers] pa cannot work as a B2BUA? > *********************** > Pas de virus detecté dans l'attachement no filename > > Votre Mail a été vérifié par InterScan. > ***********-*********** > > > > -----Original Message----- > From: Jamey Hicks [mailto:jamey.hicks@hp.com] > Sent: Monday, June 13, 2005 11:23 PM > To: mma > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] pa cannot work as a B2BUA? > > > mma wrote: > > > hi, > > > > i read in pa module that it cannot subsribe to presence of users > > using SIP, > > so how can i do to manage the presence of my contact list > > > PA still does not have support to SUBSCRIBE to contacts. What you can > do is set up PA so that it only handles presence for users who register > against and publish presence information to your SER server, and to > forward subscribe requests belonging to other registrars or UAs. > > > i added pa module to SER and a had this errors in response to a > > SUBSCRIBE request > > > > parse_hfs(): Error while parsing headers > > > I would have to see the message to be able to tell what is wrong. Which > UA was generating the subscription? Which version of SER? > > -Jamey > > Hi Jamey, > I have a similar Error , when using MSN Messenger Ver 4.6, and > Ser-0.9.0 , > Please let me know if you have any hints to look for problems ! > > Cheers. > Simith > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > ______________________________________________________________________ > This email has been scanned via a message filtering facility. > If you have any issues with your email, please contact the IT Dept. > ______________________________________________________________________ This email has been scanned via a message filtering facility. If you have any issues with your email, please contact the IT Dept.

1 0

[Serusers] Problem with cancel a invite
by Anders Hansson 17 Jun '05

17 Jun '05

Hi all, I´ve got a problem when trying to cancel an invite. The situation: A makes a call to B (INVITE). B press the "hang up" button. An here comes the problem. From As side, everything is OK, but from Bs side, the phone is still ringing... How can a solve this problem? Anyone? Thanks, Anders

5 8

[Serusers] MediaProxy No Sound
by TCCHAN 17 Jun '05

17 Jun '05

Dear All, I found out that the following: (1) if I comment out the use_media_proxy in the on_reply_route, I got half of the stream working, ie. the called party could hear the caller party. Sessions will dump a seesion as before. (2) if I comment out the use_media_proxy in the if (method=="INVITE") in main route, I got the whole sessions working, ie called party can hear caller and vice versa. But sessions didn't get a session dumped, even mediaproxy don't show any activities, what happened? ( my device are behind NAT, how could they hear if mediaproxy is not working, and if mediaproxy is working, how come there is not print out on mediaproxy and sessions show No RTP sessions) (3) If I use use_media_proxy on both the main route and the on_reply_route (which is as shown on the example ser.cfg in the mediaproxy module) I 've got nothing, both the caller and called party can't hear anything. But seesions shown a session. If seems that I can't call use_media_proxy twice or something like that, but I really not quite sure. Any suggestion? Thank for any help available. Regards, TC Chan On Thu, 2005-06-16 at 12:22 +0200, harry gaillac wrote: > Hello, > > Look at your onreply_route in ser.cfg . > > Harry > > > #debug=3 #fork=no #log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" #-------------------------------------------------------------------------------- #TCN: Need this line for ver 0.9.0 fifo_db_url="mysql://ser:heslo@localhost/ser" #-------------------------------------------------------------------------------- # ------------------ module loading ---------------------------------- #loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/textops.so" #loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" #---------------------------------------------------------------------- # mediaproxy modparam("mediaproxy", "natping_interval", 60) #modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("registrar", "nat_flag", 2) #---------------------------------------------------------------------- #---------------------------------------------------------------------- # uri #modparam("uri", "db_url", "sql://root:heslo@localhost/ser") #modparam("uri", "uri_table", "subscriber") #---------------------------------------------------------------------- #---------------------------------------------------------------------- # TCN: no forking #modparam("registrar","desc_time_order",1) modparam("usrloc","desc_time_order",1) modparam("registrar","append_branches",0) #----------------------------------------------------------------------- # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 1) #modparam("usrloc", "db_mode", 2) #------------------------------------------------------------------------------------- #TCN: To chnage the database loc to remote PC change the following lines #grep -r 'db_url' sersrc review that the following modules need to be changed: # group,vm,usrloc,uri,domain,jabber,auth_db,pdt,acc,msilo #modparam("auth_db", "db_url", "sql://root:heslo@219.94.42.174/ser") #modparam("usrloc", "db_url", "sql://root:heslo@219.94.42.174/ser") #modparam("group", "db_url", "sql://root:heslo@219.94.42.174/ser") modparam("acc", "db_url", "mysql://root:heslo@localhost/ser") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("group", "db_url", "mysql://ser:heslo@localhost/ser") #------------------------------------------------------------------------------------- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 30 ) ################### NAT ##################### # We will you flag 6 to mark NATed contacts #modparam("registrar", "nat_flag", 6) # Enable NAT pinging #modparam("nathelper", "natping_interval", 3) # Ping only contacts that are known to be # behind NAT #modparam("nathelper", "ping_nated_only", 1) ############################################## #---------------------------------------------------------------------- # accounting #modparam("acc", "log_flag", 2) #modparam("acc", "db_flag", 2) #---------------------------------------------------------------------- alias="redtone" alias="redtone.sip" alias="redtone.sip.com" alias="redtone.sip.com.my" # ------------------------- request routing logic ------------------- route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog ("L_INFO","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too Large"); xlog ("L_INFO","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; }; if (method=="NOTIFY") { break; } if (method=="REGISTER") { xlog("L_INFO","xlog %Tf: REGISTER route uri:%ru method:%rm\r\nfrom: %fu To:%tu Contact:%ct\r\n\r\n\r\n"); #if (is_from_local()) #{ xlog("L_INFO","xlog %Tf: FROM LOCAL REGISTER route uri:%ru method:%rm\r\nfrom:%fu To:%tu Contact:%ct\r\n\r\n\r\n"); # Mark as NAT'ed if (client_nat_test("3")) { #setflag(2); force_rport(); fix_contact(); }; xlog("L_INFO","xlog %Tf: start REGISTER route uri:%ru method:%rm\r \nfrom:%fu To:%tu Contact:%ct\r\n\r\n\r\n"); if (!www_authorize("redtone", "subscriber")) { xlog("L_INFO","%fu failed authentification!\r\n"); www_challenge("redtone", "0"); xlog("L_INFO","%fu failed authentification and challenge!\r\n"); xlog ("L_INFO","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; }; save("location"); xlog("L_INFO","username %fu location REGISTERed!\r\n"); xlog ("L_INFO","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; /* } else { xlog("L_INFO","This domain is not served here!\r\n"); sl_send_reply("403", "This domain is not served here"); }; */ break; }; /* if (method=="INVITE") { if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; t_on_failure("1"); } else */ if (method == "BYE" || method == "CANCEL") { xlog("L_INFO","end_media_session uri:%ru method:%rm \r\nfrom:%fu To:%tu Contact:%ct\r\n"); end_media_session(); }; /* if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; */ # Force subsequent messages to pass trough this proxy if (method == "INVITE") { xlog("L_INFO","invite record route uri:%ru method:%rm \r\nfrom:%fu To:%tu Contact:%ct\r\n"); record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed xlog("L_INFO","force port and fixcontact uri:%ru method:%rm \r \nfrom:%fu To:%tu Contact:%ct\r\n"); force_rport(); fix_contact(); }; if (method=="INVITE") { xlog("L_INFO","set t_onreply uri:%ru method:%rm \r\nfrom:%fu To:%tu Contact:%ct\r\n"); t_on_reply("1"); setflag(6); }; # if (is_uri_host_local()) { lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; # }; #if (method=="INVITE" || method=="ACK") { if (method=="INVITE") { #if (method=="ACK") { xlog("L_INFO","use mediapproxy uri:%ru method:%rm \r\nfrom:%fu To:% tu Contact:%ct\r\n"); use_media_proxy(); }; xlog("L_INFO","relay out\r\n"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { xlog("L_INFO","on reply uri:%ru method:%rm \r\nfrom:%fu To:%tu Contact:%ct\r\n"); #if (status=~"(183)|(2[0-9][0-9])") { if (status=="200") { xlog("L_INFO","REPLY OK uri:%ru method:%rm \r\nfrom:%fu To:%tu Contact:%ct\r\n"); #if (client_nat_test("1")) { fix_contact(); #}; if (!isflagset(7)) { use_media_proxy(); setflag(7); } }; }

1 0

[Serusers] rtpproxy and serctl with problem
by Bruno Oliveira 17 Jun '05

17 Jun '05

hi i have rtpproxy runing and serctl. where it is file config of rtpproxy ? and when I type this command "serctl start" show me in /var/log/syslog this message: "Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: Maxfwd module- initializing Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: INFO: udp_init: SO_RCVBUF is initially 111616 Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: INFO: udp_init: SO_RCVBUF is finally 223232 Jun 16 14:45:33 uiara /usr/local/sbin/ser[3237]: ERROR: tcp_init: bind(6, 0x813753c, 16) on 192.168.1.2: Address already in use" and in pront command show this message: "uiara:~# serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed" when I type "pstree" show me ser and rtpproxy in the list. anyone can a help me! thanks for your time! -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

2 1

[Serusers] error: 400; check if you use aliases in SER
by TCCHAN 17 Jun '05

17 Jun '05

Dear All, After switching from 0.8.14 to 0.9.0 I have this problem of getting error: 400; check if you use aliases in SER when I try to add a user with serctl, eg.serctl add 01548900 999 01548900(a)redtone.com I searched the web and knew someone else has posted this question before and I apologize for asking again ( didn't manage to get the answer). Thanks for any help available. Regards, TC Chan

2 1

Re: Fw: [Serusers] voice mail problem on alias
by Rosario Pingaro 17 Jun '05

17 Jun '05

I fixed the problem commenting the revert_uri(), so it call each time the modified uri that i have after lookup("aliases"). very easy after all. What I am not able to preview is the future implication of commenting revert_uri() giving the fact that in all samples it is present. Thanks Rosario ----- Original Message ----- From: "Stefan Sayer" <sayer(a)fokus.fraunhofer.de> To: "Rosario Pingaro" <rpingar(a)italycom.it> Sent: Thursday, June 16, 2005 6:49 PM Subject: Re: Fw: [Serusers] voice mail problem on alias > Hello, > > that is interesting - if you find the answer can you mail me as well? > (I'd like to add that to the documentation.) > > I think it depends on how you read the email adress from the db - probably > you are loading it with something like > avp_db_load( "$ruri", "$email/$email_scheme"); > and if it is not found you'd have to check if it is an alias and lookup > the real name. > > Another option is to use user_prefix_separator, only a single ser > instance, I think with this scheme there is not this problem with the > alias (but i am not sure). > > Stefan > Rosario Pingaro wrote: >> Debugging sems seems that the probelm is related on how ser pass the user >> to sems; >> in fact when I call the alias and the use is ofline ser pass the real >> user to sems. >> when ser get a timeout pass to sems the alias, sems is not able to find >> an email for the alias and so i get the error. >> >> How to workaround this? >> >> Thanks >> Rosario >> >> >> ----- Original Message ----- >> From: Rosario Pingaro To: serusers(a)lists.iptel.org Sent: Thursday, June 16, >> 2005 5:20 PM >> Subject: [Serusers] voice mail problem on alias >> >> >> Hi I am implementing voice mail using ser and sems, ser 0.9.2 and latesr >> sems from cvs. >> >> I have the failure_route[x] to activate the voicemail also when i get an >> invite timeoute for my users. >> failure_route[1] { >> revert_uri(); >> rewritehostport("xxx.xxx.xxx.xxx:5090"); >> append_branch(); >> t_relay_to_udp("xxx.xxx.xxx.xxx", "5090"); >> } >> >> I don't have any problem when i try to leave a voice mail for an "user". >> On the "alias" the voicemail works fine when the user is offline, but >> when it is online and get a timeout sems respond with a 404 no email >> address for user >> >> the fact that the voicemail works on alias when it is off line push me to >> think the problem is into the failure_route. But I don't know what to add >> to get it working. >> >> Any help is welcome. >> >> Thanks >> Rosario >> >> >> >> -------------------------------------------------------------------------------- >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Users] free-TLS released in experimental tree
by Cesc 16 Jun '05

16 Jun '05

Hi, This is a notification that the first release of free-TLS for SER has been commited to the experimental tree in the CVS repository for SER. For openSER it will be (i think) committed to the core CVS repository right away. User testing is required to fast-track it into the main core, so I encourage you to give it a try. More will be added soon to ease testing (for example, test certificates ready to use ... but you can also create your own). It contains a README file, which should be enough to get things started. It will be improved with a fully working tls_only.ser.cfg file in no time. Also more developer help will be added in the README, for people who want to collaborate. This is all for now. Regards, Cesc

1 0

Re: [Serusers] RE: Domain module
by harry gaillac 16 Jun '05

16 Jun '05

yes i read it -:) I would have spent less time if i had read it but it's not wasted time . I use SER I have to read OPENSER docs. Something is wrong ! Thanks all for help. Harry --- Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> a écrit : > Harry, > > you may find this useful > > http://www.openser.org/docs/modules/domain.html > also you can find full up2date docs for all modules > there > > bogdan > > harry gaillac wrote: > > >According to domain.c file "is_uri_host_local()" > >function > >is needed to check R-URI. > > > >Harry > > > >--- Juha Heinanen <jh(a)tutpro.com> a écrit : > > > > > > > >>put xlog debug into the script. print from header > >>before the tests, > >>etc. also, wheck with fifo command domain_dump > (or > >>something like that) > >>that domain table is really loaded. > >> > >>-- juha > >> > >> > >> > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] Why does added headers show up in the CANCEL?
by Henrik Gustafsson 16 Jun '05

16 Jun '05

Hello! I'm playing around with my own module which is adding or replacing a header in proxied INVITE requests. I'm using the "parse_headers ... anchor_lump ... insert_new_lump_before" method on the proxied request and the outgoing INVITE looks great with the new header. But the request is forwarded by using the tm module and when the transaction receives a CANCEL the outgoing CANCEL gets the header added to the initial INVITE. That would not be a problem for a normal SIP UA, since unknown headers should be ignored, but this UA from a big telecom vendor throws a 400 Bad request. Why is the added headers added to the CANCEL? Shouldn't just the required and transaction identifying headers be enough in a CANCEL? Could I do anything different to prevent this from happening? /Henrik Gustafsson

1 0

[Serusers] Handling CANCELS when a UAC doesn't answer and t_relay has maintained state
by Evan Borgström 16 Jun '05

16 Jun '05

Hello All, Looking for some input if anyone else has run into this situation and/or figured a way around it. - Call comes in from PSTN gateway to a local user - SER finds the user in the location table and t_relay pushes the request off to them - The user is behind an ACL and is not sending keep-alives (yes I know the client is broken) so the INVITE never makes it to them and hence no 180/183 Ringing message comes back - PSTN user gives up and hangs up the phone after 3 seconds of no ringing - PSTN gateway forwards the CANCEL message in and SER cancels the branch - However since t_relay is stateful and re-tries transmissions the INVITE is sent another time or two - Finally the connection times out and the failure_route set for the t_relay kicks in and the INVITE gets pushed off to the voicemail server. - Voicemail server OK's the INVITE and starts retransmitting the OK over and over and over since the PSTN GW doesn't ACK the OK since it knows that call-id as being cancelled. So, should I handle cancel's a little more explicitly by doing something similar to: if ((method=="CANCEL") && (t_lookup_request())) { t_release(); }; What side effects, if any, would this cause? Is there a better way to handle it? Thanks, -Evan

2 1

[Users] Conclusions
by Bogdan-Andrei Iancu 16 Jun '05

16 Jun '05

Folks, There were a lot of ideas, discussions, fears vehiculated in the last days. I try to bring some response as quick and as short as possible just to close the subject. 1) there is and it will be no war - I think the idea to associate war with new thing is in human nature and we cannot fight it. Media Proxy was also welcomed with war and it proved to be a good thing (a complementary solution to nathelper); avpops got the same treatment and now I believe to be a very used module. So let's try t get rid of the war idea - OpenSER is just an alternative to SER and everybody is free to choose what to use. 2) very common idea (no idea who came with it?!) was that OpenSER is less stable and it will be mainly an place for new thing to get mature - wrong ! OpenSER's stability will be one of its strong attribute - if you look to the NEW FEATURE section we will be surprise how many bugs were hunted and fixed. Shortly - we will try to keep OpenSER as stable as SER and maybe more - of course, nothing is bug free :) 3) getting back into SER team - maybe the GCC story will happen, but for this the SER must prove (with facts and not words) that it became a real open project - politics and development are not dictated by any hidden commercial reasons; SER becomes an independent and neutral project (not faked "propriety" of anybody), anybody being free to contribute; the maintainer dictatorship must end - a module maintainer must not have "live or death" right on its modules, it must obey the management group of the project!. 4) documentation management - that a hard problem which frustrated the users a lot - OpenSER tries a different cycle of documentation which will make it easier to use and to update for user: main storage point will be sgml files on CVS -> HTML files on web for reading -> on web wiki page for user to contrib/comment -> filtered time to time by maintainers and moved into sgml; and here ends; easy and straight for both users and maintainers; also Makefiles has built-in commands for generating READMEs and HTMLS from sgml. now, some short answers to some interesting ideas: Greger ------ what was the attitude when "commercial components start popping up with alternative contributions" ? IGNORE! "Do they(voice system) contribute their commercial code to OpenSER? Of course not, they have exactly the same problems as iptelorg.com" Maybe you should look twice to avpops, speeddial, alias_db, uac modules, maintenance for xlog and pdt and all patches to usrloc, rr , register and nathlper - many enterprise solution fixes are there (nat traversal and replication) Kristian Galway, reticent, Maxim, Klaus, Giudace ------------------------------------------------ there is no war and let's forget the word! it's simple matter of choosing and liberty (even if history showed us that all this were gain through war) Everybody is free to use whatever he like. If you don't like OpenSER, fine by me, forget about it, but don't misplace words! Dave, Ingo, Maxim, Atle ------------------------------------------------ by splitting the project nothing is lost, but better (efficient) used; for developers - no more struggle to put your stuff on cvs; more time to work on real things; anyhow you worked was blocked on SER :(. For users - if they know what they want, they can have the liberty to choose based on the preferences, which is a step forward. Let's not hide behind the mentality - simple thing and less options make our life easier... Harry, Samuel ------------- interesting idea of being looser....wonder way?? If like SER you can stick to it and you don't loose or win; If you find OpenSER more suitable for you, you win.....so....where is the looser may I ask?? Cesc ----- |And this process (n. devel) must be well-defined and NOT CONTROLLED exclusively by a company (be iptel or voice-systems, same-o, same-o). If the contribution is voted as desirable by a community of users, it is well tested and so on, it should be accepted." you made a strong point here !! with this email I want to close the subject. Please *do not reply* to it, but only in privately or use openser mailing list if you have related questions. End of subject Bogdam

1 0

[Serusers] sample cfg for ser + asterisk
by Roger Lewau 16 Jun '05

16 Jun '05

Does any one have a sample cfg for a ser + asterisk setup they would like to share? We are planning to use asterisk to provide vm, conferencing and pstn functionality. So any input on this issue would be nice.

1 0

[Serusers] Re: How to execute failure_route block on tcp connect failure.
by Dmitry Isakbayev 16 Jun '05

16 Jun '05

Hello, I have some more information about this issue. There are actually two problems. First - When the 1st attempt fails because connection was refused, connection timed out, or there was no route to host, an error message as logged and the failure_route block is never executed. Second - When the 2nd or 3rd, etc attempt fails due to a connection problem, an error is logged and only 30 seconds later the failure_route block is executed. I would really appreciate if some body could explain what is going on or point me in the right direction. Thank You, Dmitry ----- Original Message ----- From: Dmitry Isakbayev To: serusers(a)lists.iptel.org Sent: Wednesday, June 15, 2005 9:12 PM Subject: How to execute failure_route block on tcp connect failure. Hello, I am having a problem executing a failure_route block when t_relay fails to establish a tcp connection. Everything works as expected when t_relay connects to a proxy and receives back an error like 501. Is there a way to configure t_relay to execute a failure_route block an a tcp connection error? Thank You in advance. Dmitry Here are the logs from connection failure - Jun 15 20:52:14 localhost ser[21870]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Jun 15 20:52:14 localhost ser[21870]: ERROR: tcpconn_connect: tcp_blocking_connect failed Jun 15 20:52:14 localhost ser[21870]: ERROR: tcp_send: connect failed Jun 15 20:52:14 localhost ser[21870]: msg_send: ERROR: tcp_send failed Jun 15 20:52:14 localhost ser[21870]: ERROR: t_forward_nonack: sending request failed

2 3

Re: [Serusers] peering
by Joao Pereira 16 Jun '05

16 Jun '05

Sorry, but I didnt explained my self corectly, what I want is to redirect the calls that aren't for my domain ( for example, calls to sip:maria@fwd.pulver.com or calls to sip:manel@iptel.org). What I whant to know is what to put after this: if( ! uri=~"@fccn.pt" ) { #redirect to fwd or iptel Thanks João Klaus Darilion wrote: > The suggest way is ENUM, but only a few of them creates ENUM entries > for there users. > > regards, > klaus > > Joao Pereira wrote: > >> Hello to all >> I whant my SIP clients to be able to do calls to any other SIP >> network, and to do It, I writed this line (for the IPTEL.ORG case): >> >> if(uri=~"@iptel.org"){ forward(iptel.org,5060); >> break; >> } >> >> But this way, I need to put theese 4 lines to all VoIP providers. >> Is there any way of saying: >> if( ! uri=~"@fccn.pt" ) { # if the invite isnt for my domain (fccn.pt) >> send it to the correct domain >> >> Thanks >> >> João Pereira >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> > >

3 4

[Serusers] Problems with ser and Micronet SP5012
by gsalvucci＠ing.unrc.edu.ar 16 Jun '05

16 Jun '05

Hi, I´m using ser-0.8.14 in a LAN with the ser.cfg from the distribution. I have many ATAs and softphones(x-lite) and a PSTN gateway Micronet SP5012 connected to PBX. Problem is the following: When I call to a PSTN number everything works fine but when call finishes the port FXO of the gateway stays busy. Thanks Gustavo. ---------------------------------------------------------------- Este mensaje ha sido enviado usando el servicio de Webmail de la Universidad Nacional de Río Cuarto. Basado en Horde/IMP

1 0

[Serusers] peering
by Joao Pereira 16 Jun '05

16 Jun '05

Hello to all I whant my SIP clients to be able to do calls to any other SIP network, and to do It, I writed this line (for the IPTEL.ORG case): if(uri=~"@iptel.org"){ forward(iptel.org,5060); break; } But this way, I need to put theese 4 lines to all VoIP providers. Is there any way of saying: if( ! uri=~"@fccn.pt" ) { # if the invite isnt for my domain (fccn.pt) send it to the correct domain Thanks João Pereira

3 2

[Serusers] ser , its forks and knives, and docs, and everything else
by Iqbal 16 Jun '05

16 Jun '05

Hi Another day of going back and forth, feel sorry for the poor newbie who might have signed up for this mailing list, and realized he'd was standing in no mans land between too factions who had dug deep into their trenches.... I cant recall the names of who posted what, and who wants what, but I think a summation, together with some sort of action towards an outcome, is really needed, so I'll try, and please add to it. 1. There is a definant call for two modes, one stable and one more fast paced for those who like to try out new things a) A mode where these two can be linked is required, someone suggested maintaining two threads in CVS, b) do you need to mailing lists, yes/no c) Do you need two websites yes/no d) Should iptel be inherently linked with SER, as most users think, or should their be a non-biased site, I think since SER has its own name, why not use something based upon that, just as other opensource has, if companies wish to sponser, add there name etc, then they should be allowed by the maintainers discretion 2. Documentation, yes it is poor, but that means it can only get better , or at least the way I look at it, a)Who maintains it, is there a overwhelming need for help on this, if so, can we not on the website above, include a link for those who would like to help contribute, and then set some standards as to how it needs to be done etc b) Informal documentation, I think sites such as onsip etc have started something good, and many users will eventually have their own websites etc, with tips and tricks, however if this could also be added to in one place it would be good for the newbie, maybe a wiki, maybe onsip. 3. Working together, now this is a tricky one, since no matter how we look at it, SER seems to be splitting (unless I am misreading the posts), which means even in theory if we want to make it work like i did, its just wont, and we need to accept that, its growing and people want/wish to do it there way, and I guess the compromise is in 1, above. No matter how many emails go back and forth, going back to what it was is not possible, cause human nature will not allow it. 4. Justification and blame, well I think alot of that has happened, and as a user, since I dont know all the details I cannot really comment, except to say, that I am sure its all justified, and everyone is too blame :-), but its time too move on, it was a project and like all projects starts a little chaotically, and somewhere down the line a radical shake up needs to be done, this also happens in companies, trust me I have built quite a few, the casual working modes are dropped for a formal approach, suddenly hand written scraps of paper are no longer acceptable, everything needs to have 3 signatures on it...such is life. I guess this is what SER is going through, and maybe this is what it needs, who's fault it is, or who realized a change is required is irrelevant, but without change I think SER will not work, simply because some of the core team want a change, and you cannot argue with that. Soooo write down what is needed take a vote, ask a few users what they think, and lets get it done. As a side note, I hope the above doesn't seem biased one way or the other, it is not meant to, its meant to get things moving and a decision reached, before everyone goes on holiday, and starts smoking a pipe of peace....without having invited me. Iqbal

1 0

[Serusers] forwarding to voicemail (SEMS)
by Llanos Serna García-Conde 16 Jun '05

16 Jun '05

2 3

[Serusers] Domain module
by harry gaillac 16 Jun '05

16 Jun '05

Hello, I checked my ser.cfg my table domain in db ser and re-read the documentation of domain module . After reading domain.c file I found a exported function for R-URI which is not is_from_local() for outbound domain but is_uri_host_local(): I add this function in my conf file now i can call others domains however we don't find this function anywhere else that domain.c I use documentation from onsip.org which talk about nat however the examples may be wrong or incomplete. Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 1

[Serusers] RE: Domain module
by harry gaillac 16 Jun '05

16 Jun '05

According to domain.c file "is_uri_host_local()" function is needed to check R-URI. Harry --- Juha Heinanen <jh(a)tutpro.com> a écrit : > put xlog debug into the script. print from header > before the tests, > etc. also, wheck with fifo command domain_dump (or > something like that) > that domain table is really loaded. > > -- juha > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

Fw: [Serusers] voice mail problem on alias
by Rosario Pingaro 16 Jun '05

16 Jun '05

Debugging sems seems that the probelm is related on how ser pass the user to sems; in fact when I call the alias and the use is ofline ser pass the real user to sems. when ser get a timeout pass to sems the alias, sems is not able to find an email for the alias and so i get the error. How to workaround this? Thanks Rosario ----- Original Message ----- From: Rosario Pingaro To: serusers(a)lists.iptel.org Sent: Thursday, June 16, 2005 5:20 PM Subject: [Serusers] voice mail problem on alias Hi I am implementing voice mail using ser and sems, ser 0.9.2 and latesr sems from cvs. I have the failure_route[x] to activate the voicemail also when i get an invite timeoute for my users. failure_route[1] { revert_uri(); rewritehostport("xxx.xxx.xxx.xxx:5090"); append_branch(); t_relay_to_udp("xxx.xxx.xxx.xxx", "5090"); } I don't have any problem when i try to leave a voice mail for an "user". On the "alias" the voicemail works fine when the user is offline, but when it is online and get a timeout sems respond with a 404 no email address for user the fact that the voicemail works on alias when it is off line push me to think the problem is into the failure_route. But I don't know what to add to get it working. Any help is welcome. Thanks Rosario -------------------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Re: problem in ser
by Kamran Ahmad 16 Jun '05

16 Jun '05

hello i was using ser-0.9.0 with radiusclient, freeradisus, but unable to configure on this mechine. I am already using these kind of setup on my fedora core mechine with no problem. but i have many problems on this mechine. i am using same ser.cfg as my successfull setups but no success. now i am using ser-0.8.0 with radiusclient.0.4.8 with kernel 2.6.9-1.667 after lot of compiliation i am able to compile auth_radius.so but now i am getting 0(4156) auth module - initializing 0(4156) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(4156) DEBUG: init_mod: auth_radius 0(4156) auth_radius - Initializing 0(4156) auth_radius: No `Cisco' vendor in Radius dictionary 0(4156) find_export: found <pre_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(4156) find_export: found <post_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(4156) ERROR: auth_radius: can't get code for the Digest-Response attribute 0(4156) init_mod(): Error while initializing module auth_radius ERROR: error while initializing modules /var/log/messages ----------------- Jun 9 18:48:50 voice ser: WARNING: could not rev. resolve 217.8.112.104 Jun 9 18:48:50 voice ser[4153]: ERROR: auth_radius: can't get code for the Digest-Response attribute Jun 9 18:48:50 voice ser[4153]: init_mod(): Error while initializing module auth_radius __________________________________ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html

1 0

[Users] LCR module
by Bogdan-Andrei Iancu 16 Jun '05

16 Jun '05

Hi folks, As more an more people are asking for a LCR module in 0.9.4 release, I prepared an official backport of the module for OpenSER 0.9.4 (sources): http://www.openser.org/pub/openser/0.9.4/src/lcr-opsenser-0.9.4.tgz Also there is more and more positive feedback regarding the stability of the module and we are looking for including it in a second fast release on 0.9 branch - a 0.9.5 which will additionally bring the LCR module. Any opinions on this? regards, bogdan

1 0

[Serusers] lost in cvs tags - difference between 0.9.3 and rel_0_9_0
by Klaus Darilion 16 Jun '05

16 Jun '05

Hi! I'm lost in the name of new ser versions and the CVS tags. There is talk about version 0.9, 0.9.2 and 0.9.3, although I only found rel_0_9_0 on cvs. What is the relation between this versions and the CVS tag. thanks, klaus

4 3

[Serusers] voice mail problem on alias
by Rosario Pingaro 16 Jun '05

16 Jun '05

Hi I am implementing voice mail using ser and sems, ser 0.9.2 and latesr sems from cvs. I have the failure_route[x] to activate the voicemail also when i get an invite timeoute for my users. failure_route[1] { revert_uri(); rewritehostport("xxx.xxx.xxx.xxx:5090"); append_branch(); t_relay_to_udp("xxx.xxx.xxx.xxx", "5090"); } I don't have any problem when i try to leave a voice mail for an "user". On the "alias" the voicemail works fine when the user is offline, but when it is online and get a timeout sems respond with a 404 no email address for user the fact that the voicemail works on alias when it is off line push me to think the problem is into the failure_route. But I don't know what to add to get it working. Any help is welcome. Thanks Rosario

1 0

[Serusers] sipath, ser on openwrt
by Klaus Darilion 16 Jun '05

16 Jun '05

Hi all! I just found this project: http://sipath.sourceforge.net/ It uses ser+rtpproxy as outboundproxy on openwrt. Looks like the modified the code - the REGISTER will be stored locally, the contact will be re-written and the REGISTER will be forwarded to the real proxy. regards, klaus

1 0

[Users] CVS head
by Bogdan-Andrei Iancu 16 Jun '05

16 Jun '05

Hi folks, the CVS head now is up-to-date. All new changes (from 0.9.0 to head) from SER were integrated into OpenSER cvs head. LCR and UAC modules are also there. Feel free to give it a try. now the CVS head is ready for forward developing :) bogdan

1 0

RE: [Serusers] OpenSER release
by Giudice, Salvatore 16 Jun '05

16 Jun '05

I would have to agree with you. The ser module documentation is definitely in the 'abysmal' category, although it has improved somewhat in the last 18 months. For example, at least some documentation on modules like registrar now tell you that the save function returns a 200 ok. To the best of my recollection, this sort of detail wasn't there 18 months ago. If openser can produce decent docs, I can see it getting much broader support. -----Original Message----- From: Kristin Galway [mailto:klgalway@yahoo.com] Sent: Wednesday, June 15, 2005 7:31 PM To: info(a)beeplove.com; greger(a)teigre.com; Giudice, Salvatore; serdev(a)lists.iptel.org; serusers(a)lists.iptel.org Subject: Re: [Serusers] OpenSER release This is my first posting to this list but I have been following for about 5 months now. I began to use SER back in February and have used versions from 0.8.14 to cvs versions and have seen plenty of changes here and there. The biggest thing that really troubles me is the almost non-existent documentation for the modules. There has been more than one occasion where I have run into problems to check the documentation to find no mention of moving things from one module to another, or other things such as new features and what not. Most times I have to dig through the source code to actually find information. The sum of this all is that if the fork of OpenSER is going to present itself as a moving package, with up-to-date documention and more so, not having to wait months upon months for a request or a fix to occur. If OpenSER has that intention then I am all for supporting it and using the code. Just because the current code works for some, does not mean that the rest of us users have to wait for something to happen. Kristin Chapter 2. Developer's Guide To be done. _________________________________________________________ Chapter 3. Frequently Asked Questions 3.1. What is the meaning of life ? 3.1. What is the meaning of life ? 42 --- "m36828253-1(a)imap.1and1.com" <m36828253-1(a)imap.1and1.com> wrote: > Why the bug tracking page in a different website. > Why not under iptel.org ? > > Mohammad > > > Original Message: > ----------------- > From: Greger V. Teigre greger(a)teigre.com > Date: Wed, 15 Jun 2005 21:03:21 +0200 > To: Salvatore.Giudice(a)FMR.COM, serdev(a)lists.iptel.org, > serusers(a)lists.iptel.org > Subject: Re: [Serusers] OpenSER release > > > I completely agree with you. I have been told that > there was an attempt at > introducing a bug-tracking system earlier, but that > it has been difficult. > Anyhow, in setting up policies and procedures around > the experimental > directory, we have decided that usage of > http://bugs.sip-router.org will be > mandatory. Hopefully recent, better integration > between the bug tracking > system and the CVS will make it more convenient to > use also for other CVS > modules (however, I don't have a say there). > g-) > > Giudice, Salvatore wrote: > > I am not an advocate for either ser or openser, > but I would like to > > comment. > > > > Is openser going to be equipped with a > forum/ticket system where > > people can document bugs, feature requests, etc > (non-configuration > > issues)? > > > > This is just my observation and you may not agree, > but I believe this > > project could be much better maintained if it used > a more structured > > ticketing style system to manage development > issues instead of the > > current mailing lists. In my experience, mailing > lists like this > > foster a terrible user experience where many > development issues can > > go on without response. > > > > Ideally, if there was a mailing list to address > user issues and > > ticketing system like the one Digium uses to > manage Asterisk, I think > > everyone would benefit by being better informed > and ser would > > ultimately be a better product for it. How many > people out there feel > > that their issues have fallen through the cracks > in the past couple > > years? > > > > -----Original Message----- > > From: Daniel-Constantin Mierla > [mailto:daniel@voice-system.ro] > > Sent: Wednesday, June 15, 2005 4:28 AM > > To: Andrei Pelinescu-Onciul > > Cc: SER developer mailing list; serusers; > users(a)openser.org; > > devel(a)openser.org > > Subject: Re: [Serusers] OpenSER release > > > > On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: > > > >> On Jun 14, 2005 at 22:48, Daniel-Constantin > Mierla > > <daniel(a)voice-system.ro> wrote: > >> > >> [...] > >> > >> > >>> It is your opinion, but I repeat myself, that > the SER code > >>> maintained by us will go further -- I don't > think that someone can > >>> claim that we didn't do the job for our code > (the only discrepancy > >>> is some last-minute adds in xlog (to print avps) > - will be > >>> committed on unstable very soon > > > >>> with the new color patch). The cvs was created > just to ease the > >>> maintainance. The patches would be a nightmare. > >>> > >>> > >> > >> Maybe I've misunderstood you: is this only a > parallel "stabilized" > >> version + some features or is it a full fork (do > you intend to fork > >> unstable also)? > >> > >> > > It is fork for the code that we changed (acc > module, usrloc module > > ...), > > > > in the future may be other that they do not find > the path in SER. We > > will maintain and upgrade our part of code from > SER continuously. > > > >> I have no problem with another stable version, > what worries me is > >> fragmenting the development for unstable (which > is the place where > >> major changes are made). > >> > >> > > I see no fragmenting there -- the situation is the > same for SER as it > > was before. For example, there is no fragment for > acc module, it will > > be > > > > maintained by who did it till now, adding what he > considers necessary > > there. But we came to meet a lot of requests of > why the acc patch is > > not > > > > included in the CVS (it was fully backward > compatible and had new > > features requested by many SER users) and we want > to promote _more > > open_ > > > > approach to contributions to all parts of code. > The acc patch was sent > > on November 1, 2004. No real response (neither > negative, nor positive) > > from maintainer to the submission since then ... > are you aware of a > > good > > > > reason?!?! ... should we wait just about (or more) > half an year for > > each > > > > contribution?!? I will not do that anymore!!! > > > > Daniel > > > >> > >> Andrei > >> > >> > >> > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > -------------------------------------------------------------------- > mail2web - Check your email from the web at > http://mail2web.com/ . > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > __________________________________ Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html

2 1

RE: [Serusers] Re: [Serdev] openser/ser - avoiding forks
by Giudice, Salvatore 16 Jun '05

16 Jun '05

So 31 submitted patches/feature requests were basically ignored in the past year and a half? How can this situation be improved? Is there going to be a commitment to monitor this new ticketing system? How many patches and more feature requests will need to be reconciled 18 months from now? -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Wednesday, June 15, 2005 8:57 PM To: Andrei Pelinescu-Onciul Cc: serdev(a)lists.iptel.org; serusers(a)lists.iptel.org Subject: [Serusers] Re: [Serdev] openser/ser - avoiding forks Hello, I am trying to reconcile patches and improvement suggestions that have been left unanswered on the mailing lists. I went through serusers and serdev mailing list archives from the beginning of 2004 till now. I was mainly focusing on submitted patches and non-trivial feature requests. Below is a list of issues that I was able to find in the mailing list archives. Given the number of messages that were sent to the lists in the last year I am pretty sure that the list below will miss many suggestions. If you submitted a patch or improvement and it was neither accepted nor rejected, and it is still relevant, please resubmit your proposal to serdev(a)lists.iptel.org mailing list or (preferably) create an issue in the bug tracking system at: http://bugs.sip-router.org Please re-submit also improvements sent in individual messages to developers. Summary ------- When it comes to accepting improvements, there surely is space for improvements but the situation, in my opinion, is not as bad as it may seem from the recent discussions. Properly reported bugs get usualy fixed quickly. The more detailed description the faster the fix, so I think there are no big problems with that. The list at the end of this message contains patches and feature requests that I was able to find in the archives. From the list the following issues have not yet been closed: 1, 4, 7, 18, 19, 21, 22, 23, 24, 25, 30, 31 And from those only 1, 4, 18, 21, 22, 25, 31 contained patches that have been left unanswered. That 7 patches which still need to be processed. The following patches have in my opinion low importance: 1 - date header in REGISTER replies 18 - radius fix for acc 21 - check_to for digest credentials without database The following patches can be classified as important: 4 - AVP support in acc 22 - transactional auth replies 25 - Free TLS 31 - Xten improvements of pa In addition to that there are the following improvement suggestions which did not contain any patches: 7, 19, 23, 24 So I was able to find 4 important and 3 less important patches that were not integrated in the last 1.5 year from the 31 items below. Also I created issues in the bug tracking system for all items below that are not yet closed. The winner among them in terms of e-mail volume is free TLS, of course. Jan. PS: Code back-porting is another issue and is not covered here. ------------------------------------------------------------------------ ------ SERUSERS - 2004 --------------- 1) Date Header in REGISTER responses Feature request by TeleSIP, I have a patch for that already from Robert Sanders which will be integrated Status: OPEN Bug: SER-30 2) PA interoperability with RTC, patch submitted by Klaus Darilion, integrated by Jamey HICKS Status: CLOSED 3) branch=0 problem reported on 16 Jul 2004, closed by Jiri as "not a bug" Status: CLOSED 4) AVP patch for acc module, submitted by Ramona on 31 Oct 2004 Status: OPEN Bug: SER-31 SERDEV - 2004 ------------- 5) Video - related patch for nathelper, integrated by Maxim Status: CLOSED 6) fix_nated_contact for nathelper (replied by Maxim) Status: CLOSED 7) 16 Apr 2004 Juha proposed adding npdi and pstn URI parameters (no patches) Status: OPEN Bug: SER-32 8) Maxim submitted patch adding PIDs of all processes into the pid file, resolved by another means Status: CLOSED 9) 20 Apr. 2004 Alexander Mayhofer submitted patch for rtpproxy, commited by Maxim Status: CLOSED 10) Test for realm in auth_radius too tight, reported by Cesar Hernandez, fixed by Jan Status: CLOSED 11) 23 Apr 2004 Multicast support patch, commited by Andrei Status: CLOSED 12) 24 Apr 2004 Postgres patch by Alexander Mayhofer, in CVS Status: CLOSED 13) max_expires patch submitted by Jamey Hicks, in CVS Status: CLOSED 14) start/stop commands fro SEMS into serctl by Klaus Darilion, rejected. Status: CLOSED 15) 30 Jun 2004 User agent support in xlog module by Alexander Mayhofer, in cvs Status: CLOSED 16) 7 Jul 2004 Patch for configurable user agent string by Maxim, resolved by other means Status: CLOSED 17) 21 Jul 2004 Patch for storing user agent string in location table, commited by Maxim Status: CLOSED 18) 21 Jul 2004 Radius-related patch for acc module Status: OPEN Bug: SER-33 19) 2 Oct 2004 Escaped uri characters are not interpreted properly, no patch Status: DEFERED Bug: SER-34 20) 27 Oct 2004 Mysql_ping patch by Dan Pascu, in cvs Status: CLOSED 21) 10 Nov 2004 uri_db patch to make it possible to use check_to and check_from without database (submitted by Marian Dumitru). Status: OPEN Bug: SER-35 22) 25 Dec 2004 Maxim's patch to make it possible send transactional replies from authentication modules Status: OPEN Bug: SER-36 23) Additional header fields requested by Juha (server, refer-to) Status: OPEN Bug: SER-37 SERUSERS - 2005 --------------- 24) feature request route("string") Status: DEFERED Bug: SER-38 25) Free TLS Status: OPEN Bug: SER-39 26) 19 Apr 2005 branch=0 in ACK problem, rejected by Jiri Status: CLOSED 27) 20 Apr 2005 mysql_ping patch for 0.8.14, not important enough to be commited to 0.8.14 Status: CLOSED 28) Backport of radius auth changes from unstable to stable, rejected Status: CLOSED SERDEV - 2005 ------------- 29) Transaction replies in auth modules (by Maxim) Status: OPEN Bug: SER-36 30) NAT support in usrloc and registrar, partially done Status: OPEN Bug: SER-40 31) RFC3265,RFC3903 support in pa, I could not find the patch Status: OPEN Bug: SER-41 _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

RE: [Serusers] OpenSER release
by Giudice, Salvatore 16 Jun '05

16 Jun '05

Maybe extending the documentation with a Wiki would prove effective without putting too much responsibility on any one maintainer's shoulders. This would also cut down on the number of repeat questions on the SER user's list. -----Original Message----- From: Maxim Sobolev [mailto:sobomax@portaone.com] Sent: Thursday, June 16, 2005 4:30 AM To: Alberto Cruz Cc: Greger V. Teigre; SER developer mailing list; Giudice, Salvatore; serusers Subject: Re: [Serusers] OpenSER release On Wed, Jun 15, 2005 at 10:57:06PM -0500, Alberto Cruz wrote: > And what about documentation? When was the last time that Administration > Guide was updated? And what about the Modules documentation? Are you volunteering? -Maxim

1 0

[Serusers] storing voice messages
by Llanos Serna García-Conde 16 Jun '05

16 Jun '05

Hi all, I'm using SEMS and it works fine. I know it stores the voices messages in /tmp directory for a very short time before sending them to the e-mail addresses. I would like to store them permanently, how can I do that? Thanks for your help, Llanos

1 0

[Users] RE: [Serusers] OpenSER release
by Giudice, Salvatore 16 Jun '05

16 Jun '05

Policy is only as good as its communication to the people it applies to. Besides the initial announcement, how many times was this site publicized in the mailing list or enforced as standard practice to have patches added to the project? -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Wednesday, June 15, 2005 9:08 PM To: info(a)beeplove.com Cc: Giudice, Salvatore; daniel(a)voice-system.ro; andrei(a)iptel.org; serdev(a)iptel.org; serusers(a)iptel.org; devel(a)openser.org; users(a)openser.org Subject: Re: [Serusers] OpenSER release There is such a bug tracking system running at http://bugs.sip-router.org It is powered by Jira and is quite easy to use. I announced it to serdev mailing list some time ago but there seemed to be no interest, from core developers only myself and Andrei created accounts. I was using it for a while and I find it easy to use and convenient. Jan. On 15-06-2005 14:24, info(a)beeplove.com wrote: > I agree with you. > We should have proper bug tracking system, including feature request. > > MOhammad > > > > Original Message: > ----------------- > From: Giudice, Salvatore Salvatore.Giudice(a)FMR.COM > Date: Wed, 15 Jun 2005 14:12:15 -0400 > To: daniel(a)voice-system.ro, andrei(a)iptel.org, serdev(a)iptel.org, > serusers(a)iptel.org, devel(a)openser.org, users(a)openser.org > Subject: RE: [Serusers] OpenSER release > > > I am not an advocate for either ser or openser, but I would like to > comment. > > Is openser going to be equipped with a forum/ticket system where people > can document bugs, feature requests, etc (non-configuration issues)? > > This is just my observation and you may not agree, but I believe this > project could be much better maintained if it used a more structured > ticketing style system to manage development issues instead of the > current mailing lists. In my experience, mailing lists like this foster > a terrible user experience where many development issues can go on > without response. > > Ideally, if there was a mailing list to address user issues and > ticketing system like the one Digium uses to manage Asterisk, I think > everyone would benefit by being better informed and ser would ultimately > be a better product for it. How many people out there feel that their > issues have fallen through the cracks in the past couple years? > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Wednesday, June 15, 2005 4:28 AM > To: Andrei Pelinescu-Onciul > Cc: SER developer mailing list; serusers; users(a)openser.org; > devel(a)openser.org > Subject: Re: [Serusers] OpenSER release > > On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: > > >On Jun 14, 2005 at 22:48, Daniel-Constantin Mierla > <daniel(a)voice-system.ro> wrote: > > > >[...] > > > > > >>It is your opinion, but I repeat myself, that the SER code maintained > by > >>us will go further -- I don't think that someone can claim that we > >>didn't do the job for our code (the only discrepancy is some > last-minute > >>adds in xlog (to print avps) - will be committed on unstable very soon > > >>with the new color patch). The cvs was created just to ease the > >>maintainance. The patches would be a nightmare. > >> > >> > > > >Maybe I've misunderstood you: is this only a parallel "stabilized" > >version + some features or is it a full fork (do you intend to fork > >unstable also)? > > > > > It is fork for the code that we changed (acc module, usrloc module ...), > > in the future may be other that they do not find the path in SER. We > will maintain and upgrade our part of code from SER continuously. > > >I have no problem with another stable version, what worries me is > >fragmenting the development for unstable (which is the place where > major > >changes are made). > > > > > I see no fragmenting there -- the situation is the same for SER as it > was before. For example, there is no fragment for acc module, it will be > > maintained by who did it till now, adding what he considers necessary > there. But we came to meet a lot of requests of why the acc patch is not > > included in the CVS (it was fully backward compatible and had new > features requested by many SER users) and we want to promote _more open_ > > approach to contributions to all parts of code. The acc patch was sent > on November 1, 2004. No real response (neither negative, nor positive) > from maintainer to the submission since then ... are you aware of a good > > reason?!?! ... should we wait just about (or more) half an year for each > > contribution?!? I will not do that anymore!!! > > Daniel > > > > >Andrei > > > > > > > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > -------------------------------------------------------------------- > mail2web - Check your email from the web at > http://mail2web.com/ . > > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Problem on pthread and dtmf
by Nicky 16 Jun '05

16 Jun '05

Dear all, 1. Please help, when i run sems -E i experienced this error, what can be the caused of it:~ ~~11352) ERROR: start (AmThread.cpp:73): pthread create failed with code 12 2. There is signal of DTMF which is not pressed but is being send to the server, please see below for the log. Please advice what can be the caused and problem. Thank you in advance. ~~(4395) DEBUG: isdn_audio_eval_dtmf_relative (IvrDtmfDetector.cpp:349): Posting D TMF 0: 5 (5) (4395) DEBUG: postEvent (AmEventQueue.cpp:47): AmEventQueue: trying to post even t (4395) DEBUG: postEvent (AmEventQueue.cpp:55): AmEventQueue: event posted (4395) DEBUG: processEvents (AmEventQueue.cpp:68): before processing event (4395) DEBUG: process (IvrPython.cpp:988): IvrPython processing event... (4395) DEBUG: process (IvrPython.cpp:991): IvrDialog processing event... (4395) DEBUG: onDTMFEvent (IvrPython.cpp:1130): IvrPython::onDTMFEvent(): callin g onDTMFCallback key is 5... (4395) DEBUG: ivrEmptyMediaQueue (IvrPython.cpp:102): IVR: emptying media queue . (4395) DEBUG: IvrMediaEvent (IvrEvents.cpp:31): New Media Event: 1, (4395) DEBUG: postEvent (AmEventQueue.cpp:47): AmEventQueue: trying to post even t (4395) DEBUG: postEvent (AmEventQueue.cpp:55): AmEventQueue: event posted (4395) DEBUG: ivrStopRecording (IvrPython.cpp:132): IVR: stop recording. best regards, nicky

2 1

[Serusers] MediaProxy NO sound
by TCCHAN 16 Jun '05

16 Jun '05

Dear All, I apologize for asking a repeated question. I tried ser 0.8.14 and 0.9 to work with mediaproxy. I could setup call but there are silence. I did try to bypass the proxydispatcher modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") #modparam("mediaproxy", "mediaproxy_socket","/var/run/proxydispatcher.sock") and the result is the same. I have been stuck for days. Thanks in advenced for any help available. Regards, TC Chan The proxydispatcher dump the following: [root@ser mediaproxy]# ./proxydispatcher.py --no-fork Listening for commands on local socket `/var/run/proxydispatcher.sock' Default mediaproxy server is on `/var/run/mediaproxy.sock' command request fe5a6f36-c6092310(a)192.228.118.240 192.228.118.240:18278:audio 210.187.111.86 219.94.42.174 remote 210.187.111.86 remote Sipura/SPA2000-2.0.13(g) info=from:01548900@219.94.42.174,to:01548901@219.94.42.174,fromtag:d9170fe31e4cbe8o0,totag: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '127.0.0.1 35000' command execution time: 16.89 ms command lookup fe5a6f36-c6092310(a)192.228.118.240 192.228.118.240:18280:audio 210.187.111.86 219.94.42.174 remote 219.94.42.174 unknown Sipura/SPA2000-2.0.13(g) info=from:01548900@219.94.42.174,to:01548901@219.94.42.174,fromtag:d9170fe31e4cbe8o0,totag:8f2251e89d625018i1 forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '127.0.0.1 35000' command execution time: 3.92 ms After a long while .......... command delete fe5a6f36-c6092310(a)192.228.118.240 info= forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '' command execution time: 2.53 ms While Mediaproxy dump the following: [root@ser mediaproxy]# ./mediaproxy.py --no-fork Listening for commands on local socket `/var/run/mediaproxy.sock' Listening for remote commands is disabled Using IP address `127.0.0.1' for the RTP/RTCP proxy command request fe5a6f36-c6092310(a)192.228.118.240 192.228.118.240:18278:audio 210.187.111.86 219.94.42.174 remote 210.187.111.86 remote Sipura/SPA2000-2.0.13(g) info=from:01548900@219.94.42.174,to:01548901@219.94.42.174,fromtag:d9170fe31e4cbe8o0,totag:,dispatcher session fe5a6f36-c6092310(a)192.228.118.240: started. listening on 127.0.0.1:35000 command execution time: 11.72 ms command lookup fe5a6f36-c6092310(a)192.228.118.240 192.228.118.240:18280:audio 210.187.111.86 219.94.42.174 remote 219.94.42.174 unknown Sipura/SPA2000-2.0.13(g) info=from:01548900@219.94.42.174,to:01548901@219.94.42.174,fromtag:d9170fe31e4cbe8o0,totag:8f2251e89d625018i1,dispatcher command execution time: 1.32 ms command status command execution time: 0.36 ms session fe5a6f36-c6092310(a)192.228.118.240: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) session fe5a6f36-c6092310(a)192.228.118.240: ended (did timeout). command status command execution time: 0.18 ms command delete fe5a6f36-c6092310(a)192.228.118.240 info=dispatcher command execution time: 0.24 ms Session dump the following: ./[root@ser mediaproxy]# ./sessions.py Caller Via Called Status Duration Codec Type Traffic ------------------------------------------------------------------------------------------------------------ 192.228.118.240:18278 - 127.0.0.1:35000 - 192.228.118.240:18280 inactive 0'04" Unknown Audio 0/0/0 Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 [root@ser mediaproxy]# ./sessions.py No RTP sessions My ser script is as follow: #debug=3 #fork=no #log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- #loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/textops.so" #loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" #---------------------------------------------------------------------- # mediaproxy modparam("mediaproxy", "natping_interval", 60) #modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("registrar", "nat_flag", 2) #---------------------------------------------------------------------- #---------------------------------------------------------------------- # uri #modparam("uri", "db_url", "sql://root:heslo@localhost/ser") #modparam("uri", "uri_table", "subscriber") #---------------------------------------------------------------------- #---------------------------------------------------------------------- # TCN: no forking #modparam("registrar","desc_time_order",1) #modparam("usrloc","desc_time_order",1) #modparam("registrar","append_branches",0) #----------------------------------------------------------------------- # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 1) #modparam("usrloc", "db_mode", 2) #------------------------------------------------------------------------------------- #TCN: To chnage the database loc to remote PC change the following lines #grep -r 'db_url' sersrc review that the following modules need to be changed: # group,vm,usrloc,uri,domain,jabber,auth_db,pdt,acc,msilo #modparam("auth_db", "db_url", "sql://root:heslo@219.94.42.174/ser") #modparam("usrloc", "db_url", "sql://root:heslo@219.94.42.174/ser") #modparam("group", "db_url", "sql://root:heslo@219.94.42.174/ser") modparam("acc", "db_url", "mysql://root:heslo@localhost/ser") modparam("auth_db", "db_url", "mysql://root:heslo@localhost/ser") modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser") modparam("group", "db_url", "mysql://root:heslo@localhost/ser") #------------------------------------------------------------------------------------- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 30 ) ################### NAT ##################### # We will you flag 6 to mark NATed contacts #modparam("registrar", "nat_flag", 6) # Enable NAT pinging #modparam("nathelper", "natping_interval", 3) # Ping only contacts that are known to be # behind NAT #modparam("nathelper", "ping_nated_only", 1) ############################################## #---------------------------------------------------------------------- # accounting #modparam("acc", "log_flag", 2) #modparam("acc", "db_flag", 2) #---------------------------------------------------------------------- alias="redtone" alias="redtone.sip" alias="redtone.sip.com" alias="redtone.sip.com.my" # ------------------------- request routing logic ------------------- route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog ("L_ERR","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too Large"); xlog ("L_ERR","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; }; if (method=="REGISTER") { xlog("L_ERR","xlog %Tf: REGISTER route uri:%ru method:%rm\r\nfrom:% fu To:%tu Contact:%ct\r\n\r\n\r\n"); #if (is_from_local()) #{ xlog("L_ERR","xlog %Tf: FROM LOCAL REGISTER route uri:%ru method:%rm\r\nfrom:%fu To:%tu Contact:%ct\r\n\r\n\r\n"); # Mark as NAT'ed if (client_nat_test("3")) { #setflag(2); force_rport(); fix_contact(); }; xlog("L_ERR","xlog %Tf: start REGISTER route uri:%ru method:%rm\r \nfrom:%fu To:%tu Contact:%ct\r\n\r\n\r\n"); if (!www_authorize("redtone", "subscriber")) { xlog("L_ERR","%fu failed authentification!\r\n"); www_challenge("redtone", "0"); xlog("L_ERR","%fu failed authentification and challenge!\r\n"); xlog ("L_ERR","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; }; save("location"); xlog("L_ERR","username %fu location REGISTERed!\r\n"); xlog ("L_ERR","-----------------------------------------------------------------\r\n\r\n\r\n\r\n"); break; /* } else { xlog("L_ERR","This domain is not served here!\r\n"); sl_send_reply("403", "This domain is not served here"); }; */ break; }; /* if (method=="INVITE") { if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; t_on_failure("1"); } else */ if (method == "BYE" || method == "CANCEL") { end_media_session(); }; /* if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; */ # Force subsequent messages to pass trough this proxy if (method == "INVITE") { record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; if (method=="INVITE") { t_on_reply("1"); }; # if (is_uri_host_local()) { if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; # }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { #if (status=~"200") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; }

3 2

[Users] documentation site
by Daniel-Constantin Mierla 16 Jun '05

16 Jun '05

Hello, there is a Wiki page where users can contribute to OpenSER documentation. It is open for everybody and we hope that everybody will have a fair attitude. http://openser.org/dokuwiki/ I have started a new document "OpenSER Core CookBook", we hope there will be many contributions. Best regards, Daniel

1 0

[Serusers] uri matching for dial plans
by Iqbal 16 Jun '05

16 Jun '05

Hi I have a nice table of prefixes and gateways, which works great using lcr, however I am now working on dial plans, i.e user can call 001, not 0044 etc etc, based on what I have in my grp table. Problem here is with codes like 001 since this would allow 001234 also, which for instance could be puerto rico (which it aint) but if it was I wouldnt want the call to go there. so I am doing something like if (uri=001 && !uri=001234) make the call...BUT this is a pain, especially for 001, I have had to block out 30 odd routes manually, which is making my ser config look really ugly. add into that other routes and suddenly we have a painful config file. So what I wanted to know is it possible to pull info from a DB, using exec, and store the value in a variable which can be used later. the grp table only really allows u to bind a "name" and then see if the from user is in that group, but it dont let you to the break that down into finer options, especially dial plans Iqbal

2 2

[serusers] PA module
by michael p 16 Jun '05

16 Jun '05

hi i tried to load the module pa but it doesn't work. how can i check for the presentity? i tried with the softphone eyebeam to add a contact to my contact list but the little man is like off line. i added in ser.cfg : loadmodule "/usr/lib/ser/modules/pa.so" and modparam("pa", "default_expires", 3600) and " handle_subscription("registrar"); " in : if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("dev.annatel.net", "subscriber")) { www_challenge("dev.annatel.net", "0"); break; }; save("location"); handle_subscription("registrar"); break; }; i don't use jabber so i want when the user registrar that do the suscribe and transmit to the others... is that correct what i did? why there are so little doc about all the modules of SER? thanks for the help... M.

1 0

[Serusers] ask about using SER 0.9 with Cisco PSTN Gateway
by gunawan 16 Jun '05

16 Jun '05

i hav problem using ser 0.9 with Cisco PSTN gateway... i use Cisco Router 2620XM with voice module VIC2FXO as the RJ-11 connection and IOS Cisco 12.2(13)T4... does anyone in forum ever use cisco 2620XM as a PSTN gateway???? I need help in Cisco Configuration and Ser.cfg scripts.... it's work when i use Cisco Call Manager which use H323 Protocol as my server , and right now i want to try use SIP as my protocol and SER as my server.... thanks in advance..... __________________________________ Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html

3 2

[serusers] PA module
by michael p 16 Jun '05

16 Jun '05

hi where can i got informations about pa module? Is that true it doesn't work with sip messages and only work with jabber? where can i find a step by step for jabber and pa? thanks M.

1 0

[Serusers] Onsip.org ser.cfg + pstn + uac - can't authenticate on gateway
by Cameron Beattie 16 Jun '05

16 Jun '05

I am trying the get the uac module working with the onsip.org ser.cfg (the pstn + mediaproxy version). I think the problem is related to the authentication on the gateway device. It doesn't seem to be challenging SER. Below is the SIP traffic and then the ser.cfg file. Any suggestions would be appreciated. For information, the gateway (198.22.67.70) is provided by Nufone.net and is an Asterisk server, I believe. Thanks in advance for any advice. Regards Cameron U 147.202.xx.xxx:5060 -> 60.234.xxx.xxx:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-fff17e3f;rport=5060;received=60.234 .xxx.xxx..From: <sip:user1@mydomain.com>;tag=f9493da2f8a6d10ao0..To: <sip:00442070800000@mydomain.com>..Call-ID: ce674b 82-e2fd1caa@192.168.0.11..CSeq: 102 INVITE..Server: Sip EXpress router (0.9.2 (i386/linux))..Content-Length: 0..Warning: 392 147.202.xx.xxx:5060 "Noisy feedback tells: pid=6752 req_src_ip=60.234.xxx.xxx req_src_port=5060 in_uri=sip:00442070800000@mydomain.com out_uri =sip:00442070800000@198.22.67.70 via_cnt==1".... ## U 147.202.xx.xxx:5060 -> 198.22.67.70:5060 INVITE sip:00442070800000@198.22.67.70 SIP/2.0..Record-Route: <sip:00442070800000@147.202.xx.xxx:5060;nat=yes;ftag=f9493da2f8a6d10ao0;lr=on>. .Via: SIP/2.0/UDP 147.202.xx.xxx;branch=z9hG4bK8b0a.2871d5c2.0..Via: SIP/2.0/UDP 192.168.0.11:5060;rport=5060;received=60.234.xxx.xxx;bran ch=z9hG4bK-fff17e3f..From: <sip:user1@mydomain.com>;tag=f9493da2f8a6d10ao0..To: <sip:00442070800000@mydomain.com>..Cal l-ID: ce674b82-e2fd1caa@192.168.0.11..CSeq: 102 INVITE..Max-Forwards: 16..Contact: <sip:user1@60.234.xxx.xxx:5060>..Expires: 240..User- Agent: Sipura/SPA3000-3.1.3(GWa)..Content-Length: 424..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura ..Content-Type: application/sdp....v=0..o=- 757701 757701 IN IP4 192.168.0.11..s=-..c=IN IP4 147.202.xx.xxx..t=0 0..m=audio 35026 RTP/AVP 0 2 4 8 18 96 97 98 100 101..a=rtpmap:0 PCMU/8000..a=rtpmap:2 G726-32/8000..a=rtpmap:4 G723/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729 a/8000..a=rtpmap:96 G726-40/8000..a=rtpmap:97 G726-24/8000..a=rtpmap:98 G726-16/8000..a=rtpmap:100 NSE/8000..a=rtpmap:101 telephone-event /8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. # U 147.202.xx.xxx:5060 -> 60.234.xxx.xxx:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)..Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bK-fff17e3f;rport=5060;recei ved=60.234.xxx.xxx..From: <sip:user1@mydomain.com>;tag=f9493da2f8a6d10ao0..To: <sip:00442070800000@mydomain.com>;tag=66 9cac32ae43cfbf664b867e0fb4dd5a.572f..Call-ID: ce674b82-e2fd1caa@192.168.0.11..CSeq: 102 INVITE..Server: Sip EXpress router (0.9.2 (i386/l inux))..Content-Length: 0..Warning: 392 147.202.xx.xxx:5060 "Noisy feedback tells: pid=6752 req_src_ip=60.234.xxx.xxx req_src_port=5060 i n_uri=sip:00442070800000@mydomain.com out_uri=sip:00442070800000@198.22.67.70 via_cnt==1".... ## U 198.22.67.70:5060 -> 147.202.xx.xxx:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 147.202.xx.xxx;branch=z9hG4bK8b0a.2871d5c2.0..Via: SIP/2.0/UDP 192.168.0.11:5060;received=60.234. xxx.xxx;branch=z9hG4bK-fff17e3f..From: <sip:user1@mydomain.com>;tag=f9493da2f8a6d10ao0..To: <sip:00442070800000@mydomain.com>;tag=as40c6caec..Call-ID: ce674b82-e2fd1caa@192.168.0.11..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, O PTIONS, BYE, REFER, NOTIFY..Contact: <sip:00442070800000@198.22.67.70>..Content-Length: 0.... # U 147.202.xx.xxx:5060 -> 198.22.67.70:5060 ACK sip:00442070800000@198.22.67.70 SIP/2.0..Via: SIP/2.0/UDP 147.202.xx.xxx;branch=z9hG4bK8b0a.2871d5c2.0..From: <sip:user1@mydomain.com>;tag=f9493da2f8a6d10ao0..Call-ID: ce674b82-e2fd1caa@192.168.0.11..To: <sip:00442070800000@mydomain.com>;tag=as40c6caec. .CSeq: 102 ACK..User-Agent: Sip EXpress router(0.9.2 (i386/linux))..Content-Length: 0.... My ser.cfg is below: debug=3 fork=yes #fork=no log_stderror=yes check_via=no dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://user:password@localhost/ser" sock_mode=0666 #fifo_mode=0666 #fifo_user=root listen=147.202.xx.xxx port=5060 children=4 loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/uac.so" modparam("auth_db|permissions|uri_db|usrloc|acc", "db_url", "mysql:// user:password @localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("uac","credential","user:asterisk:password") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("147.202.xx.xxx:5060;nat=yes"); # insert IP address } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { setflag(1); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact: \*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (uri=~"^sip:[+|00][0-9]*@") { # International PSTN route(4); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{8}@") { # Domestic PSTN route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; setflag(1); setflag(3); route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- avp_write("i:45", "inv_timeout"); route(5); t_on_failure("4"); resetflag(8); t_relay_to_udp("198.22.67.70","5060"); route(1); } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(1); break; }; lookup("location"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ 0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } failure_route[5] { if (t_check_status("401|407")) { if (isflagset(8)) { t_reply("503","Auth failed"); break; } if (uac_auth()) { setflag(8); t_on_failure("5"); append_branch(); t_relay(); } } }

2 1

[Serusers] ultra monkey & ser
by Jack Wei 16 Jun '05

16 Jun '05

hi, I'm trying to set up a cluster of ser boxes using ultramonkey. i want ldirectord to detect if any of the ser boxes is down, but am not able to set it up properly. Here's my ldirector.cf: checktimeout=10 checkinterval=2 autoreload=no logfile="local0" quiescent=yes virtual=10.1.201.189:5060 real=10.1.201.191:5060 gate real=10.1.201.193:5060 gate scheduler=wlc protocol=udp # checktype=connect persistent=600 Using the above configuration, ipvsadm reports the real servers are up even if they aren't. If i enable "checktype=connect", ipvsadm reports the real servers are always down. if anyone know of solutions, please let me know. thanks. Jack Wei __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

Re: [Serusers] OpenSER release
by info＠beeplove.com 16 Jun '05

16 Jun '05

Are you guys planning to have a feature request page as well? Mohammad Original Message: ----------------- From: Greger V. Teigre greger(a)teigre.com Date: Wed, 15 Jun 2005 21:03:21 +0200 To: Salvatore.Giudice(a)FMR.COM, serdev(a)iptel.org, serusers(a)iptel.org Subject: Re: [Serusers] OpenSER release I completely agree with you. I have been told that there was an attempt at introducing a bug-tracking system earlier, but that it has been difficult. Anyhow, in setting up policies and procedures around the experimental directory, we have decided that usage of http://bugs.sip-router.org will be mandatory. Hopefully recent, better integration between the bug tracking system and the CVS will make it more convenient to use also for other CVS modules (however, I don't have a say there). g-) Giudice, Salvatore wrote: > I am not an advocate for either ser or openser, but I would like to > comment. > > Is openser going to be equipped with a forum/ticket system where > people can document bugs, feature requests, etc (non-configuration > issues)? > > This is just my observation and you may not agree, but I believe this > project could be much better maintained if it used a more structured > ticketing style system to manage development issues instead of the > current mailing lists. In my experience, mailing lists like this > foster a terrible user experience where many development issues can > go on without response. > > Ideally, if there was a mailing list to address user issues and > ticketing system like the one Digium uses to manage Asterisk, I think > everyone would benefit by being better informed and ser would > ultimately be a better product for it. How many people out there feel > that their issues have fallen through the cracks in the past couple > years? > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Wednesday, June 15, 2005 4:28 AM > To: Andrei Pelinescu-Onciul > Cc: SER developer mailing list; serusers; users(a)openser.org; > devel(a)openser.org > Subject: Re: [Serusers] OpenSER release > > On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: > >> On Jun 14, 2005 at 22:48, Daniel-Constantin Mierla > <daniel(a)voice-system.ro> wrote: >> >> [...] >> >> >>> It is your opinion, but I repeat myself, that the SER code >>> maintained by us will go further -- I don't think that someone can >>> claim that we didn't do the job for our code (the only discrepancy >>> is some last-minute adds in xlog (to print avps) - will be >>> committed on unstable very soon > >>> with the new color patch). The cvs was created just to ease the >>> maintainance. The patches would be a nightmare. >>> >>> >> >> Maybe I've misunderstood you: is this only a parallel "stabilized" >> version + some features or is it a full fork (do you intend to fork >> unstable also)? >> >> > It is fork for the code that we changed (acc module, usrloc module > ...), > > in the future may be other that they do not find the path in SER. We > will maintain and upgrade our part of code from SER continuously. > >> I have no problem with another stable version, what worries me is >> fragmenting the development for unstable (which is the place where >> major changes are made). >> >> > I see no fragmenting there -- the situation is the same for SER as it > was before. For example, there is no fragment for acc module, it will > be > > maintained by who did it till now, adding what he considers necessary > there. But we came to meet a lot of requests of why the acc patch is > not > > included in the CVS (it was fully backward compatible and had new > features requested by many SER users) and we want to promote _more > open_ > > approach to contributions to all parts of code. The acc patch was sent > on November 1, 2004. No real response (neither negative, nor positive) > from maintainer to the submission since then ... are you aware of a > good > > reason?!?! ... should we wait just about (or more) half an year for > each > > contribution?!? I will not do that anymore!!! > > Daniel > >> >> Andrei >> >> >> > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

2 1

RE: [Serusers] OpenSER release
by Giudice, Salvatore 16 Jun '05

16 Jun '05

Is anyone using this bug tracking system? Is this system sponsored by iptel.org? -----Original Message----- From: m36828253-1(a)imap.1and1.com [mailto:m36828253-1@pop.1and1.com] Sent: Wednesday, June 15, 2005 3:16 PM To: greger(a)teigre.com; Giudice, Salvatore; serdev(a)lists.iptel.org; serusers(a)lists.iptel.org Subject: Re: [Serusers] OpenSER release Why the bug tracking page in a different website. Why not under iptel.org ? Mohammad Original Message: ----------------- From: Greger V. Teigre greger(a)teigre.com Date: Wed, 15 Jun 2005 21:03:21 +0200 To: Salvatore.Giudice(a)FMR.COM, serdev(a)lists.iptel.org, serusers(a)lists.iptel.org Subject: Re: [Serusers] OpenSER release I completely agree with you. I have been told that there was an attempt at introducing a bug-tracking system earlier, but that it has been difficult. Anyhow, in setting up policies and procedures around the experimental directory, we have decided that usage of http://bugs.sip-router.org will be mandatory. Hopefully recent, better integration between the bug tracking system and the CVS will make it more convenient to use also for other CVS modules (however, I don't have a say there). g-) Giudice, Salvatore wrote: > I am not an advocate for either ser or openser, but I would like to > comment. > > Is openser going to be equipped with a forum/ticket system where > people can document bugs, feature requests, etc (non-configuration > issues)? > > This is just my observation and you may not agree, but I believe this > project could be much better maintained if it used a more structured > ticketing style system to manage development issues instead of the > current mailing lists. In my experience, mailing lists like this > foster a terrible user experience where many development issues can > go on without response. > > Ideally, if there was a mailing list to address user issues and > ticketing system like the one Digium uses to manage Asterisk, I think > everyone would benefit by being better informed and ser would > ultimately be a better product for it. How many people out there feel > that their issues have fallen through the cracks in the past couple > years? > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Wednesday, June 15, 2005 4:28 AM > To: Andrei Pelinescu-Onciul > Cc: SER developer mailing list; serusers; users(a)openser.org; > devel(a)openser.org > Subject: Re: [Serusers] OpenSER release > > On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: > >> On Jun 14, 2005 at 22:48, Daniel-Constantin Mierla > <daniel(a)voice-system.ro> wrote: >> >> [...] >> >> >>> It is your opinion, but I repeat myself, that the SER code >>> maintained by us will go further -- I don't think that someone can >>> claim that we didn't do the job for our code (the only discrepancy >>> is some last-minute adds in xlog (to print avps) - will be >>> committed on unstable very soon > >>> with the new color patch). The cvs was created just to ease the >>> maintainance. The patches would be a nightmare. >>> >>> >> >> Maybe I've misunderstood you: is this only a parallel "stabilized" >> version + some features or is it a full fork (do you intend to fork >> unstable also)? >> >> > It is fork for the code that we changed (acc module, usrloc module > ...), > > in the future may be other that they do not find the path in SER. We > will maintain and upgrade our part of code from SER continuously. > >> I have no problem with another stable version, what worries me is >> fragmenting the development for unstable (which is the place where >> major changes are made). >> >> > I see no fragmenting there -- the situation is the same for SER as it > was before. For example, there is no fragment for acc module, it will > be > > maintained by who did it till now, adding what he considers necessary > there. But we came to meet a lot of requests of why the acc patch is > not > > included in the CVS (it was fully backward compatible and had new > features requested by many SER users) and we want to promote _more > open_ > > approach to contributions to all parts of code. The acc patch was sent > on November 1, 2004. No real response (neither negative, nor positive) > from maintainer to the submission since then ... are you aware of a > good > > reason?!?! ... should we wait just about (or more) half an year for > each > > contribution?!? I will not do that anymore!!! > > Daniel > >> >> Andrei >> >> >> > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

2 1

Re: [Serusers] OpenSER release
by m36828253-1＠imap.1and1.com 16 Jun '05

16 Jun '05

Why the bug tracking page in a different website. Why not under iptel.org ? Mohammad Original Message: ----------------- From: Greger V. Teigre greger(a)teigre.com Date: Wed, 15 Jun 2005 21:03:21 +0200 To: Salvatore.Giudice(a)FMR.COM, serdev(a)iptel.org, serusers(a)iptel.org Subject: Re: [Serusers] OpenSER release I completely agree with you. I have been told that there was an attempt at introducing a bug-tracking system earlier, but that it has been difficult. Anyhow, in setting up policies and procedures around the experimental directory, we have decided that usage of http://bugs.sip-router.org will be mandatory. Hopefully recent, better integration between the bug tracking system and the CVS will make it more convenient to use also for other CVS modules (however, I don't have a say there). g-) Giudice, Salvatore wrote: > I am not an advocate for either ser or openser, but I would like to > comment. > > Is openser going to be equipped with a forum/ticket system where > people can document bugs, feature requests, etc (non-configuration > issues)? > > This is just my observation and you may not agree, but I believe this > project could be much better maintained if it used a more structured > ticketing style system to manage development issues instead of the > current mailing lists. In my experience, mailing lists like this > foster a terrible user experience where many development issues can > go on without response. > > Ideally, if there was a mailing list to address user issues and > ticketing system like the one Digium uses to manage Asterisk, I think > everyone would benefit by being better informed and ser would > ultimately be a better product for it. How many people out there feel > that their issues have fallen through the cracks in the past couple > years? > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Wednesday, June 15, 2005 4:28 AM > To: Andrei Pelinescu-Onciul > Cc: SER developer mailing list; serusers; users(a)openser.org; > devel(a)openser.org > Subject: Re: [Serusers] OpenSER release > > On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: > >> On Jun 14, 2005 at 22:48, Daniel-Constantin Mierla > <daniel(a)voice-system.ro> wrote: >> >> [...] >> >> >>> It is your opinion, but I repeat myself, that the SER code >>> maintained by us will go further -- I don't think that someone can >>> claim that we didn't do the job for our code (the only discrepancy >>> is some last-minute adds in xlog (to print avps) - will be >>> committed on unstable very soon > >>> with the new color patch). The cvs was created just to ease the >>> maintainance. The patches would be a nightmare. >>> >>> >> >> Maybe I've misunderstood you: is this only a parallel "stabilized" >> version + some features or is it a full fork (do you intend to fork >> unstable also)? >> >> > It is fork for the code that we changed (acc module, usrloc module > ...), > > in the future may be other that they do not find the path in SER. We > will maintain and upgrade our part of code from SER continuously. > >> I have no problem with another stable version, what worries me is >> fragmenting the development for unstable (which is the place where >> major changes are made). >> >> > I see no fragmenting there -- the situation is the same for SER as it > was before. For example, there is no fragment for acc module, it will > be > > maintained by who did it till now, adding what he considers necessary > there. But we came to meet a lot of requests of why the acc patch is > not > > included in the CVS (it was fully backward compatible and had new > features requested by many SER users) and we want to promote _more > open_ > > approach to contributions to all parts of code. The acc patch was sent > on November 1, 2004. No real response (neither negative, nor positive) > from maintainer to the submission since then ... are you aware of a > good > > reason?!?! ... should we wait just about (or more) half an year for > each > > contribution?!? I will not do that anymore!!! > > Daniel > >> >> Andrei >> >> >> > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

4 3

[Serusers] SER not responding
by Ryan Pagquil 16 Jun '05

16 Jun '05

Hi guys, I have a problem with SER... whenever I register using X-lite i never get logged in.... X-lite always prompts me that the connection timed out.... in my SER machine I run tcpdump -i gem0 | grep 5060 and i only see the request comming from my firewall and the SER machine is not responding on the request. here is the diagnostic of x-lite: SEND TIME: 3530347 SEND >> 202.84.24.107:5060 REGISTER sip:philonline.com SIP/2.0 Via: SIP/2.0/UDP 202.84.24.126:5060;rport;branch=z9hG4bK1165436EDDDD11D9BFB500E04CAB4AB4 From: Ryan Pagquil <sip:rpagquil@philonline.com>;tag=2916895101 To: Ryan Pagquil <sip:rpagquil@philonline.com> Contact: "Ryan Pagquil" <sip:rpagquil@202.84.24.126:5060> Call-ID: 11654266DDDD11D9BFB500E04CAB4AB4(a)philonline.com CSeq: 53597 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 here is my ser.cfg: #debug=4 # debug level (cmd line: -dddddddddd) #fork=no #log_stderror=yes # (cmd line: -E) # Uncomment these lines to enter debugging mode #debug=3 #fork=yes #log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; record_route(); if (loose_route()) { t_relay(); break; }; if (uri==mysql) { if (method=="REGISTER") { if (!www_authorize("philonline.com", "subscriber")) { www_challenge("philonline.com", "0"); break; }; save("location"); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; if (!t_relay()) { sl_reply_error(); }; } my ser.cfg contains the default configuration and i just edited and uncommented some parts of it. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

2 2

[Serusers] How to execute failure_route block on tcp connect failure.
by Dmitry Isakbayev 16 Jun '05

16 Jun '05

Hello, I am having a problem executing a failure_route block when t_relay fails to establish a tcp connection. Everything works as expected when t_relay connects to a proxy and receives back an error like 501. Is there a way to configure t_relay to execute a failure_route block an a tcp connection error? Thank You in advance. Dmitry Here are the logs from connection failure - Jun 15 20:52:14 localhost ser[21870]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Jun 15 20:52:14 localhost ser[21870]: ERROR: tcpconn_connect: tcp_blocking_connect failed Jun 15 20:52:14 localhost ser[21870]: ERROR: tcp_send: connect failed Jun 15 20:52:14 localhost ser[21870]: msg_send: ERROR: tcp_send failed Jun 15 20:52:14 localhost ser[21870]: ERROR: t_forward_nonack: sending request failed

1 0

[Users] RE: [Serusers] OpenSER release
by info＠beeplove.com 16 Jun '05

16 Jun '05

I agree with you. We should have proper bug tracking system, including feature request. MOhammad Original Message: ----------------- From: Giudice, Salvatore Salvatore.Giudice(a)FMR.COM Date: Wed, 15 Jun 2005 14:12:15 -0400 To: daniel(a)voice-system.ro, andrei(a)iptel.org, serdev(a)iptel.org, serusers(a)iptel.org, devel(a)openser.org, users(a)openser.org Subject: RE: [Serusers] OpenSER release I am not an advocate for either ser or openser, but I would like to comment. Is openser going to be equipped with a forum/ticket system where people can document bugs, feature requests, etc (non-configuration issues)? This is just my observation and you may not agree, but I believe this project could be much better maintained if it used a more structured ticketing style system to manage development issues instead of the current mailing lists. In my experience, mailing lists like this foster a terrible user experience where many development issues can go on without response. Ideally, if there was a mailing list to address user issues and ticketing system like the one Digium uses to manage Asterisk, I think everyone would benefit by being better informed and ser would ultimately be a better product for it. How many people out there feel that their issues have fallen through the cracks in the past couple years? -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: Wednesday, June 15, 2005 4:28 AM To: Andrei Pelinescu-Onciul Cc: SER developer mailing list; serusers; users(a)openser.org; devel(a)openser.org Subject: Re: [Serusers] OpenSER release On 06/14/05 23:21, Andrei Pelinescu-Onciul wrote: >On Jun 14, 2005 at 22:48, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > >[...] > > >>It is your opinion, but I repeat myself, that the SER code maintained by >>us will go further -- I don't think that someone can claim that we >>didn't do the job for our code (the only discrepancy is some last-minute >>adds in xlog (to print avps) - will be committed on unstable very soon >>with the new color patch). The cvs was created just to ease the >>maintainance. The patches would be a nightmare. >> >> > >Maybe I've misunderstood you: is this only a parallel "stabilized" >version + some features or is it a full fork (do you intend to fork >unstable also)? > > It is fork for the code that we changed (acc module, usrloc module ...), in the future may be other that they do not find the path in SER. We will maintain and upgrade our part of code from SER continuously. >I have no problem with another stable version, what worries me is >fragmenting the development for unstable (which is the place where major >changes are made). > > I see no fragmenting there -- the situation is the same for SER as it was before. For example, there is no fragment for acc module, it will be maintained by who did it till now, adding what he considers necessary there. But we came to meet a lot of requests of why the acc patch is not included in the CVS (it was fully backward compatible and had new features requested by many SER users) and we want to promote _more open_ approach to contributions to all parts of code. The acc patch was sent on November 1, 2004. No real response (neither negative, nor positive) from maintainer to the submission since then ... are you aware of a good reason?!?! ... should we wait just about (or more) half an year for each contribution?!? I will not do that anymore!!! Daniel > >Andrei > > > _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

2 1

RE: [Serusers] DNS Resolution of SDP-"Connection Address" at SER
by Gerhard Zweimüller 16 Jun '05

16 Jun '05

Thanks for your answer. The funny thing is: I am trying quite hard to convince the vendor (actually a well-known VoIP-chip/module/appliance maker) that FQDN in SDP is a good (and allowed) thing. Even more funny: In the preceding firmware-release FQDN was no problem. But in the current release they apparently *removed* the FQDN-support and now they try to convince me that FQDN in SDP is not allowed :-( Although I sent them RFCs and everything to prove them wrong. But there you go. I tried your suggestion with fix_nated_sdp and so far it seems to work. Thanks for the tip! Hopefully the vendor will change its RFC-support policy soon. Kind regards, Gerhard -----Original Message----- From: Zeus Ng [mailto:zeus.ng@isquare.com.au] Sent: Sunday, June 12, 2005 8:30 AM To: Gerhard Zweimüller Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] DNS Resolution of SDP-"Connection Address" at SER Forcing SER to do stuff in order to fix problem UA implementation, I don't think this is a good approach. Using FQDN as connection information in SDP is the recommended approach in RFC 2327, though few UA do this. I would suggest you contact the vendor to fix the UA that can't work with DNS. Have a look on the fix_nated_sdp() function in nathelper. I haven't tried this before but it might work. However, there is no guarantee. Let us know if it works. > -----Original Message----- > From: Gerhard Zweimüller > Sent: Saturday, 11 June 2005 11:07 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] DNS Resolution of SDP-"Connection Address" at SER > > > Hi Serusers, > > I have a problem with DNS resolution in SIP/SDP "Connection > Information" > / "ConnectionAddress". Maybe somebody can help me: > > In the network we use SIP-UAs that are integrated into ADSL-Modems > from Allied Telesyn (AT RG 634). > The SIP-client itself works OK. But when the unit is given a > system-name in the configuration, it will always use its DNS-name > instead of its IP-addr as "Connection Address" in the "Connection > Information" in SDP in INVITE and corresponding OK messages. > > Now we want to add SIP-UAs that are NOT capable of resolving DNS-Names > in the "Connection Address" field. > Alle messages pass through SER of course and the SDP-part so far > remains unchanged. > > Now my question to the list: > Is there a simple way of forcing SER to do the DNS-resolution of in > incoming message, put in the IP-address in "Connection Address" and > forward it to the other UA? > > Thanks a lot in advance! > Gerhard > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > __________________________________________________________________________________ Dieses Mail wurde vom Infotech SecureMail Service ueberprueft und fuer sicher befunden. Fuer weitere Informationen zu Infotech SecureMail Service waehlen Sie bitte: www.infotech.at/securemail/ This email has been scanned by Infotech SecureMail Service and it has been classified as secure. For more information on Infotech SecureMail direct your web browser to: www.infotech.at/securemail/

3 2

[Serusers] failed to lookup for few seconds for every one hour
by Karun Chemudugunta 15 Jun '05

15 Jun '05

Hi Group, I am seeing issues where lookup() failing for every one hour for period of few seconds. ( ~ 20 to 30 seconds). Is it known issues ? Any house keeping happening for every hour ??? Version SER software: 0.8.12 Please find my attached ser.cfg Thanks in advance. Karun _______________________________________________________________________ Jun 13 12:41:14 encasip001 /usr/sbin/ser[4805]: lookup failed :SIP request from config Jun 13 12:41:14 encasip001 /usr/sbin/ser[4810]: lookup failed :SIP request from config Jun 13 13:40:44 encasip001 /usr/sbin/ser[4805]: lookup failed :SIP request from config Jun 13 13:41:15 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request from config Jun 13 13:41:15 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request from config Jun 13 13:41:17 encasip001 /usr/sbin/ser[4807]: lookup failed :SIP request from config Jun 13 13:41:17 encasip001 /usr/sbin/ser[4806]: lookup failed :SIP request from config Jun 13 13:41:17 encasip001 /usr/sbin/ser[4807]: lookup failed :SIP request from config Jun 13 13:41:17 encasip001 /usr/sbin/ser[4806]: lookup failed :SIP request from config Jun 13 13:41:17 encasip001 /usr/sbin/ser[4807]: lookup failed :SIP request from config Jun 13 13:41:18 encasip001 /usr/sbin/ser[4811]: lookup failed :SIP request from config Jun 13 13:41:18 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request from config Jun 13 13:41:18 encasip001 /usr/sbin/ser[4811]: lookup failed :SIP request from config Jun 13 13:41:19 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request from config Jun 13 13:41:19 encasip001 /usr/sbin/ser[4805]: lookup failed :SIP request from config Jun 13 13:41:19 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request from config Jun 13 13:41:20 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request from config Jun 13 14:40:57 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request from config Jun 13 14:40:57 encasip001 /usr/sbin/ser[4811]: lookup failed :SIP request from config Jun 13 14:40:57 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request from config Jun 13 14:40:58 encasip001 /usr/sbin/ser[4806]: lookup failed :SIP request from config Jun 13 14:40:58 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request from config Jun 13 14:40:59 encasip001 /usr/sbin/ser[4811]: lookup failed :SIP request from config Jun 13 14:41:00 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request from config Jun 13 15:40:35 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request from config Jun 13 15:40:35 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request from config Jun 13 15:40:36 encasip001 /usr/sbin/ser[4807]: lookup failed :SIP request from config Jun 13 15:40:36 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request from config Jun 13 15:40:36 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request from config Jun 13 15:40:36 encasip001 /usr/sbin/ser[4806]: lookup failed :SIP request from config Jun 13 15:40:36 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request from config Jun 13 15:40:37 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request from config Jun 13 15:40:37 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request from config Jun 13 15:40:39 encasip001 /usr/sbin/ser[4810]: lookup failed :SIP request from config ___________________________________________________________________________ # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=9 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) #dns=no # (cmd. line: -r) #rev_dns=no # (cmd. line: -R) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" alias=209.233.189.177 alias=voice.engca.bevocal.com # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/exec.so" loadmodule "/usr/lib/ser/modules/textops.so" #loadmodule "/usr/lib/ser/modules/acc.so" #modparam("acc", "log_level", 2) #modparam("acc", "log_flag", 1) #modparam("acc", "report_cancels", 1) #modparam("acc", "failed_transactions", 1) # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if(method=="OPTIONS") { sl_send_reply("200", "Ok"); break; }; if (method=="REGISTER") { save("location"); break; }; if(method=="INVITE") { log(1,"about lookup in route zero"); if (!lookup("location","encamed")) { log(1,"lookup failed :SIP request from config"); sl_send_reply("404", "Not Found"); break; }; t_on_failure("5"); }; }; # setflag(1); # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } failure_route[5] { revert_uri(); if (!lookup("location","encamed")) { log(1,"lookup failed :SIP request from config"); t_reply("404", "Not Found"); break; }; append_branch(); append_urihf("CC-Diversion: ", "\r\n"); t_on_failure("6"); t_relay(); } failure_route[6] { log(1,"Inside failure_route six"); exec_msg("printenv|grep '^SIP'|mail -v -s onfailure6 krcgunta(a)yahoo.com"); } _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] RE: CPL: serctl fifo LOAD_CPL problem
by Aisling O'Driscoll 15 Jun '05

15 Jun '05

I realise now what I've done wrong - I wasn't following the dtd file correctly (/tmp/ser-0.9.0/modules/cpl-c/cpl-06.dtd). ---- Original Message ---- From: ashling.odriscoll(a)cit.ie To: serusers(a)iptel.org Subject: RE: CPL: serctl fifo LOAD_CPL problem Date: Wed, 15 Jun 2005 15:20:52 +0100 >Hello, > >I am trying to upload a script onto SER using the "serctl fifo >UPLOAD_CPL user(a)xxx.xxx.xxx.xxx /path/to/file.xml" command.It is very >simple script to reject a call from caller 2093. Unfortunately I have >the errors seen below. I have included my simple cpl script, the >errors and the relevant part of the ser.cfg. Any help would be >greatly appreciated. > >Many Thanks, >Aisling. > >CPL SCRIPT [saved in /opt/ser/etc/ser]: > ><?xml version="1.0" encoding="UTF-8"?> ><!DOCTYPE cpl PUBLIC '-//IETF//DTD RFCxxxx CPL 1.0//EN'"cpl-06.dtd"> > <cpl> > <string-switch field="from"> > <string matches="2093(a)157.190.74.152"> > <response status="reject" reason="I dont want to"/> > </string> > </string-switch> > </cpl> > >SER.CFG: > >if (method == "INVITE"){ > if(!cpl_run_script("incoming", "is_stateless")) > { > # script execution failed > t_reply("500", "CPL script execution failed"); > }; > route(3); > break; > } else if (method == "REGISTER"){ > #handle REGISTER messages with CPL script > cpl_process_register(); > route(2); > break; > }; >} > >ERRORS: > >localhost:/opt/ser/etc/ser # :0: Element cpl content does not follow >the DTD >Expecting (ancillary? , subaction* , outgoing? , incoming?), got >(string-switch ) >:0: Element string content does not follow the DTD >Expecting (location | lookup | remove-location | address-switch | >string-switch | language-switch | time-switch | priority-switch | >proxy | redirect | reject | mail | log | sub)?, got (response ) >:0: No declaration for attribute matches of element string >:0: No declaration for element response >:0: No declaration for attribute status of element response >:0: No declaration for attribute reason of element response > 1(20171) ERROR:cpl-c:encodeCPL: CPL script doesn't respect CPL >grammar > 1(20171) ERROR: fifo_server: command (LOAD_CPL) processing failed > -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] serweb for ser 0.9.0?
by maka 15 Jun '05

15 Jun '05

hi all, is there a working serweb for ser-0.9.0? I tried using serweb_2004-07-27 but it appears to be using the old database structure with 'perms' column in subscriber table in numerous queries. is there a working serweb for ser-0.9.0 or I would have to modify this one? cheers

2 1

[Serusers] CPL: serctl fifo LOAD_CPL problem
by Aisling O'Driscoll 15 Jun '05

15 Jun '05

Hello, I am trying to upload a script onto SER using the "serctl fifo UPLOAD_CPL user(a)xxx.xxx.xxx.xxx /path/to/file.xml" command.It is very simple script to reject a call from caller 2093. Unfortunately I have the errors seen below. I have included my simple cpl script, the errors and the relevant part of the ser.cfg. Any help would be greatly appreciated. Many Thanks, Aisling. CPL SCRIPT [saved in /opt/ser/etc/ser]: <?xml version="1.0" encoding="UTF-8"?> <!DOCTYPE cpl PUBLIC '-//IETF//DTD RFCxxxx CPL 1.0//EN'"cpl-06.dtd"> <cpl> <string-switch field="from"> <string matches="2093(a)157.190.74.152"> <response status="reject" reason="I dont want to"/> </string> </string-switch> </cpl> SER.CFG: if (method == "INVITE"){ if(!cpl_run_script("incoming", "is_stateless")) { # script execution failed t_reply("500", "CPL script execution failed"); }; route(3); break; } else if (method == "REGISTER"){ #handle REGISTER messages with CPL script cpl_process_register(); route(2); break; }; } ERRORS: localhost:/opt/ser/etc/ser # :0: Element cpl content does not follow the DTD Expecting (ancillary? , subaction* , outgoing? , incoming?), got (string-switch ) :0: Element string content does not follow the DTD Expecting (location | lookup | remove-location | address-switch | string-switch | language-switch | time-switch | priority-switch | proxy | redirect | reject | mail | log | sub)?, got (response ) :0: No declaration for attribute matches of element string :0: No declaration for element response :0: No declaration for attribute status of element response :0: No declaration for attribute reason of element response 1(20171) ERROR:cpl-c:encodeCPL: CPL script doesn't respect CPL grammar 1(20171) ERROR: fifo_server: command (LOAD_CPL) processing failed -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

RE: [Serusers] Ser Presence Server + Windows MSN 4.7
by PHAN, Quang-Minh 15 Jun '05

15 Jun '05

Hi, Yes, the PA module in SER does handle SUBSCRIBE, NOTIFY messages but it's not compatible with windows messenger 4.7. I don't know if someone is working on this module but at the moment it cannot be use to provide presence service. Minh > -----Message d'origine----- > De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la > part de Simith Nambiar > Envoyé : jeudi 9 juin 2005 07:47 > À : 'serusers(a)lists.iptel.org' > Objet : [Serusers] Ser Presence Server + Windows MSN 4.7 > > > Hello Serusers, > I'am new to the List , please can you give me the > below details , > > 1.) Does the Presence server in Ser do SUBSCRIBE/ NOTIFY processing and is > it compatible with Windows Messenger 4.7 ? > what i mean to say is that , does it reflect the current status of > the > logged in users, and generate appropriate Notifications ? > > Please let me know ! > > Thank You, > Cheers. > Simith > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] And the winner is ...
by harry gaillac 15 Jun '05

15 Jun '05

Samuel, You don't offend me. I just think all developpers should work together on the same project freely. It would be better for open community (and business) . If you want to do business don't give your work to open community (see Microsoft) . Go away SER players. Harry /////////////////////////////////////////////////////// > Business is the motivation for all SER users, I do > not > think that is > someone that do stuff with SER just because he has > no > other thing to do /////////////////////////////////////////////////////// --- Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> a écrit : > And the looser is...... > Harry! > > Samuel. > > P.D. It's nothing personal, I took you as an example > of user, I hope that it does not offend you. > > Unclassified. > >>> harry gaillac <gaillacharry(a)yahoo.fr> 06/15/05 > 12:08PM >>> > > Either voice-sistem or iptelorg.com !? > > ///////////////////////////////////////////////////////// > From: Daniel-Constantin Mierla > > Business is the motivation for all SER users, I do > not > think that is > someone that do stuff with SER just because he has > no > other thing to do > -- many work for companies that use SER, others do > studies and use ser > for projects... So the business behind is the engine > for development. > //////////////////////////////////////////////////////// > > Harry > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] 478 Error Response
by naresh 15 Jun '05

15 Jun '05

Hi, I'm receiving 478 status for a REGISTER sent to SER Proxy. I'm sending Register request URI in the form of domain name. But 478 status code as given by SER is an invalid response as per RFC 3261. When does SER responds with status code 478? (Below is SIP register request packet and response packet) Is it configuration problem or any thing wrong in my packet? -------------------------------------------------------------------------------- Register Request to SER:--> Request-Line: REGISTER sip:ser.testlab.com:5060 SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.3.180:5060 From: <sip:1801@ser.testlab.com:5060>;tag=425215125 SIP from address: sip:1801@ser.testlab.com:5060 SIP tag: 425215125 To: <sip:1801@ser.testlab.com:5060> SIP to address: sip:1801@ser.testlab.com:5060 Call-ID: 1002330992425(a)ser.testlab.com CSeq: 1 REGISTER Contact: <sip:1801@192.168.3.180:5060>;expires=60 max-forwards: 70 Content-Length: 0 Response from SER: <-- Status-Line: SIP/2.0 478 Unresolveable destination (478/TM) Status-Code: 478 Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.3.180:5060 From: <sip:1801@ser.testlab.com:5060>;tag=425215125 SIP from address: sip:1801@ser.testlab.com:5060 SIP tag: 425215125 To: <sip:1801@ser.testlab.com:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-d8b4 SIP to address: sip:1801@ser.testlab.com:5060 SIP tag: a6a1c5f60faecf035a1ae5b6e96e979a-d8b4 Call-ID: 1002330992425(a)ser.testlab.com CSeq: 1 REGISTER Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 192.168.3.198:5060 "Noisy feedback tells: pid=2361 req_src_ip=192.168.3.180 req_src_port=5060 in_uri=sip:ser.testlab.com:5060 out_uri=sip:ser.testlab.com:5060 via_cnt==1" -------------------------------------------------------------------------------- Thanks and regards, naresh

2 1

[Serusers] SERusers, developers and those undecided...a users point of view
by Iqbal 15 Jun '05

15 Jun '05

Hi Wow this fight is better than Mike Tysons..not that it took much beating...but as a user I felt I may as well give my $0.02 worth. I have used alot of opensource SW in my time, to build some great businesses, and in return helped bring the internet to a whole lot of people who would never have used it. From the days of sendmail and apache, to msql and mysql, perl and php...its all added to this virtual world we call the Internet. Since my development days are way past me, I cant really contribute in code, hence I'll do a little in my own way, together with the numerous others who have helped me along the way.... This argument (which is exactly what it is), boils down to moving fast or keeping stability. In this list there are both types of users, and I think both are needed, if we dont move fast, then SER will come to a standstill, but if we move too quickly, and dont get new users problems resolved like basic setup (which Greger et al at doing) then we will be short of new users...who always come with new ideas. Whether the solution is in a fork or not, is irrelevant here, because having core developers not working together just aint good. Judging by opensource in other projects, I think the key is too keep it as one (or so it should seem to the casual observer), and from there you could have two threads, one where the fast paced users would be able to pull from and test, and the other more safer stable environment. Let the user decide what he wishes to opt for, but dont make it look as if there are two separate projects going on. The logistics of this I havent a clue on, but I am sure some on this list do, 2 URL's etc doesnt make much difference, except as Klaus pointed out I've got too mailing lists, and if I try to reply to user questions I know I will only really have time for one list, and you will just get alot of cross posts happening. If the core developers do not pull together, then how can the community. Maybe split the core team into two (which is what is happening), one who can work on stable releases, and one on a more faster release...slightly less stable , eventually the latter one being sucked into the former. Also I think there is a call to allow others to submit modules....and these should not always have to be approved, I think if approved they go into stable ser, else users should be able to use them at there own risk. If you look at asterisk you will see lots of software developed for it, and modules etc, and I am sure not all these are approved, but what it helps do is to spread SER. Whether we have two sites 3 or even more, unless the core developers are happy, this wont work, guys you gotta sort it out, else SER and VoIP will not reach its full potential...a plea from a user :-) Iqbal

1 0

[Serusers] CDR (radacct) mediation
by Wojciech Ziniewicz 15 Jun '05

15 Jun '05

Hello, i've got accounting from radius, do you know any opensource or well know algorithms (may be bash, C or mysql ) to calculate the prefix and be sure that the calcluation is sure (having the prefix tables in the CSV or any other format)? -- Pozdrawiam, Wojciech Ziniewicz Optocomp sp.z.o.o, www.optocomp.pl mailto: wojtekz(a)optocomp.pl +48(0)691031535

1 0

Re: [Serusers] And the winner is ...
by Samuel Osorio Calvo 15 Jun '05

15 Jun '05

And the looser is...... Harry! Samuel. P.D. It's nothing personal, I took you as an example of user, I hope that it does not offend you. Unclassified. >>> harry gaillac <gaillacharry(a)yahoo.fr> 06/15/05 12:08PM >>> Either voice-sistem or iptelorg.com !? ///////////////////////////////////////////////////////// From: Daniel-Constantin Mierla Business is the motivation for all SER users, I do not think that is someone that do stuff with SER just because he has no other thing to do -- many work for companies that use SER, others do studies and use ser for projects... So the business behind is the engine for development. //////////////////////////////////////////////////////// Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Users] OpenSER release
by Bogdan-Andrei Iancu 15 Jun '05

15 Jun '05

Hi SER community, there are almost two years from the last official SER release and things do not promise too much right now. Not only that the progress stuck somewhere on the way (rel 0.9.0 was started more than half a year ago), but even any attempt to push thing forward seems to be denied - I tried along with Daniel to push the release, but seems that not everybody shares our and comunity's interest regarding the public part of SER - upgrades were rolled back, new software contributions haven't found their way in (like TLS and other new modules), modules maintained by other developers are inaccessible. Unfortunately this is not a good environment if we what to have some future progress for SER. And this is the main reason for starting a new project called OpenSER - http://www.openser.org . It's called open because its most important attribute is its opening to new ideas and contributions, fast developing and more involvement of the comunity. Along with quality, the progress is the main concern. We will continue to support and develop the SER project as much as so far and as much as possible, but OpenSER will give the liberty for more. OpenSER serves the interest of all SER users and will not change its purpose - as a fact I have the pleasure to announce its first release - OpenSER 0.9.4. The web site offers a comprehensive listing of new features and fixes - http://www.openser.org/index.php#features. For people already familiar to SER 0.9.3, going to http://www.openser.org/diffs-0.9.0.php will be more helpful. Also the documentation (http://www.openser.org/index.php#docs) will help for a better understanding of this release. For any additional question please use the project mailing lists: users(a)openser.org - dedicated for general purpose discussions about OpenSER stable releases. devel(a)openser.org - dedicated for discussions related to development version and next steps of OpenSER. bogdan

14 30

[Serusers] And the winner is ...
by harry gaillac 15 Jun '05

15 Jun '05

Either voice-sistem or iptelorg.com !? ///////////////////////////////////////////////////////// From: Daniel-Constantin Mierla Business is the motivation for all SER users, I do not think that is someone that do stuff with SER just because he has no other thing to do -- many work for companies that use SER, others do studies and use ser for projects... So the business behind is the engine for development. //////////////////////////////////////////////////////// Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] Log messages don't appear in /var/log/messages
by Cameron Beattie 15 Jun '05

15 Jun '05

I am trying to debug my ser.cfg and so have inserted log statements in it. However they are not appearing in the messages file. I'm sure that there's something very basic I'm doing incorrectly. Relevant bits from my ser.cfg follows: debug=8 fork=no log_stderror=yes ... route{ log(1,"My dog has fleas/n"); ... Any suggestions greatfully received. Regards Cameron

1 0

[Serusers] Request Status Dump
by TCCHAN 15 Jun '05

15 Jun '05

Dear All, I know that I can dump a request to a file using : exec_msg("cat >> /usr/local/sbin/log.txt"); How do I get a dump of the response? I try this and it won't run: onreply_route[1] { if (status == "200" || status=="202") { .... ... exec_msg("cat >> /usr/local/sbin/log.txt"); Is there a way that I could log the response to a file. Thank for any help available. Regards, TC Chan

2 1

RE: [Serusers] failed to lookup for few seconds for every one hour
by PHAN, Quang-Minh 15 Jun '05

15 Jun '05

Hi, One hour is the default interval between REGISTER messages sent by softphone. After one hour, if SER doesn't receive any new REGISTER messages, it will delete all expired contacts. In your case, I guess that softphones send REGISTER messages a little late. From the moment SER delete expired contacts to the moment it receives new REGISTER message, lookup will failed. Minh > -----Message d'origine----- > De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la > part de Karun Chemudugunta > Envoyé : mardi 14 juin 2005 02:19 > À : serusers(a)lists.iptel.org > Objet : [Serusers] failed to lookup for few seconds for every one hour > > Hi Group, > > I am seeing issues where lookup() failing for every one hour for period > of few seconds. ( ~ 20 to 30 seconds). > > Is it known issues ? > > Any house keeping happening for every hour ??? > > Version SER software: 0.8.12 > > Please find my attached ser.cfg > > Thanks in advance. > > Karun > _______________________________________________________________________ > Jun 13 12:41:14 encasip001 /usr/sbin/ser[4805]: lookup failed :SIP request > from config > Jun 13 12:41:14 encasip001 /usr/sbin/ser[4810]: lookup failed :SIP request > from config > Jun 13 13:40:44 encasip001 /usr/sbin/ser[4805]: lookup failed :SIP request > from config > Jun 13 13:41:15 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request > from config > Jun 13 13:41:15 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request > from config > Jun 13 13:41:17 encasip001 /usr/sbin/ser[4807]: lookup failed :SIP request > from config > Jun 13 13:41:17 encasip001 /usr/sbin/ser[4806]: lookup failed :SIP request > from config > Jun 13 13:41:17 encasip001 /usr/sbin/ser[4807]: lookup failed :SIP request > from config > Jun 13 13:41:17 encasip001 /usr/sbin/ser[4806]: lookup failed :SIP request > from config > Jun 13 13:41:17 encasip001 /usr/sbin/ser[4807]: lookup failed :SIP request > from config > Jun 13 13:41:18 encasip001 /usr/sbin/ser[4811]: lookup failed :SIP request > from config > Jun 13 13:41:18 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request > from config > Jun 13 13:41:18 encasip001 /usr/sbin/ser[4811]: lookup failed :SIP request > from config > Jun 13 13:41:19 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request > from config > Jun 13 13:41:19 encasip001 /usr/sbin/ser[4805]: lookup failed :SIP request > from config > Jun 13 13:41:19 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request > from config > Jun 13 13:41:20 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request > from config > Jun 13 14:40:57 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request > from config > Jun 13 14:40:57 encasip001 /usr/sbin/ser[4811]: lookup failed :SIP request > from config > Jun 13 14:40:57 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request > from config > Jun 13 14:40:58 encasip001 /usr/sbin/ser[4806]: lookup failed :SIP request > from config > Jun 13 14:40:58 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request > from config > Jun 13 14:40:59 encasip001 /usr/sbin/ser[4811]: lookup failed :SIP request > from config > Jun 13 14:41:00 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request > from config > Jun 13 15:40:35 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request > from config > Jun 13 15:40:35 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request > from config > Jun 13 15:40:36 encasip001 /usr/sbin/ser[4807]: lookup failed :SIP request > from config > Jun 13 15:40:36 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request > from config > Jun 13 15:40:36 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request > from config > Jun 13 15:40:36 encasip001 /usr/sbin/ser[4806]: lookup failed :SIP request > from config > Jun 13 15:40:36 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request > from config > Jun 13 15:40:37 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request > from config > Jun 13 15:40:37 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request > from config > Jun 13 15:40:39 encasip001 /usr/sbin/ser[4810]: lookup failed :SIP request > from config > __________________________________________________________________________ > _ > > # > # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ > # > # simple quick-start config script > # > > # ----------- global configuration parameters ------------------------ > > #debug=9 # debug level (cmd line: -dddddddddd) > #fork=yes > #log_stderror=no # (cmd line: -E) > > /* Uncomment these lines to enter debugging mode > debug=7 > fork=no > log_stderror=yes > */ > > check_via=no # (cmd. line: -v) > #dns=no # (cmd. line: -r) > #rev_dns=no # (cmd. line: -R) > > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > #port=5060 > #children=4 > fifo="/tmp/ser_fifo" > > alias=209.233.189.177 > alias=voice.engca.bevocal.com > > > # ------------------ module loading ---------------------------------- > > # Uncomment this if you want to use SQL database > #loadmodule "/usr/lib/ser/modules/mysql.so" > > loadmodule "/usr/lib/ser/modules/sl.so" > loadmodule "/usr/lib/ser/modules/tm.so" > loadmodule "/usr/lib/ser/modules/rr.so" > loadmodule "/usr/lib/ser/modules/maxfwd.so" > loadmodule "/usr/lib/ser/modules/usrloc.so" > loadmodule "/usr/lib/ser/modules/registrar.so" > loadmodule "/usr/lib/ser/modules/exec.so" > loadmodule "/usr/lib/ser/modules/textops.so" > #loadmodule "/usr/lib/ser/modules/acc.so" > > #modparam("acc", "log_level", 2) > #modparam("acc", "log_flag", 1) > #modparam("acc", "report_cancels", 1) > #modparam("acc", "failed_transactions", 1) > > > > # Uncomment this if you want digest authentication > # mysql.so must be loaded ! > #loadmodule "/usr/lib/ser/modules/auth.so" > #loadmodule "/usr/lib/ser/modules/auth_db.so" > > # ----------------- setting module-specific parameters --------------- > > # -- usrloc params -- > > modparam("usrloc", "db_mode", 0) > > # Uncomment this if you want to use SQL database > # for persistent storage and comment the previous line > #modparam("usrloc", "db_mode", 2) > > # -- auth params -- > # Uncomment if you are using auth module > # > #modparam("auth_db", "calculate_ha1", yes) > # > # If you set "calculate_ha1" parameter to yes (which true in this config), > # uncomment also the following parameter) > # > #modparam("auth_db", "password_column", "password") > > # -- rr params -- > # add value to ;lr param to make some broken UAs happy > modparam("rr", "enable_full_lr", 1) > > # ------------------------- request routing logic ------------------- > > # main routing logic > route{ > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) > { > sl_send_reply("483","Too Many Hops"); > break; > }; > if ( msg:len > max_len ) { > sl_send_reply("513", "Message too big"); > break; > }; > > # we record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > # record_route(); > # loose-route processing > if (loose_route()) { > t_relay(); > break; > }; > # if the request is for other domain use UsrLoc > # (in case, it does not work, use the following command > # with proper names and addresses in it) > if (uri==myself) > { > > if(method=="OPTIONS") > { > > sl_send_reply("200", "Ok"); > break; > }; > if (method=="REGISTER") > { > save("location"); > break; > }; > > if(method=="INVITE") > { > log(1,"about lookup in route zero"); > if (!lookup("location","encamed")) { > log(1,"lookup failed :SIP request from config"); > sl_send_reply("404", "Not Found"); > break; > }; > t_on_failure("5"); > }; > }; > > # setflag(1); > # forward to current uri now; use stateful forwarding; that > # works reliably even if we forward from TCP to UDP > if (!t_relay()) { > sl_reply_error(); > }; > > } > > failure_route[5] { > revert_uri(); > if (!lookup("location","encamed")) { > log(1,"lookup failed :SIP request from config"); > t_reply("404", "Not Found"); > break; > }; > append_branch(); > append_urihf("CC-Diversion: ", "\r\n"); > t_on_failure("6"); > t_relay(); > } > > > failure_route[6] { > log(1,"Inside failure_route six"); > exec_msg("printenv|grep '^SIP'|mail -v -s onfailure6 > krcgunta(a)yahoo.com"); > } > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Open letter to the SER Community
by Greger V. Teigre 15 Jun '05

15 Jun '05

Dear Core Developers, Contributors, and Users, I am sure there are many users of SER who now ask the question: What the h... is going and how will this impact SER and how I use SER? I will try to contribute with my interpretation of all this, hopefully to the benefit of those less familiar with SER. I ask you to bare with me, this email is a bit longer than the norm... In open source development, everybody is equal and on the surface have the same motives, but in cases like this, it's useful to know where people are coming from. I, myself, work for a company that uses SER as a component in one of its products. My interests in SER are twofold: 1. The more features and the better open source SER gets, the better for my company 2. The more carrier-grade features (stability, robustness, and scalability), the better When I first started using SER, I realised that #1 was held down by a very high threshold for first use + a sluggishness to include external contributions into the project. I also saw that there was a lack of some carrier-grade features (like load balancing, monitoring, etc). So, this caused me to participate in the Getting Started documentation project, as well as to set up http://onsip.org to lower the getting started threshold and facilitate access to information relevant for carrier-grade setups. And I also volunteered to be the administrator of the new experimental directory for SER modules. Iptelorg.com and Voice System have both established businesses with SER as a core of their business model. Before an open source project gets the momentum (in terms of development community), a company with commercial interests in the open source project is crucial to the development of the project. Once the project gets a life on its own, there will be conflicting interests between the commercial company and the open source project. (Fraunhofer and then) Iptelorg.com has been this company for SER. Iptelorg.com's business model has included building other components/extending SER and offer this as a carrier-grade platform. Of course, they don't want to submit their commercial code to the public CVS, it's the basis for their revenues. However, when the community gets going, their commercial components start popping up with alternative contributions (like TLS). What to do? The reactions can be: 1. Ignore it 2. Submit own commercial code when there is no other choice 3. Pro-actively push commercial code to the open-source project and continously improve own commercial code Voice System has a slightly different business model, but also they have their own commercial components and I believe they focus more on enterprises. Do they contribute their commercial code to OpenSER? Of course not, they have exactly the same problems as iptelorg.com, but they need more features (to enterprises), while iptelorg.com needs stability (to carriers). So, my interpretation of the OpenSER announcement: It's a branch with a different release philosophy, release sooner, bug-fix after release and bring in as much contributions as possible. Daniel-Constantin Mierla states "the SER code maintained by us will go further [...] The cvs was created just to ease the maintainance." However, the communication, both in the announcement, as well as on the website is another with a language clearly indicating that this is *something different from SER at iptel.org*. To the casual user, this means choosing between two projects and also choosing where to contribute. (To me, this means that allthough I would love to get new functionality faster, I cannot possibly use the OpenSER project due to my need for testing and stability.) To the iptelorg.com and Voice System guys: Even though SER originates in your Fraunhofer pet project, SER is no longer yours. Close to 700 people have registered at ONsip.org. There is a large and thriving community of SER users! This gives you, the core developers, a responsibility you didn't have before and you must act accordingly. Jiri and Bogdan-Andrei: I plead you to talk together and resolve this to the benefit of the *community*, not your individual companies. There is no question that the openser.org website brings additional value to the beginner user (documentation + functionality) and that Jan Janak's release philosophy has been a benefit to those of us who rely on stability. Your indifferences MUST be possible to resolve within the context of the community. Jiri: Be open to how contributions and releases can be handled without sacrificing stability. SER needs contributions and Voice System has a point there! Bogdan-Andrei: Try to figure out how you can resolve this without splitting the community in two projects. You really need the other iptel.org contributors in making your fresh releases stable as fast as possible! If this only boils down to different release philosophies and how contributions shall be taken into SER, I have the following suggestions: - Let loose the Voice System guys on releasing new SER versions with new functionality, don't keep functionality in CVS HEAD too long - However, do as ex. OpenLDAP and state CLEARLY which version is considered stable, make sure that source packages are available for all versions - If absolutely necessary, keep two sites: iptel.org and openser.org, but please don't communicate that these are two different projects. If the code base is the same, they are not! Ex. have one site for stable releases and one for newer releases. I will be happy to mediate in the talks if necessary. After all, I'm Norwegian and conflict mediation is one of the few things Norwegians are known for... :-) Best regards, Greger g-) PS! This open letter has also been posted to http://ONsip.org

1 0

[Serusers] Re: Module acc doesn't compile
by Francesco Bottà 15 Jun '05

15 Jun '05

Here the typo error in the Makefile of acc module: # uncomment the next two lines if you wish to enable RADIUS accounting DEFS+=-DRAD_ACC -I$(LOCALBASE)/include LIBS=-L$(LOCALBASE)/lib -l$(RADIUSCLIENT_LIB) Regards, Francesco ----- Original Message ----- From: "Francesco Bottà" <francesco.botta(a)eutelia.it> To: <serdev(a)lists.iptel.org>; <serusers(a)lists.iptel.org> Sent: Wednesday, June 15, 2005 9:26 AM Subject: Module acc doesn't compile Hi all, the module acc. with the latest cvs code of the 0.9.3 doestn't compile: make[1]: Entering directory `/usr/local/src/sip_router/modules/acc' ../../Makefile.rules:80: acc.d: No such file or directory ../../Makefile.rules:80: acc_mod.d: No such file or directory ../../Makefile.rules:80: diam_avp.d: No such file or directory ../../Makefile.rules:80: diam_message.d: No such file or directory ../../Makefile.rules:80: diam_tcp.d: No such file or directory make[1]: Leaving directory `/usr/local/src/sip_router/modules/acc' make[1]: Entering directory `/usr/local/src/sip_router/modules/acc' gcc -fPIC -DPIC -g -Wcast-align -pg -DNAME='"ser"' -DVERSION='"0.9.3"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DEXTRA_DEBUG -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DSQL_ACC -DRAD_ACC -I/usr/local/include -c acc.c -o acc.o acc.c:56:33: radiusclient-ng.h: No such file or directory acc.c:603: error: parse error before "phrase2code" acc.c: In function `phrase2code': acc.c:605: error: `UINT4' undeclared (first use in this function) acc.c:605: error: (Each undeclared identifier is reported only once acc.c:605: error: for each function it appears in.) acc.c:605: error: parse error before "code" acc.c:609: error: `code' undeclared (first use in this function) acc.c: At top level: acc.c:618: error: parse error before "rad_status" acc.c: In function `acc_rad_request': acc.c:640: error: `VALUE_PAIR' undeclared (first use in this function) acc.c:641: error: `UINT4' undeclared (first use in this function) acc.c:641: error: parse error before "av_type" acc.c:655: error: invalid lvalue in assignment acc.c:666: error: `av_type' undeclared (first use in this function) acc.c:764: error: `OK_RC' undeclared (first use in this function) make[1]: *** [acc.o] Error 1 make[1]: Leaving directory `/usr/local/src/sip_router/modules/acc' What's wrong? Many thanx Francesco

1 0

[Serusers] Module acc doesn't compile
by Francesco Bottà 15 Jun '05

15 Jun '05

Hi all, the module acc. with the latest cvs code of the 0.9.3 doestn't compile: make[1]: Entering directory `/usr/local/src/sip_router/modules/acc' ../../Makefile.rules:80: acc.d: No such file or directory ../../Makefile.rules:80: acc_mod.d: No such file or directory ../../Makefile.rules:80: diam_avp.d: No such file or directory ../../Makefile.rules:80: diam_message.d: No such file or directory ../../Makefile.rules:80: diam_tcp.d: No such file or directory make[1]: Leaving directory `/usr/local/src/sip_router/modules/acc' make[1]: Entering directory `/usr/local/src/sip_router/modules/acc' gcc -fPIC -DPIC -g -Wcast-align -pg -DNAME='"ser"' -DVERSION='"0.9.3"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DEXTRA_DEBUG -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DSQL_ACC -DRAD_ACC -I/usr/local/include -c acc.c -o acc.o acc.c:56:33: radiusclient-ng.h: No such file or directory acc.c:603: error: parse error before "phrase2code" acc.c: In function `phrase2code': acc.c:605: error: `UINT4' undeclared (first use in this function) acc.c:605: error: (Each undeclared identifier is reported only once acc.c:605: error: for each function it appears in.) acc.c:605: error: parse error before "code" acc.c:609: error: `code' undeclared (first use in this function) acc.c: At top level: acc.c:618: error: parse error before "rad_status" acc.c: In function `acc_rad_request': acc.c:640: error: `VALUE_PAIR' undeclared (first use in this function) acc.c:641: error: `UINT4' undeclared (first use in this function) acc.c:641: error: parse error before "av_type" acc.c:655: error: invalid lvalue in assignment acc.c:666: error: `av_type' undeclared (first use in this function) acc.c:764: error: `OK_RC' undeclared (first use in this function) make[1]: *** [acc.o] Error 1 make[1]: Leaving directory `/usr/local/src/sip_router/modules/acc' What's wrong? Many thanx Francesco

1 0

[Serusers] Radiusclient NG library 0.5.1 and 0.4.9 has been released
by Maxim Sobolev 15 Jun '05

15 Jun '05

Folks, Due to popular demand I've just made a new releases 0.5.1 and 0.4.9 of Radiusclient NG library, both of them include latest fixes to make proper validation of incoming Radius replies. Also I've created RELENG_0_4 branch just in the case we will need to make some more fixes to 0.4.x series. Therefore please make sure you are using the latest version before reporting any bugs. -Maxim

1 0

[Serusers] openser-0.9.4-fc1.0.i386.rpm
by fedora＠voice.dyndns.tv 15 Jun '05

15 Jun '05

-rw-r--r-- 1 root root 793837 6月 15 12:13 openser-0.9.4-fc1.0.i386.rpm -rw-r--r-- 1 root root 1194843 6月 15 12:13 openser-0.9.4-fc1.0.src.rpm -rw-r--r-- 1 root root 64617 6月 15 12:13 openser-cpl-0.9.4-fc1.0.i386. rpm -rw-r--r-- 1 root root 3472507 6月 15 12:13 openser-debuginfo-0.9.4-fc1.0 .i386.rpm -rw-r--r-- 1 root root 62838 6月 15 12:13 openser-jabber-0.9.4-fc1.0.i3 86.rpm -rw-r--r-- 1 root root 112310 6月 15 12:13 openser-mysql-0.9.4-fc1.0.i38 6.rpm -rw-r--r-- 1 root root 19969 6月 15 12:13 openser-radius-0.9.4-fc1.0.i3 86.rpm -rw-r--r-- 1 root root 5753 6月 15 12:21 openser.spec.fedora /////////////////////////////////////////////////////////////////////// extracting debug info from /var/tmp/openser-0.9.4-root/usr/sbin/openser extracting debug info from /var/tmp/openser-0.9.4-root/usr/sbin/gen_ha1 extracting debug info from /var/tmp/openser-0.9.4-root/usr/sbin/openserunix extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/acc.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/alias_db.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/auth.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/auth_db.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/auth_diameter.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/avpops.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/dbtext.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/dispatcher.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/diversion.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/domain.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/enum.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/exec.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/flatstore.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/gflags.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/group.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/mangler.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/maxfwd.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/mediaproxy.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/msilo.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/nathelper.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/options.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/pa.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/pdt.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/permissions.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/pike.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/registrar.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/rr.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/sl.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/sms.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/speeddial.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/textops.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/tm.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/uri.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/uri_db.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/usrloc.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/xlog.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/mysql.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/jabber.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/cpl-c.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/avp_radius.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/auth_radius.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/group_radius.so extracting debug info from /var/tmp/openser-0.9.4-root/usr/lib/openser/modules/uri_radius.so

1 0

[Serusers] SER with MySql - Error! Need Help!
by Valério Vendrame Costa 14 Jun '05

14 Jun '05

I have one problem to use SER with Mysql. I want to autenticate user by mysql database acount... When I try to start SER, system show this message: Help me please! [root@myserver tester]# /usr/local/sbin/ser -E 0(6250) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/lib/libmysqlclient.so.10: invalid ELF header 0(6250) parse error (31,13-49): failed to load module ERROR: bad config file (1 errors) Before I run Ser I input this commands: [root@myserver tester]# declare -x SIP_DOMAIN="mysertest.com" [root@myserver tester]# /usr/local/sbin/serctl start My config file "ser.cfg" is: # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" alias=mysertest.com # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri=~"mysertest.com") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("mysertest.com", "subscriber")) { www_challenge("mysertest.com", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } If somebody can help me... Thanks! Valerio. _________________________________________________________________ MSN Busca: fácil, rápido, direto ao ponto. http://search.msn.com.br

2 1

[Serusers] problem in ser
by Kamran Ahmad 14 Jun '05

14 Jun '05

hello i have some problems i am using ser-0.9.0 with radiusclient-ng. but unable to compile radius modules. and when i am trying to start ser. and unable to stop ser. or even i cannot kill ser. any comments [root@voice ser-0.9.0]# serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config ----------------------------------------------------- #ser.cfg dns=no rev_dns=no fifo="/tmp/ser_fifo" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock","unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 0) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 1) ----------------------------------------------------- make_and_install_output.txt make[1]: Entering directory `/ser-0.9.0/modules/auth_radius' ../../Makefile.rules:77: authorize.d: No such file or directory ../../Makefile.rules:77: authrad_mod.d: No such file or directory ../../Makefile.rules:77: sterman.d: No such file or directory make[1]: Leaving directory `/ser-0.9.0/modules/auth_radius' make[1]: Entering directory `/ser-0.9.0/modules/auth_radius' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -I/opt/axxany/siprouter/radiusclient/include -DNAME='"ser"' -DVERSION='"0.9.1"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -c authorize.c -o authorize.o gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -I/opt/axxany/siprouter/radiusclient/include -DNAME='"ser"' -DVERSION='"0.9.1"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -c authrad_mod.c -o authrad_mod.o authrad_mod.c: In function `mod_init': authrad_mod.c:110: error: `DICT_VENDOR' undeclared (first use in this function) authrad_mod.c:110: error: (Each undeclared identifier is reported only once authrad_mod.c:110: error: for each function it appears in.) authrad_mod.c:110: error: `vend' undeclared (first use in this function) authrad_mod.c:134: warning: assignment makes pointer from integer without a cast authrad_mod.c:139: error: too many arguments to function `rc_conf_str' authrad_mod.c:139: error: too many arguments to function `rc_read_dictionary' authrad_mod.c:144: warning: implicit declaration of function `rc_dict_findvend' authrad_mod.c:159: error: too many arguments to function `rc_dict_findattr' authrad_mod.c:159: error: too many arguments to function `rc_dict_findval' make[1]: *** [authrad_mod.o] Error 1 make[1]: Leaving directory `/ser-0.9.0/modules/auth_radius' make[1]: Entering directory `/ser-0.9.0/modules/group_radius' ../../Makefile.rules:77: group.d: No such file or directory ../../Makefile.rules:77: grouprad_mod.d: No such file or directory make[1]: Leaving directory `/ser-0.9.0/modules/group_radius' make[1]: Entering directory `/ser-0.9.0/modules/group_radius' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -I/opt/axxany/siprouter/radiusclient/include -DNAME='"ser"' -DVERSION='"0.9.1"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -c group.c -o group.o group.c: In function `radius_is_user_in': group.c:194: warning: passing arg 2 of `rc_avpair_add' makes integer from pointer without a cast group.c:194: warning: passing arg 3 of `rc_avpair_add' makes pointer from integer without a cast group.c:194: warning: passing arg 4 of `rc_avpair_add' makes integer from pointer without a cast group.c:194: error: too many arguments to function `rc_avpair_add' group.c:203: warning: passing arg 2 of `rc_avpair_add' makes integer from pointer without a cast group.c:203: warning: passing arg 3 of `rc_avpair_add' makes pointer from integer without a cast group.c:203: warning: passing arg 4 of `rc_avpair_add' makes integer from pointer without a cast group.c:203: error: too many arguments to function `rc_avpair_add' group.c:209: warning: passing arg 2 of `rc_avpair_add' makes integer from pointer without a cast group.c:209: warning: passing arg 3 of `rc_avpair_add' makes pointer from integer without a cast group.c:209: warning: passing arg 4 of `rc_avpair_add' makes integer from pointer without a cast group.c:209: error: too many arguments to function `rc_avpair_add' group.c:215: warning: passing arg 1 of `rc_auth' makes integer from pointer without a cast group.c:215: warning: passing arg 3 of `rc_auth' from incompatible pointer type group.c:215: warning: passing arg 4 of `rc_auth' from incompatible pointer type group.c:215: error: too many arguments to function `rc_auth' make[1]: *** [group.o] Error 1 make[1]: Leaving directory `/ser-0.9.0/modules/group_radius' __________________________________ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html

2 2

[Users] Re: [Serusers] OpenSER release
by Bogdan-Andrei Iancu 14 Jun '05

14 Jun '05

Thanks Bayan, if you have more patches/idea to integrate in the next release (current cvs head version), feel free to send them to see what about. bogdan Bayan William Towfiq wrote: > Great job on the OpenSER release Bogdan. We hope to assist in the > OpenSER development efforts, a lot of the changes in your list were > patches we had been planning to write or had written already. > > Thanks again, > William > > Bogdan-Andrei Iancu wrote: > >> Dave, >> >> openser is an alternative for people who needs fast progress. There >> are more and more people asking when some feature will be available >> in a stable release or if a contribution will be ever accepted. For >> example our guys came up with a ACC patch to implement custom >> accounting - feature witch was ask by some users. But for almost half >> a year the patch was ignore by module maintainers....the same with >> TLS or other features. OpenSER wants to overcome this problem. And >> btw, the scripting is the same, compatibility will be >> maintained...but with more features and more module. We shouldn't >> confuse interoperability with enhancement. >> >> We are SER people, even core developers of SER, but we cannot impose >> our point of view over the whole project - there are other people >> responsible for other parts of code and I cannot neither go over >> their code, neither to force them to progress. >> It's just a matter of effort versus waisted effort........ >> >> bogdan >> >> Dave wrote: >> >>> I do not think openser is a good idea. This will only >>> lead to further confusion and interoperability >>> problems vis-a-vis ser.cfg. >>> Why can't the openser guys invest their time and >>> efforts to work with ser people rather than forking on >>> a new path? >>> It seems to me voice-systems.ro is trying to endorse >>> itself by launching openser. >>> >>> Dave >>

1 0

[Users] about openser serial forking
by Juha Heinanen 14 Jun '05

14 Jun '05

after reading the documentation it didn't become clear to me if openser tm enhancement "proper serial forking" includes serial forking based on q values, i.e., can i just make one t_relay() call and tm tries the contacts in proper q value order. -- juha

2 3

RE: [Serusers] OpenSER release
by Rose, John 14 Jun '05

14 Jun '05

I agree, openser should be working with the ser people instead of forking it. Why is ser so stable? One of the reasons is that there aren't new releases every month. These guys will create confusion. John -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Dave Sent: Tuesday, June 14, 2005 12:14 PM To: serusers(a)lists.iptel.org Subject: Re: [Serusers] OpenSER release I do not think openser is a good idea. This will only lead to further confusion and interoperability problems vis-a-vis ser.cfg. Why can't the openser guys invest their time and efforts to work with ser people rather than forking on a new path? It seems to me voice-systems.ro is trying to endorse itself by launching openser. Dave --- Andrei Pelinescu-Onciul <andrei(a)iptel.org> wrote: > On Jun 14, 2005 at 20:10, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > wrote: > > Hi SER community, > > > > there are almost two years from the last official > SER release and things > > do not promise too much right now. Not only that > the progress stuck > > somewhere on the way (rel 0.9.0 was started more > than half a year ago), > > but even any attempt to push thing forward seems > to be denied - I tried > > along with Daniel to push the release, but seems > that not everybody > > shares our and comunity's interest regarding the > public part of SER - > > upgrades were rolled back, new software > contributions haven't found > > their way in (like TLS and other new modules), > modules maintained by > > other developers are inaccessible. > > > The release is delayed due to lack of time. > Current show stoppers were me reviewing the whole tcp code (after > finding a minor bug) and some radius makefile problem. > Forking ser is a very bad ideea and your exposed reason are far from > enough to motivate it. > Anyway anybody can cvs co -rrel_0_9_0 . > > > > > > Unfortunately this is not a good environment if we > what to have some > > future progress for SER. And this is the main > reason for starting a new > > project called OpenSER - http://www.openser.org . > > > > It's called open because its most important > attribute is its opening to > > new ideas and contributions, fast developing and > more involvement of the > > comunity. Along with quality, the progress is the > main concern. > > We will continue to support and develop the SER > project as much as so > > far and as much as possible, but OpenSER will give > the liberty for more. > > > > ser just got an experimental module repository for new stuff that is > not tested and/or not reviewed by a core developed (so that it can be > added to the ser main repository). > > > OpenSER serves the interest of all SER users and > will not change its > > purpose - as a fact I have the pleasure to > announce its first release - > > OpenSER 0.9.4. The web site offers a comprehensive > listing of new > > features and fixes - > http://www.openser.org/index.php#features. For > > people already familiar to SER 0.9.3, going to > > http://www.openser.org/diffs-0.9.0.php will be > more helpful. > > Some of the changes listed in the diffs will break compatibility with > current ser configuration scripts. I wonder also when have you tested > all your changes. > > > > Andrei > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > __________________________________ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] SER - Not listening on any interface ?
by Amith 14 Jun '05

14 Jun '05

hi all, i have installed SER from src. using the commands 1) # make prefix=/ all 2) #make prefix=/ install Now when i run SER , it shows me the different interfaces it is listening to . But , when i try to register i get an ICMP destination port unreachable error. I observe using #netstat -l that SER is not listening on any interface at port 5060. This (5060) port is not used by any other app . Please let me know why is it so ? and how to set this right ? best, Amith --------------------------------- Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more.

4 4

[Serusers] SER 0.9.3 crashing on Solaris 10 (sparc 64bit) server
by lists 14 Jun '05

14 Jun '05

Hi, I have migrated my SER proxy from Linux servers to Sun Solaris 10 (sparc 64 bit). Same SER version (0.9.3) and config file work perfectly fine under Linux and now its crashing. Every time SER needs to fork of a new branch (append_branch();) it core dumps. any idea why this would be happening? thanks fil here is the portion of ser.cfg which is crashing: failure_route[4] { log (1, "**************** FAILURE_ROUTE[4] CALLING VOICEMAIL\n"); # forward to voicemail now if (isflagset(4)){ rewritehostport("192.168.0.5:5060"); append_branch(); route(1); }; } here is the the SER startup with debuging set: 0(17715) failure routing table 4: 0(17715) log( 0(17715) 1 0(17715) , "**************** FAILURE_ROUTE[4] CALLING VOICEMAIL " 0(17715) ); 0(17715) if ( 0(17715) isflagset( 0(17715) 4 0(17715) ); 0(17715) isflagset( 0(17715) 4 0(17715) ); 0(17715) ) { 0(17715) sethostport( 0(17715) "192.168.0.5:5060" 0(17715) ); 0(17715) append_branch( 0(17715) Segmentation fault here is the output of gdb run: gdb -c /var/core/core.ser.17722 /usr/local/sbin/ser GNU gdb 6.3 Copyright 2004 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "sparc-sun-solaris2.10"... Core was generated by `ser'. Program terminated with signal 11, Segmentation fault. Reading symbols from /lib/sparcv9/libdl.so.1... warning: Lowest section in /lib/sparcv9/libdl.so.1 is .dynamic at 00000000000000e8 done. Loaded symbols for /lib/sparcv9/libdl.so.1 Reading symbols from /lib/sparcv9/libresolv.so.2...done. Loaded symbols for /lib/sparcv9/libresolv.so.2 Reading symbols from /lib/sparcv9/libxnet.so.1...done. Loaded symbols for /lib/sparcv9/libxnet.so.1 Reading symbols from /lib/sparcv9/libnsl.so.1...done. Loaded symbols for /lib/sparcv9/libnsl.so.1 Reading symbols from /lib/sparcv9/librt.so.1...done. Loaded symbols for /lib/sparcv9/librt.so.1 Reading symbols from /lib/sparcv9/libc.so.1...done. Loaded symbols for /lib/sparcv9/libc.so.1 Reading symbols from /lib/sparcv9/libsocket.so.1...done. Loaded symbols for /lib/sparcv9/libsocket.so.1 Reading symbols from /lib/sparcv9/libaio.so.1...done. Loaded symbols for /lib/sparcv9/libaio.so.1 Reading symbols from /lib/sparcv9/libmd5.so.1...done. Loaded symbols for /lib/sparcv9/libmd5.so.1 Reading symbols from /platform/SUNW,UltraAX-i2/lib/sparcv9/libc_psr.so.1...done. Loaded symbols for /platform/SUNW,UltraAX-i2/lib/sparcv9/libc_psr.so.1 Reading symbols from /usr/local/lib/ser/modules/mysql.so...done. Loaded symbols for /usr/local/lib/ser/modules/mysql.so Reading symbols from /usr/local/mysql/lib/mysql/libmysqlclient.so.12...done. Loaded symbols for /usr/local/mysql/lib/mysql/libmysqlclient.so.12 Reading symbols from /usr/local/lib/libz.so...done. Loaded symbols for /usr/local/lib/libz.so Reading symbols from /usr/lib/sparcv9/libcrypt_i.so.1...done. Loaded symbols for /usr/lib/sparcv9/libcrypt_i.so.1 Reading symbols from /lib/sparcv9/libgen.so.1...done. Loaded symbols for /lib/sparcv9/libgen.so.1 Reading symbols from /lib/sparcv9/libm.so.2...done. Loaded symbols for /lib/sparcv9/libm.so.2 Reading symbols from /usr/local/gcc/lib/sparcv9/libgcc_s.so.1...done. Loaded symbols for /usr/local/gcc/lib/sparcv9/libgcc_s.so.1 Reading symbols from /lib/sparcv9/libscf.so.1...done. Loaded symbols for /lib/sparcv9/libscf.so.1 Reading symbols from /lib/sparcv9/libdoor.so.1...done. Loaded symbols for /lib/sparcv9/libdoor.so.1 Reading symbols from /lib/sparcv9/libuutil.so.1...done. Loaded symbols for /lib/sparcv9/libuutil.so.1 Reading symbols from /lib/sparcv9/libmp.so.2...done. Loaded symbols for /lib/sparcv9/libmp.so.2 Reading symbols from /platform/SUNW,UltraAX-i2/lib/sparcv9/libmd5_psr.so.1...done. Loaded symbols for /platform/SUNW,UltraAX-i2/lib/sparcv9/libmd5_psr.so.1 Reading symbols from /usr/local/lib/ser/modules/sl.so...done. Loaded symbols for /usr/local/lib/ser/modules/sl.so Reading symbols from /usr/local/lib/ser/modules/tm.so...done. Loaded symbols for /usr/local/lib/ser/modules/tm.so Reading symbols from /usr/local/lib/ser/modules/rr.so...done. Loaded symbols for /usr/local/lib/ser/modules/rr.so Reading symbols from /usr/local/lib/ser/modules/maxfwd.so...done. Loaded symbols for /usr/local/lib/ser/modules/maxfwd.so Reading symbols from /usr/local/lib/ser/modules/usrloc.so...done. Loaded symbols for /usr/local/lib/ser/modules/usrloc.so Reading symbols from /usr/local/lib/ser/modules/registrar.so...done. Loaded symbols for /usr/local/lib/ser/modules/registrar.so Reading symbols from /usr/local/lib/ser/modules/auth.so...done. Loaded symbols for /usr/local/lib/ser/modules/auth.so Reading symbols from /usr/local/lib/ser/modules/auth_db.so...done. Loaded symbols for /usr/local/lib/ser/modules/auth_db.so Reading symbols from /usr/local/lib/ser/modules/acc.so...done. Loaded symbols for /usr/local/lib/ser/modules/acc.so Reading symbols from /usr/local/lib/ser/modules/exec.so...done. Loaded symbols for /usr/local/lib/ser/modules/exec.so Reading symbols from /usr/local/lib/ser/modules/group.so...done. Loaded symbols for /usr/local/lib/ser/modules/group.so Reading symbols from /usr/local/lib/ser/modules/print.so...done. Loaded symbols for /usr/local/lib/ser/modules/print.so Reading symbols from /usr/local/lib/ser/modules/textops.so...done. Loaded symbols for /usr/local/lib/ser/modules/textops.so Reading symbols from /usr/local/lib/ser/modules/uri.so...done. Loaded symbols for /usr/local/lib/ser/modules/uri.so Reading symbols from /usr/local/lib/ser/modules/uri_db.so...done. Loaded symbols for /usr/local/lib/ser/modules/uri_db.so Reading symbols from /usr/local/lib/ser/modules/mediaproxy.so...done. Loaded symbols for /usr/local/lib/ser/modules/mediaproxy.so Reading symbols from /usr/local/lib/ser/modules/domain.so...done. Loaded symbols for /usr/local/lib/ser/modules/domain.so #0 0xffffffff7e8371f0 in strlen () from /lib/sparcv9/libc.so.1 (gdb) backtrace #0 0xffffffff7e8371f0 in strlen () from /lib/sparcv9/libc.so.1 #1 0xffffffff7e89f968 in _ndoprnt () from /lib/sparcv9/libc.so.1 #2 0xffffffff7e8a1e40 in vfprintf () from /lib/sparcv9/libc.so.1 #3 0x000000010000e4b8 in dprint (format=0x1000a6140 "\"%s\"") at dprint.c:66 #4 0x000000010003d0c8 in print_action (a=0x100210380) at route_struct.c:378 #5 0x000000010003d5c8 in print_action (a=0x100210420) at route_struct.c:392 #6 0x000000010003b9f4 in print_rl () at route.c:782 #7 0x000000010001f804 in main (argc=1822720, argv=0xffffffff7ffff2a8) at main.c:1418 (gdb)

2 1

[Serusers] Radius question
by Alberto Cruz 14 Jun '05

14 Jun '05

Hi guys, I have a doubt. Is it possible to use radius related modules with out using db related modules at all? Which are going to be the side effects If I don't use db modules? For example I tried to load mediaproxy module and it is claiming for commands like "is_uri_host_local()" or "is_from_local()" in order to be load it but these commands are only available if I load the db modules. Who can help me to clarify the relationship between radius modules, mediaproxy module and db modules? Regards Alberto Cruz

2 2

[Serusers] [Fwd: TeleNova CDRS to Bestcom.xls]
by Rodrigo P. Telles 14 Jun '05

14 Jun '05

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Saudações, Conforme solicitei, o Fernando da Telenova me mandou os CDR's das ligações que efetuamos. 1a coisa que observei, fizemos mais ligações do que ele me enviou. No geral, somente uma ligação teve o tempo de duração menor no nosso sistema do que no deles (10 segundos), no restante a ligação tem tempo maior no nosso sistema (normal). Abraço, sem mais. - -- ============================================ Rodrigo P. Telles <telles(a)devel.it> Diretor de Tecnologia Devel-IT - http://www.devel.it IVOZ # 1029 +55 14 3324-1200 Grupo Bestcom ============================================ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFCrvxbiLK8unYgEMQRAjukAJ9kDpyAgDKUZafhozsrNlYvErpSpgCaA2Hg SgXkUDFp/l8lmJsKGWQ1NwY= =1YlC -----END PGP SIGNATURE-----

2 2

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users