- sr-users - kamailio.org

[Serusers] ser+gsm authentication
by jeff kwong 12 Jul '05

12 Jul '05

Hi guys! I want to try out the ser and kphone gsm authentication proof of concept in cvs. http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/gsm_auth/ However the link to the kphone source with gsm authentication can no longe be found. http://www.iptel.org/~dcm/ser-gsm/kphone-3.11.1-gsm.tgz Maybe someone here was able to have a copy of the file. Could you kindly send me a copy. I tried using kphone 3.11 but itself even without the patch can't register. It my traces it seems that kphone listens on port 5060 but sends request on another port. Thus when replies are sent on the port used in sending, kphone wasn't able to recieve it. Btw, do you guys know where I could get a decent getting started guide on ser with radius? The onsip manuals have missing pages regarding integrating radius. Thanks in advance for whatever information or ideas! _jeff

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[Serusers] Is mediaproxy of SER proxy or full proxy?
by 皮面 12 Jul '05

12 Jul '05

I'm a fresher and I was wondering which one the mediaproxy module in SER adopt to realize NAT traversal,proxy or full proxy?Thanks! --------------------------------- DO YOU YAHOO!? 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱

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[Serusers] Add prefixes to outgoing numbers
by andrew matthews 12 Jul '05

12 Jul '05

Is there a way to add prefixes to outgoing numbers before they get send to the rewritehost? I need to take a number like 800-555-1212 and change it into 3338005551212 thanks Andrew

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FW: Re: [Serusers] Module ACC_RADIUS problem
by Maximo Marin 11 Jul '05

11 Jul '05

Thanks, i am receiving some starts and stops. Now, is it possible to receive one start and one stop when calling sip to sip, sip to pstn and pstn to sip?? I am not getting the start when calling sip to pstn, i am stuck and frustrated. Maxi. >From: Jan Janak <jan(a)iptel.org> >To: Maximo Marin <maxi_tech(a)hotmail.com> >CC: serusers(a)lists.iptel.org >Subject: Re: [Serusers] Module ACC_RADIUS problem >Date: Fri, 8 Jul 2005 21:12:47 +0200 > >On 08-07-2005 10:30, Maximo Marin wrote: > > Hello everyone, i am running, ser 0.9.3 on debian stable and get this >error: > > > > Jul 8 10:21:41 sip_server ser: set_mod_param_regex: No module matching > > acc_radius found | > > Jul 8 10:21:41 sip_server ser: parse error (55,19-20): Can't set module > > parameter > > > > I have this in line 55 of my ser.cfg > > > > modparam("acc_radius", "radius_flag", 1) > > Please use modparam("acc", "radius_flag", 1) instead. The real name of > the module is acc, just the file is named acc_radius. > > Jan. _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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RE: RE: [Serusers] Try install Classkit extension
by Gareth C. Fowler 11 Jul '05

11 Jul '05

My classkit.so is default located in /usr/lib/php/modules I'm running Fedora Core 4, Apache 2.0.54, PHP 5.0.4. G. -----Original Message----- From: GeoTel [mailto:geotelcom@yahoo.fr] Sent: 12 July 2005 06:51 To: Gareth C. Fowler Subject: Re: RE: [Serusers] Try install Classkit extension Thanks Gareth but I did it as I say in php.ini. Could you check for me where is your "classkit.so" installed ? My is in '/var/tmp/pear-build-root/install-classkit-0.4//usr/local/lib/php/extensions /no-debug-non-zts-20041030/classkit.so' A little bid strange but no error. ---------------------------------------------------------------------------- -- Hi, I had the same problem. Check your php.ini config and add the following: extension=classkit.so It worked for me! G. Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of GeoTel Sent: 12 July 2005 06:29 To: serusers(a)lists.iptel.org Subject: [Serusers] Try install Classkit extension Dear team, Still having the same problem with classkit, same error: "Function aggregate_methods() doesn't exists. "Try" install Classkit extension...." yet, pear install OK Installing '/var/tmp/pear-build-root/install-classkit-0.4//usr/local/lib/php/extensions /no-debug-non-zts-20041030/classkit.so' install ok: classkit 0.4 and "php.ini" ...................................... ; Directory in which the loadable extensions (modules) reside. extension_dir = "/usr/local/lib/php/extensions/" extension=classkit.so Any info, Thanks George ---------------------------------------------------------------------------- --

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[Serusers] Try install Classkit extension
by GeoTel 11 Jul '05

11 Jul '05

Dear team, Still having the same problem with classkit, same error: "Function aggregate_methods() doesn't exists. "Try" install Classkit extension...." yet, pear install OK Installing '/var/tmp/pear-build-root/install-classkit-0.4//usr/local/lib/php/extensions/no-debug-non-zts-20041030/classkit.so' install ok: classkit 0.4 and "php.ini" ..................................... ; Directory in which the loadable extensions (modules) reside. extension_dir = "/usr/local/lib/php/extensions/" extension=classkit.so Any info, Thanks George

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[Serusers] Session-Timers for Prepaid
by Daryl Sanders 11 Jul '05

11 Jul '05

I know there was some discussion in the recently about using session-timers as a method for doing prepaid. I'm just wondering if anyone has actually played around with this, and what your experiences were? My PSTN Gateways support session-timers, so I was thinking about trying this. - Daryl

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[Serusers] Call terminates after 30 seconds
by Luca Corti 11 Jul '05

11 Jul '05

Hello, I'm using a Zyxel 2000W Wireless sip phone behind NAT (cisco 1721) which registers to ser 0.8.14 using rtpproxy. Outbound calls work flawlessly, but inbound calls are terminated after 30 seconds. Do you have any hints why this happens? thanks -- Luca Corti PGP Key ID 1F38C091

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[Serusers] contact information
by Sebastian Kühner 11 Jul '05

11 Jul '05

Hello, I'm using a Sipura device to register to my ser server. When I use a Softphone, the contact-information in the location table is OK, but when I use my Sipura device, ser saves the contact information with the internal IP address of the sipura. Obviously ser doesn't find it when I want to make a call to a Sipura-Phone. Any idea what might be the problem? THanks! Sebastian

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[Serusers] Strange REGISTER Problems
by Ozan Blotter 11 Jul '05

11 Jul '05

Dear List, I can register my ATA'S (Some are NATted and some ara Port Forwarded) to SER (on Public IP) and i use MySQL auth/acc together with Mediaproxy. All users can talk to others, and call them , receive calls from others no problem. But most of the time, they seem unregistered and this happens suddenly, i'm sure that Internet Connection is fine with SER Server and also clients are connected through Cable Modem or ADSL Connection. I unplug or unpower some ATA's, wait a bit like 30 minutes, the devices becomes cold or what's happening else, they can register and works perfect... But when i connect them for hours, they unregister themselves, i do not know what happens... I've tried both Zyxel P2002's with several firmwares, Zyxel P2000Ws, Cisco ATA186's and some LinkSYS Pap2-NA's but all same... Is there any chance to be something wrong in SER settings or Operating System ? Thanks, Ozan

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Re: [Serusers] serweb login
by Chris Mason (Lists) 11 Jul '05

11 Jul '05

harry gaillac wrote: >hello, > >use database apache serweb logs to find problem >--- Chris Mason <lists(a)masonc.com> a écrit : > > Harry: I found the problem as shown below. The applicication is reuqesting a column that does not exist in the database. I checked the contents of scripts/ser_mysql.sh from ser-0.9.3_src.tar.gz <ftp://ftp.berlios.de/pub/ser/0.9.3/src/ser-0.9.3_src.tar.gz> and find that column does not exist in the subscriber table. There's no way I can get it to work. Is the ser distribution broken or the serweb? Chris Query as executed by user ser ####################################################################### mysql> select phplib_id from subscriber where username='admin' and password='heslo' and perms='admin' and domain='mason.home' -> ; ERROR 1054 (42S22): Unknown column 'perms' in 'where clause' ####################################################################### create table statement from scripts/ser_mysql.sh ####################################################################### # # Table structure for table 'subscriber' -- user database # CREATE TABLE subscriber ( phplib_id varchar(32) NOT NULL default '', $USERCOL varchar(64) NOT NULL default '', domain varchar(128) NOT NULL default '', password varchar(25) NOT NULL default '', first_name varchar(25) NOT NULL default '', last_name varchar(45) NOT NULL default '', phone varchar(15) NOT NULL default '', email_address varchar(50) NOT NULL default '', datetime_created datetime NOT NULL default '0000-00-00 00:00:00', datetime_modified datetime NOT NULL default '0000-00-00 00:00:00', confirmation varchar(64) NOT NULL default '', flag char(1) NOT NULL default 'o', sendnotification varchar(50) NOT NULL default '', greeting varchar(50) NOT NULL default '', ha1 varchar(128) NOT NULL default '', ha1b varchar(128) NOT NULL default '', allow_find char(1) NOT NULL default '0', timezone varchar(128) default NULL, rpid varchar(128) default NULL, domn int(10) default NULL, uuid varchar(64) default NULL, UNIQUE KEY phplib_id (phplib_id), PRIMARY KEY ($USERCOL, domain), KEY user_2 ($USERCOL) ) $TABLE_TYPE; -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla(a)yahoo.com

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[Serusers] SEMS
by Daniel Poulsen 11 Jul '05

11 Jul '05

I've been using asterisk for voicemail, but I'm curious about peoples' opinions about SEMS since 0.9.0. It would be nice to run something more tightly integrated with SER, but I don't want to lose the flexibility that asterisk offers. Opinions? Dan

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[Serusers] 0.9.3 dependencies (SER is crashing)
by Steve Blair 11 Jul '05

11 Jul '05

Hello: We recently upgraded to RedHat ES 4 and SER 0.9.3. The upgrade went well however SER runs ok for a few hours then crashes. We have not been able to identify the source of the problem. The obvious thing would be to downgrade the OS but that is also the least desirable choice. Is there a list of packages that SER is dependent upon? I'm suspecting a MySQL problem but I cannot confirm this. Right now we have the following MySQL packages installed: mysql-4.1.10a-2.RHEL4.1 MySQL-shared-compat-4.0.18-0 perl-DBD-MySQL-2.9004-3.1 php-mysql-4.3.9-3.6 mysql-server-4.1.10a-2.RHEL4.1 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

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Re: [Serusers] NAT - Lots of flavours..
by Ricardo Poppi 11 Jul '05

11 Jul '05

Hi all. Thanks to Jan and Andres for the help. --- ------ Things are working better now. I did some tests and now I´m sure. Asterisk don´t give a dam to "a=direction:active". It only cares about the NAT="yes" parameter in "sip.conf" file - or sip_buddies table for realtime users... - For me its ok. My SER/Asterisk host has a public IP. So, it works. I could debug sip and rtp and I saw it erroring "could not find network to send RTP" and when the UA send it first packet...TA-DA!! Everything works fine! RTP media path goes like oil! ---------------------------------- I disabled the SDP fix from my "ser.cfg" so I can make my NATed UAs - behind the same NAT - talk to each other and establish the media path based on internal values that, in this case, are true values. To fix when a public UA calls a NATed one, I use a t_on_reply to fix the 200-0K of all UAs that are being called from public UAs. When the INVITE cames from a NATed UA, I use another route - for t_on_reply() - and it don´t fix the 200-OK and my first case - media between NATed - keeps working... Ok. But when a NATed UA from ANOTHER NATED NETWORK - calls a NATed UA from my network I got problems. Problems because my "ser.cfg" won´t deal with SDP values and the internal IP:Port for media path will be kept. With this, I will never be conversation between then. DO ANYBODY HAS ANY IDEA/PRODUCTION CASE TO DEAL WITH THAT? ---------------------------------------------------------- Thanks in advance. Ricardo Poppi. Date: Fri, 8 Jul 2005 20:51:58 +0200 From: Jan Janak <jan(a)iptel.org> Subject: Re: [Serusers] NAT - Lots of flavours... To: Ricardo Poppi <rpoppi77(a)giro.com.br> Cc: serusers(a)lists.iptel.org, sobomax(a)portaone.com Message-ID: <20050708185158.GY6497(a)localhost.localdomain> Content-Type: text/plain; charset=iso-8859-2 Hello, comments inline. On 04-07-2005 19:44, Ricardo Poppi wrote: >> >> Hi list, >> >> I´m trying to put to work a NATed environment and want to share some >> information and request some I don´t realized yet. >> >> I use an asterisk gateway, with a public IP, working really fine for UAs >> with public IPs. At the same machine I runs SER that receives all SIP >> messages and handle when it should go to a SIP UA or to asterisk, >> rewriting the port (to the one asterisk uses) and sending to it. I don´t >> replicate register to asterisk, and use the user accounts as "peer", >> instead of "friends". >> >> My ser.cfg is using the "force_rport()" and "fix_nated_contact()" for >> every REGISTER it receives from nat UAs - I know when it comes from a >> NATed UA using nat_uac_test("2"). >> >> Every INVITE that comes from NATed UA passes through a >> "fix_nated_sdp("2"), that rewrites the IP address of SDP headers. Using >> a onreply route I fix the 200 OK INVITE message, just in case that the >> NATed UA is on the called side. >> >> The UAs I´m using are X-Lite, Clipcomm CP-100 IP Phone, and Grandstream >> HT-488. >> >> >> Below I wrote the different kinds of configuration into the UA and in >> ser.cfg, and the results I got: >> >> >> 1) Using without touching the UA - It don´t know it is a NATed UA. >> ----------------------------------------------------------------------------------------------------------------------------- >> >> All REGISTER are treated ok because the force_rport make SER respond to >> the register on the same external IP:Port it received. On the same hand, >> it stores the right URI into the location database making the UA receive >> the subsequent INVITES or other SIP messages through the external IP:Port. >> >> The INVITES that comes from NATed UA have their SDP IP address rewriten >> by SER and the external IP takes place. But the port is kept the >> internal value, so when the called UA tries to reach the >> External_IP:Internal_port the NAT/Firewall probably block/drops the >> packets, and the result is a one-way audio - The one-way audio is >> probably due to the right value that comes from the SDP headers of the >> called UA - asterisk -, that has a public IP. > > I just quickly looked into nathelper sources and it looks like it can only rewrite the port number in SDP if you run force_rtp_proxy, fix_nated_sdp seems to change the IP address only. I CCed to Maxim, maybe he could clarify this better than myself. >> 2) a=direction:active >> ---------------------------------- >> >> If I add into ser.cfg a "fix_nated_sdp("1")" command, it will add the >> "a=direction:active" parameter to SDP header of INVITE that comes from >> NATed UAs. I saw that it´s happening but the asterisk seems to not >> understand that and don´t expect for the first RTP packet to get the >> IP:Port information of the media. A one-way audio is the result of that. >> The asterisk is probably sending RTP packets to the >> Ext_IP:Internal_port, and the firewall is blocking the packets. > > If asterisk does not support symmetric RTP then you will have to put the rtpproxy between the user agent and asterisk and call force_rtp_proxy instead of fix_nated_sdp in the script. I am not sure if I remember it correctly, but I think that asterisk should support symmetrict rtp, so maybe the problem is in fix_nated_sdp function which does not alter media ports. --------- Does anyone on the list know if asterisk supports symmetric RTP ? In other words, can asterisk interpret a=direction:active parameter from SDP and send media to the source IP and port of the incoming media stream, instead of the IP and port from SDP ? --------- You can also try to put rtpproxy between user agents and asterisk and call force_rtp_proxy instead fix_nated_sdp. It's not the best solution, but this way you could verify if the problem is in unaltered port number in SDP. Jan.

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Re: [Serusers] serweb: "DB Error: no such table"
by Karel Kozlik 11 Jul '05

11 Jul '05

Hello, plase (CC) the list in your replies. serweb-0.9.3 match tables of ser-0.9.3 But there are a little confusion in db schema of ser-0.9.3 and ser-cvs. Some tables have diferent names. serweb-cvs should also match tables of ser-cvs, but I didn't check it in last days. Note that cvs version is not oficial release. And is not recomended to mix develeper version of serweb from cvs and oficial release of ser. Karel harry gaillac napsal(a): > Hello Karel, > > why serweb-0.9.3 don't match ser tables? > > I think you read my posts > > Harry > --- Karel Kozlik <karel(a)iptel.org> a écrit : > > >>Hello, >>there are two ways: >>- install serweb 0.9.3 (recomended) >>- change cofiguration of your version of serweb to >>fit db schema of >> ser 0.9.3. Change the names of tables in >>config_data_layer.php >> >>Karel >> >>Timur V. Elzhov napsal(a): >> >>>Dear SER experts, >>> >>>I've downloaded and configured SER v0.9.3, as well >> >>as serweb, the >> >>>head CVS version. I succeeds by serweb with >> >>subscribtion, logging >> >>>on, and changing my own details. But the red >> >>message appears on my >> >>>user page: >>> >>> "DB Error: no such table". >>> >>>However, again, there is now problem with actions >> >>mentioned just above. >> >>>I'm not familiar with PHP, but I found that many >> >>php files under >> >>>"serweb/data_layer/" directory contain the same >> >>"if" statement comprising >> >>>the "DB_ERROR_NOSUCHTABLE" constant: >>> >>> if (DB::isError($res)) { >>> if ($res->getCode()==DB_ERROR_NOSUCHTABLE) >> >>return true; //expected, table mayn't exist in >>installed version >> >>> else {log_errors($res, $errors); return >> >>false;} >> >>> } >>> >>>- it may be useful ... >>> >>>How should I locate & resolve this problem? >>> >>>Much thanks. >>> >>> >> >>_______________________________________________ >>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

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[Serusers] Problems with my SER config
by Chris Doyle 11 Jul '05

11 Jul '05

Hi there guys, This is my first question to the list, hopefully it will be a painless issue for everyone else :) I am trying to configure SER to have my users register to it and be able to call themselves by username (SIP/cdoyle@ser), and also call PSTN numbers and have them forwarded to an asterisk PBX for PSTN termination. Everything seems to be setup properly to me, but when I try to make a call to the PSTN, I get "DEBUG: sl_filter_ACK : local ACK found -> dropping it!" My config file is at http://caves.narshe.net/ser.cfg Thanks Chris Doyle

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[Serusers] RPID privacy
by Daniel Poulsen 11 Jul '05

11 Jul '05

Hi, I am trying to give my users the option of disabling caller ID if they so choose by dialing a code. I can tack on rpid using append_rpid_hf() just fine, but for some reason my privacy flags seem to be ignored. Perhaps I am not processing them properly? One observation: I've noticed most people on the list using the format: append_rpid_hf("<sip:","@localhost; user=phone>;party=calling;screen=no;privacy=full") When use this format my tcpdumps show: Remote-Party-ID: <sip:sip:2125551212@my.sip-domain.com@localhost; user=phone>;party=calling;screen=no;privacy=full") I have to use: append_rpid_hf("<","; user=phone>;party=calling;screen=no;privacy=full") ...to get it to look correct in the headers. This strikes me as odd that everyone else uses the former and I must use the latter. I am using ser-0.9.3 (Soon to upgrade to 0.9.3!!) and the gateway is a Cisco AS5350. I've looked as Cisco DOCS and the header seems to be structured properly. I'm sure it is something stupid, but I am scratching my head on this one. Any advice? Thank you. Dan

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[Serusers] Append_hf not working for invite requests
by Singh, Amit 11 Jul '05

11 Jul '05

Hi, I have added the following lines in my cfg file: If(uri==myself) { .... if (method=="INVITE" && t_newtran()) { if (uri=~"^sip:(.+@)?(10\.1\.100\.10)([:;\?].*)?$") { append_hf("Retry-After: 120"); t_reply("480", "Stateful 480 Reply"); } else { sl_send_reply("480", "Stateless 480 Reply"); }; break; }; Now, I am expecting that when I get the response back from SER, the header should contain REPLY-AFTER header. But it does not. Any Ideas as to why the REPLY-AFTER is not appended to the message header? Here is the transaction: 10.1.100.10:5060 10.1.0.149:3490 | | <Call><PFrame><Time> | | |<--------------- REGISTER F1<| 1 PF:50 10:06:46.3533 | | |>F2 200 OK ----------------->| 1 PF:51 10:06:46.3557 | | |<------------(sdp) INVITE F3<| 2 PF:55 10:07:31.3778 | | |>F4 480 Stateful 480 Reply ->| 2 PF:56 10:07:31.4967 | | |>F5 480 Stateful 480 Reply ->| 2 PF:57 10:07:31.5715 | | |<-------------------- ACK F6<| 2 PF:58 10:07:33.5186 | | |<------------(sdp) INVITE F7<| 3 PF:167 10:26:51.5746 | | |>F8 480 Stateful 480 Reply ->| 3 PF:170 10:26:51.5939 | | |>F9 480 Stateful 480 Reply ->| 3 PF:171 10:26:52.3072 | | |<------------------- ACK F10<| 3 PF:172 10:26:53.5354 | | |<------------------- ACK F11<| 3 PF:173 10:26:53.7753 ======================================================================== ======== SIP MESSAGE 1 10.1.0.149:3490() -> 10.1.100.10:5060() UDP Frame 50 11/Jul/05 10:06:46.3533 TimeFromPreviousSipFrame=490.6428 TimeFromStart=490.6428 REGISTER sip:10.1.100.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.149:3490;branch=z9hG4bK944888c1.10.1.0.149.3490 From: "alice" <sip:alice@10.1.100.10>;tag=214fc363.10.1.0.149.3490 To: "alice" <sip:alice@10.1.100.10> Call-ID: f53fe6c39e6fcc1d(a)10.1.0.149.3490 CSeq: 1 REGISTER Contact: "alice" <sip:alice@10.1.0.149:3490;transport=udp>;expires=7200;methods="INVITE, OPTIONS, BYE, CANCEL, ACK, SUBSCRIBE, NOTIFY";+g.poc.talkburst max-forwards: 70 supported: pref Content-Length: 0 ======================================================================== ======== SIP MESSAGE 2 10.1.100.10:5060() -> 10.1.0.149:3490() UDP Frame 51 11/Jul/05 10:06:46.3557 TimeFromPreviousSipFrame=0.0024 TimeFromStart=490.6452 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.149:3490;branch=z9hG4bK944888c1.10.1.0.149.3490 From: "alice" <sip:alice@10.1.100.10>;tag=214fc363.10.1.0.149.3490 To: "alice" <sip:alice@10.1.100.10>;tag=b27e1a1d33761e85846fc98f5f3a7e58.ed9f Call-ID: f53fe6c39e6fcc1d(a)10.1.0.149.3490 CSeq: 1 REGISTER Contact: <sip:alice@10.1.0.180:3490;transport=udp>;expires=6617, <sip:alice@10.1.0.149:3490;transport=udp>;expires=7200 Server: Sip EXpress router (0.8.99-dev2 (i386/linux)) Content-Length: 0 Warning: 392 10.1.100.10:5060 "Noisy feedback tells: pid=14070 req_src_ip=10.1.0.149 req_src_port=3490 in_uri=sip:10.1.100.10 out_uri=sip:10.1.100.10 via_cnt==1" ======================================================================== ======== SIP MESSAGE 3 10.1.0.149:3490() -> 10.1.100.10:5060() UDP Frame 55 11/Jul/05 10:07:31.3778 TimeFromPreviousSipFrame=45.0221 TimeFromStart=535.6673 INVITE sip:error480@10.1.100.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.149:3490;branch=z9hG4bK411c33ff.10.1.0.149.3490 From: "alice" <sip:alice@10.1.0.149>;tag=369b1fba.10.1.0.149.3490 To: <sip:error480@10.1.100.10> Call-ID: 66fcd326(a)10.1.0.149.3490 CSeq: 1 INVITE Contact: "alice" <sip:alice@10.1.0.149:3490>;+g.poc.talkburst max-forwards: 70 accept-contact: *;+g.poc.talkburst;require;explicit Content-Type: application/sdp Content-Length: 131 v=0 o=- 1113854502 1113854502 IN IP4 10.1.0.149 s=- c=IN IP4 10.1.0.149 t=0 0 m=audio 1000 RTP/AVP 98 a=rtpmap:98 AMR/16000 ======================================================================== ======== SIP MESSAGE 4 10.1.100.10:5060() -> 10.1.0.149:3490() UDP Frame 56 11/Jul/05 10:07:31.4967 TimeFromPreviousSipFrame=0.1190 TimeFromStart=535.7863 SIP/2.0 480 Stateful 480 Reply Via: SIP/2.0/UDP 10.1.0.149:3490;branch=z9hG4bK411c33ff.10.1.0.149.3490 From: "alice" <sip:alice@10.1.0.149>;tag=369b1fba.10.1.0.149.3490 To: <sip:error480@10.1.100.10>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-d446 Call-ID: 66fcd326(a)10.1.0.149.3490 CSeq: 1 INVITE Server: Sip EXpress router (0.8.99-dev2 (i386/linux)) Content-Length: 0 Warning: 392 10.1.100.10:5060 "Noisy feedback tells: pid=14072 req_src_ip=10.1.0.149 req_src_port=3490 in_uri=sip:error480@10.1.100.10 out_uri=sip:error480@10.1.100.10 via_cnt==1" ======================================================================== ======== SIP MESSAGE 5 10.1.100.10:5060() -> 10.1.0.149:3490() UDP Frame 57 11/Jul/05 10:07:31.5715 TimeFromPreviousSipFrame=0.0747 TimeFromStart=535.8610 SIP/2.0 480 Stateful 480 Reply Via: SIP/2.0/UDP 10.1.0.149:3490;branch=z9hG4bK411c33ff.10.1.0.149.3490 From: "alice" <sip:alice@10.1.0.149>;tag=369b1fba.10.1.0.149.3490 To: <sip:error480@10.1.100.10>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-d446 Call-ID: 66fcd326(a)10.1.0.149.3490 CSeq: 1 INVITE Server: Sip EXpress router (0.8.99-dev2 (i386/linux)) Content-Length: 0 Warning: 392 10.1.100.10:5060 "Noisy feedback tells: pid=14072 req_src_ip=10.1.0.149 req_src_port=3490 in_uri=sip:error480@10.1.100.10 out_uri=sip:error480@10.1.100.10 via_cnt==1" ======================================================================== ======== SIP MESSAGE 6 10.1.0.149:3490() -> 10.1.100.10:5060() UDP Frame 58 11/Jul/05 10:07:33.5186 TimeFromPreviousSipFrame=1.9471 TimeFromStart=537.8081 ACK sip:error480@10.1.100.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.149:3490;branch=z9hG4bK411c33ff.10.1.0.149.3490 From: "alice" <sip:alice@10.1.0.149>;tag=369b1fba.10.1.0.149.3490 To: <sip:error480@10.1.100.10>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-d446 Call-ID: 66fcd326(a)10.1.0.149.3490 CSeq: 1 ACK Content-Length: 0 ======================================================================== ======== SIP MESSAGE 7 10.1.0.149:3490() -> 10.1.100.10:5060() UDP Frame 167 11/Jul/05 10:26:51.5746 TimeFromPreviousSipFrame=1158.0560 TimeFromStart=1695.8641 INVITE sip:error480@10.1.100.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.149:3490;branch=z9hG4bKd8f1b2b1.10.1.0.149.3490 From: "alice" <sip:alice@10.1.0.149>;tag=88bb6f0e.10.1.0.149.3490 To: <sip:error480@10.1.100.10> Call-ID: 749a6f74(a)10.1.0.149.3490 CSeq: 2 INVITE Contact: "alice" <sip:alice@10.1.0.149:3490>;+g.poc.talkburst max-forwards: 70 accept-contact: *;+g.poc.talkburst;require;explicit Content-Type: application/sdp Content-Length: 131 v=0 o=- 1113855662 1113855662 IN IP4 10.1.0.149 s=- c=IN IP4 10.1.0.149 t=0 0 m=audio 1000 RTP/AVP 98 a=rtpmap:98 AMR/16000 ======================================================================== ======== SIP MESSAGE 8 10.1.100.10:5060() -> 10.1.0.149:3490() UDP Frame 170 11/Jul/05 10:26:51.5939 TimeFromPreviousSipFrame=0.0193 TimeFromStart=1695.8834 SIP/2.0 480 Stateful 480 Reply Via: SIP/2.0/UDP 10.1.0.149:3490;branch=z9hG4bKd8f1b2b1.10.1.0.149.3490 From: "alice" <sip:alice@10.1.0.149>;tag=88bb6f0e.10.1.0.149.3490 To: <sip:error480@10.1.100.10>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-f855 Call-ID: 749a6f74(a)10.1.0.149.3490 CSeq: 2 INVITE Server: Sip EXpress router (0.8.99-dev2 (i386/linux)) Content-Length: 0 Warning: 392 10.1.100.10:5060 "Noisy feedback tells: pid=14068 req_src_ip=10.1.0.149 req_src_port=3490 in_uri=sip:error480@10.1.100.10 out_uri=sip:error480@10.1.100.10 via_cnt==1" ======================================================================== ======== SIP MESSAGE 9 10.1.100.10:5060() -> 10.1.0.149:3490() UDP Frame 171 11/Jul/05 10:26:52.3072 TimeFromPreviousSipFrame=0.7133 TimeFromStart=1696.5967 SIP/2.0 480 Stateful 480 Reply Via: SIP/2.0/UDP 10.1.0.149:3490;branch=z9hG4bKd8f1b2b1.10.1.0.149.3490 From: "alice" <sip:alice@10.1.0.149>;tag=88bb6f0e.10.1.0.149.3490 To: <sip:error480@10.1.100.10>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-f855 Call-ID: 749a6f74(a)10.1.0.149.3490 CSeq: 2 INVITE Server: Sip EXpress router (0.8.99-dev2 (i386/linux)) Content-Length: 0 Warning: 392 10.1.100.10:5060 "Noisy feedback tells: pid=14068 req_src_ip=10.1.0.149 req_src_port=3490 in_uri=sip:error480@10.1.100.10 out_uri=sip:error480@10.1.100.10 via_cnt==1" ======================================================================== ======== SIP MESSAGE 10 10.1.0.149:3490() -> 10.1.100.10:5060() UDP Frame 172 11/Jul/05 10:26:53.5354 TimeFromPreviousSipFrame=1.2282 TimeFromStart=1697.8249 ACK sip:error480@10.1.100.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.149:3490;branch=z9hG4bKd8f1b2b1.10.1.0.149.3490 From: "alice" <sip:alice@10.1.0.149>;tag=88bb6f0e.10.1.0.149.3490 To: <sip:error480@10.1.100.10>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-f855 Call-ID: 749a6f74(a)10.1.0.149.3490 CSeq: 2 ACK Content-Length: 0 ======================================================================== ======== SIP MESSAGE 11 10.1.0.149:3490() -> 10.1.100.10:5060() UDP Frame 173 11/Jul/05 10:26:53.7753 TimeFromPreviousSipFrame=0.2400 TimeFromStart=1698.0648 ACK sip:error480@10.1.100.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.149:3490;branch=z9hG4bKd8f1b2b1.10.1.0.149.3490 From: "alice" <sip:alice@10.1.0.149>;tag=88bb6f0e.10.1.0.149.3490 To: <sip:error480@10.1.100.10>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-f855 Call-ID: 749a6f74(a)10.1.0.149.3490 CSeq: 2 ACK Content-Length: 0 ======================================================================== ======== List of reasons 398 traced packets out of 409 in scenario were not included. 11 included. [192] Non Sip TCP/UDP Packet Filtered Out [117] TCP No Data [33] IEEE802.3 Packet [30] IGMP Packets [20] Arp Packets [6] TCP Duplicate Data ======================================================================== ======== Thanks & Regards, Amit Singh Sony Ericsson Mobile Communications, 7001 Development Drive, RTP Durham, NC 27709 Ph: +1-919-472-1601

1 0

[Users] need help to configure voicemail
by harry gaillac 11 Jul '05

11 Jul '05

Hello, I use ser-0.9.3 . I wish to send calls to offline or unavailable users to SEMS . Private Network:---------SER----------internet users agents Mediaproxy SEMS I tried to add something like this in ser.cfg without success ! Thanks for help Harry modparam("tm", "fr_inv_timer", 25) ....................................................... # Check ACL for voicemail if (is_user_in("Request-URI", "voicemail")) { setflag(5); }; # User unavailable #if (method=="INVITE" && isflagset(5)) { # t_on_failure("1"); # route(1); # break; #}; lookup("aliases"); if (!lookup("location")) { route(6); } else if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; ....................................................... route[1] { ########################### # Default Message Handler # ########################### t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[6] { ########################## # voicemail Handler # ########################## # forward to voicemail now rewritehost("192.168.0.2"); } onreply_route[1] { ########################## # Statefull Handler # ########################## if (client_nat_test("1")) { setflag(4); }; if (isflagset(4) && status=~"(180) || (183) || 2[0-9][0-9]") { if(search("^Content-Type: .*$")) { use_media_proxy(); }; }; } failure_route[1] { ########################## # Voicemail Handler # ########################## revert_uri(); rewritehost("192.168.0.2"); } ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

4 6

[Serusers] Problem with presence (400 response to subscribe)
by Daniel Machín Vazquez-Illá 11 Jul '05

11 Jul '05

Hello. I'm having problems with pa module. I try to SUBSCRIBE to a user's presence and I get a 400 response to this subscribe. Ser logs show: 12(10693) handle_subscription() entered 12(10693) parse_headers: flags=1581064 12(10693) parse_hfs(): Error while parsing headers 12(10693) handle_subscription(): Error while parsing message header 12(10693) handle_subscription about to send_reply and return -2 I'm using Ser 9.3 and Windows Messenger 4.7 for testing. My ser.cfg file looks like this: [...] loadmodule "/usr/local/lib/ser/modules/pa.so" [...] modparam("usrloc", "use_domain", 1) modparam("registrar", "use_domain", 1) [...] route { [...] if (method=="SUBSCRIBE") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); }; handle_subscription("registrar"); break; }; [...] I can post the whole file or the SIP logs if you need them. Does anyone know why it doesn't work? Isn't Windows Messenger compatible with pa module? Lots of thanks.

1 0

[Users] sems voicemail plugin
by harry gaillac 11 Jul '05

11 Jul '05

Hello, When I try to record a message for a offline i get : Jul 11 10:38:09 serveur1 postfix/smtpd[4097]: warning: Illegal address syntax from serveur1.home.net[192.168.0.1] in MAIL command: voicemail(a)80.119.7.29 I setup email.template in sems.conf ! Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 4

[Users] sems voicemail module
by harry gaillac 11 Jul '05

11 Jul '05

hello, When users are busy or offline voicemail module send to smtp server bad command. Why voicemail(a)80.119.10.19 not voicemail(a)nxs.yi.org according to voicemail.template in sems.cfg ? warning: Illegal address syntax from 19.10.119-80.rev.gaoland.net[80.119.10.19] in MAIL command: voicemail(a)80.119.10.19 SER support t_on_failure not t_on_negative unlike OPENSER what'are the differences ? For people who read my posts I solve forward to voicemail when busy (t_check_status) and offline However I need help with tm timer an t_on_failure Harry Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] Strange REGISTER Problems
by Ozan Blotter 11 Jul '05

11 Jul '05

Dear List, I can register my ATA'S (Some are NATted and some ara Port Forwarded) to SER (on Public IP) and i use MySQL auth/acc together with Mediaproxy. All users can talk to others, and call them , receive calls from others no problem. But most of the time, they seem unregistered and this happens suddenly, i'm sure that Internet Connection is fine with SER Server and also clients are connected through Cable Modem or ADSL Connection. I unplug or unpower some ATA's, wait a bit like 30 minutes, the devices becomes cold or what's happening else, they can register and works perfect... But when i connect them for hours, they unregister themselves, i do not know what happens... I've tried both Zyxel P2002's with several firmwares, Zyxel P2000Ws, Cisco ATA186's and some LinkSYS Pap2-NA's but all same... Is there any chance to be something wrong in SER settings or Operating System ? Thanks, Ozan

2 1

[Serusers] REGISTER problem with the last stable SER version.
by Ricardo Martinez 11 Jul '05

11 Jul '05

Hello list. I detected a strange behavior with my SER. I downloaded the last stable version from the CVS. And it seems to work ok. But for a particular endpoint the register proccess seems to be corrupted. This is what i have. (please check the PORT for the register) U 200.113.154.25:12196 -> 10.0.0.35:5060 REGISTER sip:sipproxy.mydomain.com SIP/2.0. U 10.0.0.35:5060 -> 200.113.154.25:12196 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 200.113.154.25:12196;branch=z9hG4bK-e4d3ecd4;rport=12196. U 200.113.154.25:12196 -> 10.0.0.35:5060 REGISTER sip:sipproxy.mydomain.com SIP/2.0. U 10.0.0.35:5060 -> 200.113.154.25:12196 SIP/2.0 200 OK. domain: 'location' aor : '5555847110' ~~~Contact(0x42326db8)~~~ domain : 'location' aor : '5555847110' Contact : 'sip:5555847110@200.113.154.25:121962196' Expires : 528 q : Call-ID : '1016393a-5c1cbaeb(a)200.113.154.25' CSeq : 45 User-Agent: 'Linksys/PAP2-3.1.3(LS)' received : '' State : CS_NEW Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ Then i try to make an invite to the endpoint from the 10.0.0119: U 10.0.0.119:5060 -> 10.0.0.35:5060 INVITE sip:5555847110@sipproxy.mydomain.com SIP/2.0. U 10.0.0.35:5060 -> 10.0.0.119:5060 SIP/2.0 100 trying -- your call is important to us. U 10.0.0.35:5060 -> 200.113.154.25:65236 INVITE sip:5555847110@200.113.154.25:121962196 SIP/2.0. U 10.0.0.35:5060 -> 200.113.154.25:65236 INVITE sip:5555847110@200.113.154.25:121962196 SIP/2.0. U 10.0.0.35:5060 -> 200.113.154.25:65236 INVITE sip:5555847110@200.113.154.25:121962196 SIP/2.0. U 10.0.0.35:5060 -> 200.113.154.25:65236 INVITE sip:5555847110@200.113.154.25:121962196 SIP/2.0. U 10.0.0.35:5060 -> 200.113.154.25:65236 INVITE sip:5555847110@200.113.154.25:121962196 SIP/2.0. U 10.0.0.35:5060 -> 200.113.154.25:65236 INVITE sip:5555847110@200.113.154.25:121962196 SIP/2.0. U 10.0.0.35:5060 -> 200.113.154.25:65236 INVITE sip:5555847110@200.113.154.25:121962196 SIP/2.0. Is this a bug? Or maybe a problem with my configuration. I hope that someone could help me. Thanks. Ricardo Martinez.-

2 1

[serusers] Message Body
by michael p 11 Jul '05

11 Jul '05

hi 2 questions : 1. how can i got the "From" line, the "To" line and the "Body" of a SIP message "MESSAGE". i want to got them in ser.cfg how can i do that please? 2. i tried to go to the iptel forum (http://www.iptel.org/phpBB/index.php) but i didn't find how can i post a message... thanks for the help ps: the most important question is the first M.

1 0

RE: [Serusers] radius authentication with 0.9.3 stable
by Ricardo Martinez 11 Jul '05

11 Jul '05

Hello Steve, You need to configure the parameters of your radius client too: the key, the auth server and port, etc. You need to modify the /usr/local/etc/radiusclient-ng/radiusclient.conf in order to to this. And of course, you need a Radius SERVER. Good luck Ricardo Martinez.- > -----Mensaje original----- > De: Steve Blair [mailto:blairs@isc.upenn.edu] > Enviado el: Sábado, 09 de Julio de 2005 8:42 > Para: 'serusers(a)lists.iptel.org' > Asunto: [Serusers] radius authentication with 0.9.3 stable > > > > Does anyone have a pointer to a how-to guide for setting up > radius authentication with SER 0.9.3 stable? I am running on > RH ES 4 with the latest radiusclient-ng installed and I've > compiled in the radius related modules. I've added site specific > modparam statements such as those in modules/auth_radius/readme > but I'm stuck on what is left. > > Thanks, Steve > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

RE: [Serusers] mediaproxy.sock
by Ricardo Martinez 11 Jul '05

11 Jul '05

Yes.. the README file is a good start, but i have some tips ('cause at the beggining i had some problems with mediaproxy too). The mediaproxy package has a "boot" directory, inside you can found a "mediaproxy.other" file. That file can help you to start mediaproxy. Another thing, i'm guess you're using phyton?. I had some problem and memory leak with python2.3, so i upgraded to python2.4 and now is working without any problem. Good luck. Ricardo Martinez.- > -----Mensaje original----- > De: Greger V. Teigre [mailto:greger@teigre.com] > Enviado el: Domingo, 10 de Julio de 2005 4:43 > Para: Abdul Lateef; serusers(a)lists.iptel.org > Asunto: Re: [Serusers] mediaproxy.sock > > > Abdul, > Mediaproxy has a very well written README file. > g-) > > Abdul Lateef wrote: > > Hello > > > > I did the following commond to check mediaproxy is > > running or not. But i think it is not running. How i > > can make it run, i means uing whcich commond. > > > > [root@localhost root]# netstat -nlp | grep proxy > > [root@localhost root]# netstat -nlp|grep proxy > > [root@localhost root]# > > > > > > Thank You > > > > Yours, > > Abdul Lateef > > Computer Programmer > > HATIF COM > > Mob: +974 - 5405022 > > Tel: +974 - 4883068 > > ICQ: 276994704 > > YM!: abdul_zu > > Fax: +974 - 4883063 > > Doha Qatar > > http://www.hatif.com > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com > > > > _______________________________________________ > > 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] postgres and mysql
by Sebastian Kühner 11 Jul '05

11 Jul '05

Hello, I'm using postgres for authentication and accounting since a few days and it's working perfect. But I'm wondering why I have to load the mysql.so - Module, too. If not (commented), I get the following error: 0(0) Maxfwd module- initializing 0(0) PG[238] db_init entry 0(0) PG[193] str2valp got int 2 0(0) PG[274] db_close entry textops - initializing 0(0) bind_dbmod: Module mysql does not export db_use_table function 0(0) ERROR: uridb_db_bind: unable to bind to the database module 0(0) ERROR: uri_db:mod_init(): No database module found 0(0) init_mod(): Error while initializing module uri_db ERROR: error while initializing modules This error also appears, if I don't have installed/running a mysql server. It seems very strange... because all the data is saved correctly in my postgresql-database. Why do I need the mysql-module then? OK, one additional information. To specify my database I use the following command: modparam("acc", "db_url", "postgres://admin:password@localhost:5432/AAA") If I use "sql://" instead of "postgres://" there appears the same error like above. Any ideas? Thanks! Sebastian

2 2

[Serusers] SER in production VOIP environment
by Jerlique Ban 11 Jul '05

11 Jul '05

Hello, I thought I would ask the list how people are using SER. We plan to use it in a production environment to be able to provide VOIP services to some clients. How do companies such as vonage etc setup their systems, is it with SER handling REGISTRATIONS and all other UA requests, and asterisk as a backend which merely does media proxying? Look forward to your answers... JB

7 9

[Serusers] PSTN gateway user lookup
by Luca Corti 11 Jul '05

11 Jul '05

Hello, I'm using ser with an AS5350 as PSTN gateway. On the AS i've defined: sip-ua sip-server ipv4:1.2.3.4 and for every PSTN number I want to map to a SIP contact for inbound calls. dial-peer voice 292 voip description test destination-pattern 0123456789 progress_ind setup enable 3 session protocol sipv2 session target ipv4:1.2.3.4 req-qos controlled-load ip qos dscp cs5 media ! Now calls for 0123456789 are forwarded to ser (1.2.3.4), but I need a way to make ser know who is going to receive the call. I solved this adding a record to the alias table containing the PSTN address for incoming calls of each user, which is then matched in ser through lookup(aliases). Does anyone use a different approach to solve this? thanks a lot -- Luca Corti PGP Key ID 1F38C091 BOFH excuse of the moment: You've been infected by the Telescoping Hubble virus.

2 1

[Users] forward to voicemail if busy or unavailabe
by harry gaillac 11 Jul '05

11 Jul '05

Hello, I posted mails last week. Anybody could help me to configure forward to voicemail when busy or anavailable . I set timer (tm) failure route without success When user is offline it' ok. Thanks for help 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] New Installation
by Abdul Lateef 11 Jul '05

11 Jul '05

Hello, Unfortunatly my lunx box storage device carshed and i loos my ser configurations. Now i am going to install it with letast ver. of ser. Can anyone tell me, is there any others thirdparty module to include during intallation to full support for VoIP phone. I want to install for the following purpose. 1 Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Vonage/Broadvoice Limit ?
by Budi Gautama 11 Jul '05

11 Jul '05

Hi, Is there any limit in the no of minutes of Vonage/Broadvoice ? if there is no limit, can we use it for terminating traffic ? Pls share your experience. Budi

1 0

[Serusers] Devices with DNS SRV Support
by Daryl Sanders 10 Jul '05

10 Jul '05

Is there an up-to-date list somewhere of devices (ATAs / Hard-Phones / Soft phones) that have been verified to work with DNS SRV load-balancing & failover? If you use a certain model that works, please let me know. I just need to know what my options are if I should I go this route. Thanks! - Daryl

1 0

Re: [Serusers] RE: [Users] need help to configure voicemail
by harry gaillac 10 Jul '05

10 Jul '05

in others words I can't put SER+SEMS on private network > or there is > already an existing "connection" from inside/private > network to the outside. What's the way ? Harry --- harry gaillac <gaillacharry(a)yahoo.fr> a écrit : > in others words I can't put SER+SEMS on private > network > > > or there is > > already an existing "connection" from > inside/private > > network to the outside. > > What's the way ? > > Harry > --- Nils Ohlmeier <lists(a)ohlmeier.org> a écrit : > > > Nobody is able to send anything from the public > side > > into the private network, > > except there is an port forwarding on the NAT, the > > NAT is broken or there is > > already an existing "connection" from > inside/private > > network to the outside. > > > > Nils > > > > On Sunday 10 July 2005 19:12, harry gaillac wrote: > > > Does SER is able to send messages to an other > SER > > > behind nat ? > > > > > > if yes what's the way ? > > > > > > Harry > > > > > > > > > > ___________________________________________________________________________ > > 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

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[Serusers] Tr: RE: [Users] need help to configure voicemail
by harry gaillac 10 Jul '05

10 Jul '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

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[Serusers] Userloc replication via FIFO
by Daryl Sanders 10 Jul '05

10 Jul '05

I'm looking for a way to replicate userloc changes to another SER box via FIFO. I am using FIFO via PHP to manipulate contacts, but I'm not sure how to replicate those changes to my other SER's memory. Any thoughts? Any help would be greatly appreciated. I'm kinda stumped on this one. - Daryl

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[Serusers] Pass variables to log command
by Chris Mason (Lists) 10 Jul '05

10 Jul '05

In learning to configure ser, it would seem very useful to log info such as the URI and other variables, but I can't see how to do it. Is there a way to include data in the log statements? Something like: # Is the call for the PSTN gateway? if (uri=~"^sip:1") { log(1,"Call for PSTN Gateway. Number called: $uri"); forward(66.250.54.132, 5065); break; } -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla(a)yahoo.com

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RE: [Serusers] ser mysql problem
by Abdul Lateef 10 Jul '05

10 Jul '05

Hello, I opned the serctl and configured to accept the DB server. But still i i have trouble to start and stop ser. here is new log: Jul 2 22:40:09 localhost ser: WARNING: fix_socket_list: could not rev. resolve 10.0.0.20 Jul 2 22:40:09 localhost ser: WARNING: fix_socket_list: could not rev. resolve 10.0.0.20 Jul 2 22:40:09 localhost ser[3194]: INFO: udp_init: SO_RCVBUF is initially 65535 Jul 2 22:40:09 localhost ser[3194]: INFO: udp_init: SO_RCVBUF is finally 262142 Jul 2 22:40:09 localhost ser[3195]: INFO: fifo process starting: 3195 Jul 2 22:40:09 localhost ser[3196]: NOTICE:init_avp_child: no avp_db_url specified -> feature disabled Jul 2 22:40:09 localhost ser[3197]: NOTICE:init_avp_child: no avp_db_url specified -> feature disabled Jul 2 22:40:09 localhost ser[3198]: NOTICE:init_avp_child: no avp_db_url specified -> feature disabled Jul 2 22:40:09 localhost ser[3199]: NOTICE:init_avp_child: no avp_db_url specified -> feature disabled Jul 2 22:40:09 localhost ser[3195]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... Jul 2 22:40:09 localhost ser[3195]: WARNING: no fifo_db_url given - fifo DB commands disabled! --- harry gaillac <gaillacharry(a)yahoo.fr> wrote: > try openserctl > --- Abdul Lateef <abdul_zu(a)yahoo.com> a écrit : > > > hi, > > > > I configured ser.cfg to use mysql authuntication > > withe > > guide of http://www.onsip.org > > > > but i have some trouble i cannot > restart/start/stop > > the ser server using serctl I would like to paste > > the > > exactly log of ser. > > > > ------------------- > > Jul 2 21:52:27 localhost ser: WARNING: > > fix_socket_list: could not rev. resolve 10.0.0.20 > > Jul 2 21:52:27 localhost ser: WARNING: > > fix_socket_list: could not rev. resolve 10.0.0.20 > > Jul 2 21:52:28 localhost ser[2385]: INFO: > udp_init: > > SO_RCVBUF is initially 65535 > > Jul 2 21:52:28 localhost ser[2385]: INFO: > udp_init: > > SO_RCVBUF is finally 262142 > > Jul 2 21:52:28 localhost ser[2385]: ERROR: > > tcp_init: > > bind(6, 0x8114d24, 16) on 10.0.0.20: Address > already > > in use > > Jul 2 21:52:28 localhost ser[2385]: INFO: signal > 15 > > received > > --------------------------------- > > > > > > here is the main configuration of file. > > > > debug=3 > > fork=yes > > log_stderror=no > > > > listen=10.0.0.20 # put your server IP > > address > > here > > port=5060 > > children=4 > > > > dns=no > > rev_dns=no > > > > fifo="/tmp/ser_fifo" > > fifo_db_url="mysql://root:123@10.0.0.26/ser" > > > > 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_db.so" > > > > modparam("auth_db|uri_db|usrloc", "db_url", > > "mysql://root:123@10.0.0.26/ser") > > modparam("auth_db", "calculate_ha1", 1) > > modparam("auth_db", "password_column", "password") > > modparam("usrloc", "db_mode", 2) > > modparam("rr", "enable_full_lr", 1) > > > > > > Yours, > > Abdul Lateef > > Computer Programmer > > HATIF COM > > Mob: +974 - 5405022 > > Tel: +974 - 4883068 > > ICQ: 276994704 > > YM!: abdul_zu > > Fax: +974 - 4883063 > > Doha Qatar > > http://www.hatif.com > > > > > > > > > ____________________________________________________ > > > > Yahoo! Sports > > Rekindle the Rivalries. Sign up for Fantasy > Football > > > > http://football.fantasysports.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 > Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail

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Re: [Serusers] mediaproxy.sock
by Abdul Lateef 10 Jul '05

10 Jul '05

Hello I did the following commond to check mediaproxy is running or not. But i think it is not running. How i can make it run, i means uing whcich commond. [root@localhost root]# netstat -nlp | grep proxy [root@localhost root]# netstat -nlp|grep proxy [root@localhost root]# Thank You Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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RE: [Serusers] Userloc replication via FIFO
by Rodrigue ESTEVE 10 Jul '05

10 Jul '05

Hi Daryl, Perhaps in your ser.cfg you can for a REGISTER message execute an external script (perl/bash) to pass argument to FIFO. For the variable to pass to your script take a look in the ser directory: ser-0.9.3/modules/exec/README for the use of fifo for registration see below: hope this could help bye, Rodrigue Suggested by Samuel Osorio Calvo: Try the following..... ################################################# #!/bin/sh cat> /tmp/ser_fifo << :ul_add:return_fifo location sip:user@domain sip:user@IP {expires} {qvalue} {replicate} {flags} ################################################## return_fifo is the name of the fifo (with a /tmp prefix) that will use SER for answering to the fifo command.If you don't want to read it, just don't worry about the possible complaint of FIFO not existing, error opening return fifo or something like that in SER log. location is the table where to add/update the record sip:user@domain-->AoR you might have to look at SER's debug output because I remember there was something with sip: prefix in either AoR or contact. I think using sip: will work but just check if you have to remove the sip: prefix from the contact.... sip:user@IP-->contact expires-->record validity in seconds qvalue-->[0..1] replicate-->bool 0|1 flags-->not 100% sure if it corresponds to the flag vector of SER or just indicate natted contacts...I recommend to use a 0 value here.. Remember to add an End Of File character in the last line or there are some funny parsing things with subsequent messages isnerted in the fifo... Hope it helps, Samuel. Unclassified. >>> "Rodrigue ESTEVE" <rodrigue(a)mobius.fr> 06/28/05 09:13AM >>> Does anybody have an explanation to use the fifo for registration (using a script and not the serctl command) ? Thanks very much _____ De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Rodrigue ESTEVE Envoyé : mardi 14 juin 2005 19:00 À : serusers(a)lists.iptel.org Objet : [Serusers] Use of fifo for registration :ul_add: command tutorial Hi all, I would like to use ser_fifo to register a user; I think the :ul_add: command is the clue, but I don't how to use the fifo to do this. My aim is to do the following: I have a Voip gateway on a customer with a lot of phone number (700) and I would like to authenticate each number. Unfortunately I have to add a small amount of configuration on the voip gateway each time I add a number (uri), moreover I'm not sure this gateway will work with so many numbers, and the bandwidth needed to register each number every 5 minutes will be wasted. The idea is to challenge only one time the gateway and then in ser.cfg if the challenge succeeds I execute a script to register by bulk all the numbers enclosed in a file on the proxy. To do this I have to pass arguments to the fifo using the :ul_add: command, but this is where I'm stuck. I've done this: echo :ul_add:receiving_fifo > /tmp/ser_fifo echo location > /tmp/ser_fifo echo 100 > /tmp/ser_fifo echo <sip:100@10.1.1.1:5060> > /tmp/ser_fifo echo 300 > /tmp/ser_fifo echo 0 > /tmp/ser_fifo echo 2 > /tmp/ser_fifo but it does not work, I think this is due to the receiving_fifo, I have to do something with it to process the response of SER, but what ?? I know I can use the following command to workaround the pbm, and I'm successful with this command: Serctl fifo ul_add location *.. But it requires one process each time I register a number, and it takes 1 minute on a Celeron 700 for 500 numbers (30 % CPU), and I think using the fifo is a more suitable solution (less CPU consuming). Thanks for your support, Rodrigue PS: I would like to register every 5minutes,cause the customer gateway is able to redirect to PSTN in case of failure. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -----Message d'origine----- De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de Daryl Sanders Envoyé : samedi 9 juillet 2005 03:22 À : serusers(a)lists.iptel.org Objet : [Serusers] Userloc replication via FIFO I'm looking for a way to replicate userloc changes to another SER box via FIFO. I am using FIFO via PHP to manipulate contacts, but I'm not sure how to replicate those changes to my other SER's memory. Any thoughts? Any help would be greatly appreciated. I'm kinda stumped on this one. - Daryl _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Host is down.
by Zaihan 10 Jul '05

10 Jul '05

Hi all, I've got this error below, tried everything i could but couldn't get it to work, i need some help on this. Thanks, Zaihan Jul 9 22:30:04 flamejob ser[53145]: ERROR: udp_send: sendto(sock,0x28421dc8,558 ,0,0x28406c50,16): Host is down(64) Jul 9 22:30:04 flamejob ser[53145]: msg_send: ERROR: udp_send failed Jul 9 22:31:23 flamejob ser[53145]: ERROR: udp_send: sendto(sock,0x28421dc8,559 ,0,0x28406c50,16): Host is down(64) Jul 9 22:31:23 flamejob ser[53145]: msg_send: ERROR: udp_send failed Jul 9 22:32:46 flamejob ser[53145]: ERROR: udp_send: sendto(sock,0x28421dc8,558 ,0,0x28406c50,16): Host is down(64) Jul 9 22:32:46 flamejob ser[53145]: msg_send: ERROR: udp_send failed Jul 9 22:34:08 flamejob ser[53145]: ERROR: udp_send: sendto(sock,0x28421dc8,558 ,0,0x28406c50,16): Host is down(64) Jul 9 22:34:08 flamejob ser[53145]: msg_send: ERROR: udp_send failed

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[Users] Just installed OpenSER
by cgtel2002＠yahoo.fr 10 Jul '05

10 Jul '05

Hello dear list, Just installed OpenSER # openser -V version: openser 0.9.5 (i386/freebsd) I have 2 x questions: # openserctl stop /start /moni not working E.g. : Stopping OpenSER : No PID file found! OpenSER probably not running Yet OpenSER is running and working for test with default cfg Second Q., how am I getting the mysql.so ? Doesn't seem to be in /lib/openser/modules. Thanks in advance, George ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] version: ser 0.9.3 (i386/freebsd) PID file /var/run/ser.pid does not exist
by GeoTel 09 Jul '05

09 Jul '05

Hi list, Just installed ser -V version: ser 0.9.3 (i386/freebsd) Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed Anyone have an idea ? Thanks in advance, Gg

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Re: [Serusers] Session-Timers for Prepaid
by Bruno Lopes F. Cabral 09 Jul '05

09 Jul '05

Sorry but no, I'm not recommending that (or anything else). I was only trying to feedback developers for this popular request (prepaid with SER stateless or even in statefull mode). Privately I'll provide you a few more thoughts on this matter, if you wish. !3runo Nils Ohlmeier wrote: > Hi Bruno, > > sorry but what is the conclusion of your mail? That everbody > who wants to do pre-paid VoIP should use Asterisk instead of SER? > > Nils

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Re: [Users] Just installed OpenSER
by Daniel-Constantin Mierla 09 Jul '05

09 Jul '05

Hello, On 07/10/05 07:47, cgtel2002(a)yahoo.fr wrote: >Hello dear list, > >Just installed OpenSER > # openser -V >version: openser 0.9.5 (i386/freebsd) > >I have 2 x questions: > ># openserctl stop /start /moni not working >E.g. : Stopping OpenSER : No PID file found! OpenSER probably not running >Yet OpenSER is running and working for test with default cfg > > it is better to use a start/stop script, the openserctl start/stop was not tested very well. >Second Q., how am I getting the mysql.so ? >Doesn't seem to be in /lib/openser/modules. > > See http://www.openser.org/dokuwiki/doku.php?id=f.a.q, there is a question about mysql.so module. Daniel >Thanks in advance, >George > > > > > > > > > >___________________________________________________________________________ >Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger >Téléchargez cette version sur http://fr.messenger.yahoo.com > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

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[Users] Correct passwords for users: openser, openserro and root dor the openser database under MySQL
by Kofi Obiri-Yeboah 09 Jul '05

09 Jul '05

In section D openser with Persistent Data Storage of the INSTALL docs at openser.doc"(you will be prompted for password of MySQL "root" user)" could someone please confirm what the correct root user password is? Is the root user the same as admin Also what are the passwords for the openser and openserro users respectively? Success is a state of mind If you want success Start thinking of yourself as a success

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[Users] location table version should be upgraded
by Juha Heinanen 09 Jul '05

09 Jul '05

i tried to start openser HEAD and got mysql error from usrloc. it turned out that the problem was new socket column that didn't exist in my ser database. i think it would be a good idea to increase location table version so that it is easy to check if a particular version of openser would work with a particular table version. so location table version should now be 7. -- juha

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[Serusers] Can't install ser 0.9.3 from onsip.org
by Yiannis Martis 09 Jul '05

09 Jul '05

2 1

[Users] voicemail serweb+sems
by harry gaillac 09 Jul '05

09 Jul '05

Hello Read my post at http://mail.iptel.org/pipermail/serweb-users/2005-July/000015.html Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] NAT - Lots of flavours...
by Ricardo Poppi 09 Jul '05

09 Jul '05

Hi list, I´m trying to put to work a NATed environment and want to share some information and request some I don´t realized yet. I use an asterisk gateway, with a public IP, working really fine for UAs with public IPs. At the same machine I runs SER that receives all SIP messages and handle when it should go to a SIP UA or to asterisk, rewriting the port (to the one asterisk uses) and sending to it. I don´t replicate register to asterisk, and use the user accounts as "peer", instead of "friends". My ser.cfg is using the "force_rport()" and "fix_nated_contact()" for every REGISTER it receives from nat UAs - I know when it comes from a NATed UA using nat_uac_test("2"). Every INVITE that comes from NATed UA passes through a "fix_nated_sdp("2"), that rewrites the IP address of SDP headers. Using a onreply route I fix the 200 OK INVITE message, just in case that the NATed UA is on the called side. The UAs I´m using are X-Lite, Clipcomm CP-100 IP Phone, and Grandstream HT-488. Below I wrote the different kinds of configuration into the UA and in ser.cfg, and the results I got: 1) Using without touching the UA - It don´t know it is a NATed UA. ----------------------------------------------------------------------------------------------------------------------------- All REGISTER are treated ok because the force_rport make SER respond to the register on the same external IP:Port it received. On the same hand, it stores the right URI into the location database making the UA receive the subsequent INVITES or other SIP messages through the external IP:Port. The INVITES that comes from NATed UA have their SDP IP address rewriten by SER and the external IP takes place. But the port is kept the internal value, so when the called UA tries to reach the External_IP:Internal_port the NAT/Firewall probably block/drops the packets, and the result is a one-way audio - The one-way audio is probably due to the right value that comes from the SDP headers of the called UA - asterisk -, that has a public IP. 2) a=direction:active ---------------------------------- If I add into ser.cfg a "fix_nated_sdp("1")" command, it will add the "a=direction:active" parameter to SDP header of INVITE that comes from NATed UAs. I saw that it´s happening but the asterisk seems to not understand that and don´t expect for the first RTP packet to get the IP:Port information of the media. A one-way audio is the result of that. The asterisk is probably sending RTP packets to the Ext_IP:Internal_port, and the firewall is blocking the packets. 3) Using STUN ------------------------ When I set the STUN server into the UA configuration - I used stunserver.org - the ua correct replaces the contact and sdp headers IP address, but do nothing about the RTP port, keeping the problem that the internal port of the UA, for media path, that the asterisk - or any other public UA - tries to reach is blocked by the firewall. DO ANYBODY WANTS TO SHARE SOME INFORMATION ABOUT THIS PROBLEM? Thanks in advance, Ricardo Poppi

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[Serusers] serweb login
by Chris Mason 09 Jul '05

09 Jul '05

I installed serweb on my Centos 4.1 server and edited all the paths, configs. When I try to login with admin and heslo, I get "Bad Password" I checked that register_globals=on and short_open_tag=on and restarted apache. I can't find the solution, I'd appreciate any help. Chris

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[Serusers] Simulate BYE Message
by Sebastian Kühner 08 Jul '05

08 Jul '05

Hello, I'm trying to send out a SIP BYE Message with sipsak to make my phones hanging up (to force them to hang up)... I used a file that is called bye.ser and looks like this: BYE sip:55@192.168.1.101:5060 SIP/2.0 Route: <sip:192.168.1.222:5061;lr> From: sip:14@192.168.1.101:5062;user=phone To: sip:55@192.168.1.101:5060;user=phone Contact: sip:14@192.168.1.101:5062;user=phone CSeq: 42759 BYE Call-ID: 746707906-47a484e6(a)192.168.1.101 User-Agent: sipsak 0.8.13 Max-Forwards: 16 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 ...using the command: sipsak -f bye.ser -s sip:192.168.1.222:5061 -vvv ... but the message doesn't go out. My ser is running on 192.168.1.222:5061 Well, my question is: How can I capture the SIP messages that are going through my ser?? Is there any debug option to do that? It would be easier for me to simulate it after... Many thanks!! Sebastian

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[Serusers] Re: serweb install problems
by Karel Kozlik 08 Jul '05

08 Jul '05

Hello, please (CC) the mailing list in your replies comments are inline > I'm trying to make the new SERWEB 0.9.3 working on a FreeBSD 5.4 and SER 0.9.3 -freeBSD > > First : php configs, ending with the same issue nothing, just blank white page. > Remark that for example Phpmyadmin and mysql 4 /php-5 works fine Could you send me your php.ini please. I know that this problem was reported but I can't reproduce it. > Second : tables does not match > Which install or release or version, ... implement table_cdr, table_uuidaliase, net_geo.... ??? Which tables don't match? If you useing both ser and serweb v0.9.3, tables must much. Tables you mentioned (table_cdr, table_uuidaliase, net_geo) are not used in default configuration. > Third: compatibility OS, pear classkit errors > pear install -f classkit > configure: error: libtool configure failed sorry I don't know freebsd and I haven't any problem with this. Try ask on some PHP forum. > Finally, which OS I have to use to make sure minimum working ? > Debian ? Which one are you using or recommend ? Debian. And use prepared packages provided by iptel. Karel > Thanks in advance for "any" help or info > geo > > > >>Hello, >>which version of serweb are you useing? >> >>Karel >> >> >>Sebastian Kühner napsal(a): >> >>>Hello, >>> >>>I have a big problem with serweb. I'm searching for the solution since >>>hours... but I didn't find it!! >>> >>>I installed serweb according to in the installation guide and the HOWTOs >>>that I found in the internet. The problem is that if I want to open the >>>index.php-page (of admin), always appears the Explorer Window >>> >>>"The page cannot be displayed" >>> >>>and when I look at the access_log of apache2 there appears something like >>>that: >>> >>>GET >>>/serweb/index.php?phplib_Session=d69f77e7ccb0d3b6c7d684bcbc368063&phplib_Ses >>>sion=4c8a0c40e09c1bfa01a91ef5832a026a&phplib_Session=5f354e9a405ade8796a54a3 >>>88b0353db&phplib_Session=769c3fa2b669f1dcdd2291c24227ef5d&phplib_Session=4ca >>>670bdd9a1f4f20441eac5f96c7b8e&phplib_Session=da194f7b1855c60daf515de13b14dab >>>2&phplib_Session=9bba333123ff119f94f924aeb0bc4710&phplib_Session=f6c9842d7cf >>>429c632608ad4ee7645c7&phplib_Session=bf06f602d09a1d313297a00f4b293133&phplib >>>_Session=6d798a6dcc84dac416415312debe47d3&phplib_Session=4d1991912b23d4a59ee >>>1f29f40830317&phplib_Session=31268d0242558745860dbacf41a1be15&phplib_Session >>>=5255e631f1cde5c708cb985db5ebfcaa&phplib_Session=7020b63c06bd983fb5eb054ac4e >>>f627f&phplib_Session=9651b78d2f391f94f23c8b46c603d0a5&phplib_Session=466c61f >>>55a87cb7ac4661e2816a3ec9d&phplib_Session=9515dbc324cc048f72fb7e1149166f18&ph >>>plib_Session=e0940531f81730793c23a703d772e738&phplib_Session=8e67ddbe3e69eec >>>9a952b2a59eed48ea&phplib_Session=621fc501c7590d413a79c58278956677&phplib_Ses >>>sion=a4f2af4b0d2e757e0568ee76f8563262&phplib_Session=70960d0cd7a32896b7fd80a >>>0a4c7f806&phplib_Session=5b78a5a53455042a992def2ab3012430&phplib_Session=0c8 >>>fb10161429bca54375b4dda5aec51&phplib_Session=b1ddb14986db2740a239b5f3494ae08 >>>3&phplib_Session=e281788f6b7533869817cce01a1ebaeb&phplib_Session=d16de271da5 >>>e37b5de9f81268975330b&phplib_Session=45a453bdc8d3969f392c2a4e230c7e0a&phplib >>>_Session=8cd36450b9f6d0592bfe9864bb4e3782&phplib_Session=db9b2e8ca0094fafc2b >>>923cd377c558c&phplib_Session=dcd92ee19095fba1c4a1475183b27352&phplib_Session >>>=38e1ab2e6f6a65a1bce8d7602db10014&phplib_Session=b074fa09bae847e186be4c22ed8 >>>280a5&phplib_Session=27e2d0c9e4172bed9752fe20a744195c&phplib_Session=d817e3e >>>3639c105042228e083f9081dd&phplib_Session=f9ac5312dcfcdf3bade2f80a7d6d98bf&ph >>>plib_Session=5f15600e879327263d4d83f489ec6422&phplib_Session=dde588458177dde >>>8907b932a4ee8a0fe&phplib_Session=d55285a056e6e5a873735a673d80a7cc&phplib_Ses >>>sion=1c51e86702ef7004638d9488cd967304&phplib_Session=bf5b51775d6208711b6904f >>>eeddf5d7b&phplib_Session=a7940c6ed31b41c83b60174ae05b023d HTTP/1.1" 302 - >>> >>> >>>... that means that serweb wants to make "thousands" of sessions. I didn't >>>find a solution for that in the internet and I hope that someone of you can >>>help me! >>> >>>Thanks! >>> >>>Sebastian >>> >>> >>>_______________________________________________ >>>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 > > > > > > > > > ___________________________________________________________________________ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] serweb: "DB Error: no such table"
by Timur V. Elzhov 08 Jul '05

08 Jul '05

Dear SER experts, I've downloaded and configured SER v0.9.3, as well as serweb, the head CVS version. I succeeds by serweb with subscribtion, logging on, and changing my own details. But the red message appears on my user page: "DB Error: no such table". However, again, there is now problem with actions mentioned just above. I'm not familiar with PHP, but I found that many php files under "serweb/data_layer/" directory contain the same "if" statement comprising the "DB_ERROR_NOSUCHTABLE" constant: if (DB::isError($res)) { if ($res->getCode()==DB_ERROR_NOSUCHTABLE) return true; //expected, table mayn't exist in installed version else {log_errors($res, $errors); return false;} } - it may be useful ... How should I locate & resolve this problem? Much thanks. -- Best regards, Timur Elzhov Warelex LLC

2 3

[Serusers] Module ACC_RADIUS problem
by Maximo Marin 08 Jul '05

08 Jul '05

Hello everyone, i am running, ser 0.9.3 on debian stable and get this error: Jul 8 10:21:41 sip_server ser: set_mod_param_regex: No module matching acc_radius found | Jul 8 10:21:41 sip_server ser: parse error (55,19-20): Can't set module parameter I have this in line 55 of my ser.cfg modparam("acc_radius", "radius_flag", 1) any help will be apreciated maxi _________________________________________________________________ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/

2 1

[Serusers] SER Function
by Razvan Nemesiu 08 Jul '05

08 Jul '05

Hi all, I want to know please if there is a SER function available to compare the username from teh From field in the sip message with the username resgistered in the database (in the subscriber table).I want to put such a function in order to restrict calls from registered users with modified caller-id. Thank you. -- Using Opera's revolutionary e-mail client: http://www.opera.com/m2/

2 1

[Serusers] Ser rpm for Centos?
by Chris Mason 08 Jul '05

08 Jul '05

Is there anywhere I can get a current rpm for Centos4, or good instrucitons for installing from the tar ball? I did the tarball install but I'm having problems with mysql and authentication. Chris

2 1

[Serusers] modparams for acc_radius ??? Please help ser 0.9.3 on debian
by Jose Bertuzzi 08 Jul '05

08 Jul '05

Hello, i have: Ser 0.9.3 Debian stable radiusclient-ng 0.5.1 and radiusclient-ng2 for debian freeradius 1.0.4 all on same machine. when starting ser and load the module acc_radius in ser.cfg i got the error: Jul 4 12:26:21 debian /usr/sbin/ser[8891]: Maxfwd module- initializing Jul 4 12:26:22 debian ser[8891]: rc_read_config: can't open /usr/local/etc/radiusclient/radiusclient.conf: No such file or directory Jul 4 12:26:22 debian ser[8891]: ERROR: acc: error opening radius config file: /usr/local/etc/radiusclient/radiusclient.conf So i renamed radiusclient-ng to radiusclient. Now ser starts up. But when i try to pass some parameters to acc_radius like: modparam("acc_radius", "radius_flag", 1) ser doesn't start and get the error: Jul 6 14:23:27 debian ser[1377]: WARNING: destroy_fifo: cannot delete fifo (/tmp/ser_fifo): Operation not permitted Jul 6 14:23:30 debian ser: set_mod_param_regex: No module matching acc_radius found | Jul 6 14:23:30 debian ser: parse error (72,19-20): Can't set module parameter what do you think the problem is? King regards, Pablo. __________________________________ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail

2 1

[Serusers] ser on win?
by Pacific 08 Jul '05

08 Jul '05

hi, somebody know how to modify the makefile of ser to compile it on a win system? Or is it impossible? thanks to all ___________________________________ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it

2 1

[Serusers] what the column "caller_UUID" and "callee_UUID" in acc table use for?
by Nilus 08 Jul '05

08 Jul '05

HI, serusers, Does anybody know what the column "caller_UUID" and "callee_UUID" in acc table use for? I use cvs head version. Is there any document about acc and avpops module if I have call_forward function in my ser using avpops? Regards, Nilus

2 1

[Serusers] SER with two domains
by Naresh Parmar 08 Jul '05

08 Jul '05

Hi, I am new to SER and currently configuring the SER to be used with two SIP domains. i.e. zone1.sip and zone2.sip. I have 2 SER proxies serving with one proxy serving zone1.sip and other proxy serving zone2.sip. Do I need to include a routing block in the se.cfg of these 2 proxies to make the interdomain call between zone1.sip and zone2.sip?? Or simply the DNS would help?? can someone suggest a soln for this to work?? best regards naresh --------------------------------- Sell on Yahoo! Auctions - No fees. Bid on great items.

2 1

Re: [Serusers] SERWEB error: "Function aggergate_method() doesn'texists."
by Karel Kozlik 08 Jul '05

08 Jul '05

Hi Leo, I didn't know about "runkit" extension. Thanks you mentioned it. I will adjust serweb to work with it. But I afraid that you can't use it now. To install classkit simply type on command line: pear install -f classkit If you get an error mentioning "phpize" you also might need to install some dev package, like php4-dev on Debian systems. regards Karel Leo napsal(a): > Karel, > > I looked all over the PHP site. I cannot figure out how to install a PHP > extension. > > I downloaded "runkit" which is the replacement for "classkit", but it is > just a directory with a bunch of XML and C files and no make file. I can't > figure out what to do. Do you know where I might learn how to install this > PHP extension on PHP 5? > > Leo P. > > -----Original Message----- > From: Karel Kozlik [mailto:karel@iptel.org] > Sent: Monday, July 04, 2005 3:04 AM > To: Leo > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] SERWEB error: "Function aggergate_method() > doesn'texists." > > Hi Leo, > yes, PHP5 users needs install Classkit extension. It contains some > functions from PHP4 which are missing in PHP5. > > Karel > > Leo napsal(a): > >>More info: >> >> >> >>This might be a PHP 5 issue. Any ideas? >> >> >> >>------------------------------------------------------------------------ >> >>*From:* serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] >>*On Behalf Of *Leo >>*Sent:* Friday, July 01, 2005 3:35 PM >>*To:* serusers(a)lists.iptel.org >>*Subject:* [Serusers] SERWEB error: "Function aggergate_method() >>doesn'texists." >> >> >> >>Dear serusers, >> >> >> >>I am running latest SER 0.9.3 with MySQL support successfully of Fedora >>Core 4. >> >> >> >>I installed SERWEB 0.9.3. My compliments to whoever wrote the INSTALL >>file, it is very good. >> >> >> >>I get this error when I try to open the >>http://mydomain.com/admin/index.php file: >> >> >> >>Function aggergate_method() doesn't exists. Try install Classkit > > extension. > >> >> >>Do I really need to do this or is something else wrong? >> >> >> >>Leo Papadopoulos >> >>leo(a)ltcjp.com <mailto:leo@ltcjp.com> >> >> >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > >

3 2

[Serusers] SERWEB error: "Function aggergate_method() doesn't exists."
by Leo 08 Jul '05

08 Jul '05

Dear serusers, I am running latest SER 0.9.3 with MySQL support successfully of Fedora Core 4. I installed SERWEB 0.9.3. My compliments to whoever wrote the INSTALL file, it is very good. I get this error when I try to open the http://mydomain.com/admin/index.php file: Function aggergate_method() doesn't exists. Try install Classkit extension. Do I really need to do this or is something else wrong? Leo Papadopoulos leo(a)ltcjp.com

3 4

[Serusers] MySQL accounting does not work in ser 0.8.12
by Yiannis Martis 08 Jul '05

08 Jul '05

3 2

[Serusers] silent users [ot]
by Iqbal 08 Jul '05

08 Jul '05

Hi Its a boring Friday, was wondering if there are any stats as to how many members there are on this list and how many are silent. Just trying to guess how big ser community is Iqbal

1 0

[Serusers] SerWeb0.9.3 show me the error :DB Error: no such table Why?
by zhaomin 08 Jul '05

08 Jul '05

I used serweb-0.9.3 and Ser CVS version, I can register and login but when I login and open the my account show me error: DB Error: no such table.but it's can open the data in the mysql .I have this error prompt at many place include admin,anyone have this prompt? How can I resolve it? Thanks advance ! ZhaoMin

5 8

[Serusers] Ser 0.8.14 to 0.9.3 database upgrade
by Luca Corti 08 Jul '05

08 Jul '05

Hello, I've used the ser_mysql.sh script to upgrade my database but when issuing ser_mysql.sh copy sernew I get MySql password for myadminuser: creating database sernew ... ERROR 1054 at line 111: Unknown column 'perms' in 'field list' and the new database has the tables but they are mostly empty. This is the output of the script with -x added: + DBNAME=ser + DBHOST=mydbserver + USERNAME=ser + DEFAULT_PW=mypass + ROUSER=serro + RO_PW=mypass2 + SQL_USER=myadminuser + CMD=mysql -h mydbserver -umyadminuser + DUMP_CMD=mysqldump -h mydbserver -umyadminuser -c -t + BACKUP_CMD=mysqldump -h mydbserver -umyadminuser -c + TABLE_TYPE=TYPE=MyISAM + USERCOL=username + GENHA1=gen_ha1 + FOREVER=2020-05-28 21:32:15 + DEFAULT_ALIASES_EXPIRES=2020-05-28 21:32:15 + DEFAULT_Q=1.0 + DEFAULT_CALLID=Default-Call-ID + DEFAULT_CSEQ=42 + DEFAULT_LOCATION_EXPIRES=2020-05-28 21:32:15 + export PW + '[' 2 -ne 0 ']' + prompt_pw ++ stty -g + savetty=500:5:bf:8a3b:3:1c:7f:15:4:0:1:0:11:13:1a:0:12:f:17:16:0:0:0:0:0:0:0:0:0:0:0:0:0:0:0:0 + printf 'MySql password for myadminuser: ' MySql password for myadminuser: + stty -echo + read PW + stty 500:5:bf:8a3b:3:1c:7f:15:4:0:1:0:11:13:1a:0:12:f:17:16:0:0:0:0:0:0:0:0:0:0:0:0:0:0:0:0 + echo + shift + '[' 1 -ne 1 ']' + tmp_file=/tmp/ser_mysql.32727 + ser_dump ser + '[' 1 -ne 1 ']' + mysqldump -h mydbserver -umyadminuser -c -t -pmypass ser + ret=0 + '[' 0 -ne 0 ']' + ser_create sernew no_init_user + '[' 2 -eq 1 ']' + '[' 2 -eq 2 ']' + INITIAL_USER= + echo 'creating database sernew ...' creating database sernew ... + sql_query + mysql -h mydbserver -umyadminuser -pmypass + ret=0 + '[' 0 -ne 0 ']' + ser_restore sernew /tmp/ser_mysql.32727 + '[' 2 -ne 2 ']' + sql_query sernew + mysql -h mydbserver -umyadminuser -pmypass sernew ERROR 1054 at line 111: Unknown column 'perms' in 'field list' + ret=1 + rm /tmp/ser_mysql.32727 + exit 1 Any ideas why it fails? thanks Luca -- Luca Corti PGP Key ID 1F38C091 BOFH excuse of the moment: Backbone Scoliosis

1 0

[Serusers] test call
by harry gaillac 08 Jul '05

08 Jul '05

Excuse me I did not thank people who called me for my tests. Thanks Olivier Taylor and others, Snom.com and Ingate.com. Send me mails if you need I call you to test your configuration. 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] mediaproxy.sock
by Abdul Lateef 08 Jul '05

08 Jul '05

Hello I downloaded the NAT example from onsip.org. but when i am trying to start ser. i am getting some error. I think coz of mediaproxy.sock file, because i cannot see file in my /var/run folder. here is my example. debug=7 fork=yes log_stderror=no listen=10.0.0.23 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:123@localhost/ser" 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/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:123@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) 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")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("10.0.0.23:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { 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(1); break; }; if (uri==myself) { 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(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); 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(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; 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(); }; } Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com ____________________________________________________ Sell on Yahoo! Auctions no fees. Bid on great items. http://auctions.yahoo.com/

5 4

[Serusers] voicemail serweb+sems
by harry gaillac 08 Jul '05

08 Jul '05

Hello all, How can we configure serweb+sems to work together for voicemail ? voicemail plugin search here for greetings here: # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for # - the file to be played is searched in the following order: # <announce_path>/<domainname>/<username>.wav # <announce_path>/<domainname>/<language>/<default_announce> # <announce_path>/<domainname>/<default_announce> # <announce_path>/<language>/<default_announce> # <announce_path>/<default_announce> # where <language> is taken from the body of P-Language header # of the request (if any). announce_path=/usr/local/lib/sems/audio/ voicemail Tab in serweb allow to upload or download greetings How can we record greetings via sems ivr plugin (record.py) with a phone which put this .wav file in a shared directory (with NFS) for sems and serweb ? 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

[Serusers] Why OutBound proxy return 483 too many hops?
by gao 08 Jul '05

08 Jul '05

I config a OP with public IP.And I encount a problem.Two UAs can't invite each other from a same OP. ua1(private ip)--->nat--->OP1---->register1 ua2(public ip)----->OP1-->register1 ua1---invite---->OP1 When ua1 invite ua2,OP1 return 483.The same from ua2 to ua1. What's wrong? Is my configure file something wrong? Can somebody here help me? # # $Id: ser.cfg,v 1.24 2003/11/11 15:32:36 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 */ #log_stderror=yes #debug=4 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=5070 listen="222.44.22.60" children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/home/ser-0.9.3/modules/sl/sl.so" loadmodule "/home/ser-0.9.3/modules/tm/tm.so" loadmodule "/home/ser-0.9.3/modules/rr/rr.so" loadmodule "/home/ser-0.9.3/modules/maxfwd/maxfwd.so" loadmodule "/home/ser-0.9.3/modules/usrloc/usrloc.so" loadmodule "/home/ser-0.9.3/modules/textops/textops.so" loadmodule "/home/ser-0.9.3/modules/nathelper/nathelper.so" loadmodule "/home/ser-0.9.3/modules/uri/uri.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) 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") modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- 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; }; #--------------- if (method == "REGISTER" || ! search("^Record-Route:")) { append_hf("Alex-hint: NAThelper\r\n"); fix_nated_register(); fix_nated_contact(); if (method == "INVITE") { append_hf("Alex-hint: SDP rewritten\r\n"); force_rtp_proxy(); }; force_rport(); setflag(6); }; #--------------- # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if(method != "REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); }; force_rtp_proxy("l"); }; append_hf("Alex-hint: Loose Route\r\n"); route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; if (uri!=myself) { route(1); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (nat_uac_test("19")) { setflag(6); }; if (uri!=myself) { route(1); break; }; if (isflagset(6)) { force_rport(); #fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } onreply_route[1] { if (isflagset(6) || status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); append_hf("NAT: 1\r\n"); }; }; }

1 0

[Users] Accounting Information and Call Transfers
by Günther Starnberger 07 Jul '05

07 Jul '05

Hello, I'm currently working on a VOIP network where the phones register to an OpenSER server - we also have a PSTN Gateway (Asterisk). To be able to account the PSTN calls of a user we insert a custom header "X-Account" which stores information about the user account. We can't just use the From-Header for this as we allow users to e.g. create unconditional forwards which forward the call to a defined SIP uri when the user isn't available. If somebody creates a forward to the PSTN it is still his account which should be accounted and which is therefor put in the X-Account header. This part works fine, but I'm still not sure how to handle call transfers - e.g. when a user wants to transfer his conversational partner to the PSTN. An unattended transfer in SIP is implemented with a REFER request which causes the remote phone to send an INVITE to the SIP uri in question[1]. I'm not sure how to account these transfers. On a traditional PSTN network the person who caused the transfer would be the one who is paying for the call. But I see no solution how to implement this behaviour in SIP (because the old call isn't transfered, but the conversational partner starts a new call with the INVITE request). Is it somehow possible to detect if a call was caused by a transfer so that I can insert the right X-Account header? Do I need a B2BUA like Asterisk to implement the traditional behaviour? How do other VOIP providers solve this problem (IMO call transfers shouldn't be such an uncommon feature). tia for all suggestions on how to solve this problem /gst [1] http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-08.…

2 4

[Serusers] having problems using avpops module
by Jack Wei 07 Jul '05

07 Jul '05

i'm trying to load the db and check if the username for the "to" sip uri is in the database, but my script is displaying the following errors: 0(0) AVPops - initializing 0(0) ERROR:avpops:parse_check_value: unknown flag <u> 0(0) ERROR:avpops:fixup_check_avp: failed to parse checked value 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration Here's my ser.cfg: less /etc/ha.d/ser.cfg /etc/ha.d/ser.cfg: No such file or directory [root@SER02 ser]# less /usr/local/etc/ser/ser.cfg # # $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ fork=yes # Uncomment these lines to enter debugging mode #debug=3 # debug level (cmd line: -dddddddddd) #fork=no log_stderror=yes # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) alias="SER02" 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" fifo_db_url="mysql://ser:heslo@10.1.201.107/ser" #user=ser #group=ser #fifo_user=ser # owner of the ser fifo #fifo_group=ser #fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) # # ------------------ 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/textops.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/avpops.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", 1) modparam("usrloc", "timer_interval", 10) # -- db_url params -- modparam("acc|auth_db|domain|group|permissions|speeddial|uri_db|usrloc|xdz_tools", "db_url","mysql://ser:heslo@10.1.201.107/ser") # -- auth params -- modparam("auth_db", "calculate_ha1", yes) 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) # -- tm params -- modparam("tm", "fr_inv_timer", 90) # -- avp params -- modparam("avpops", "avp_url", "mysql://ser:heslo@10.1.201.107/ser") modparam("avpops", "avp_table", "location") #modparam("avpops", "use_domain", "1") #modparam("avpops", "uuid_column", "callid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") #modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "contact") #modparam("avpops", "type_column", "type") #modparam("avpops", "avp_aliases", "voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702") # To use more than one tables example #modparam("avpops", "db_scheme", "scheme1:table=location;username_col=username;domain_col=domain;value_col=contact;value_type=string ") # ------------------------- 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; }; # 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) { xlog("L_INFO", "%rm New URI = %ru from %ct\n\n"); if (method=="REGISTER") { # digest authentication if (!www_authorize("localhost", "subscriber")) { www_challenge("localhost", "0"); 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 xlog("L_INFO", "pre-rr(): %rm to URI %ru\n\n"); # record_route(); record_route_preset( "10.1.201.189:5060" ); xlog("L_INFO", "post-rr(): %rm to URI %ru\n\n"); # 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"); xlog("L_INFO", "loose_route(): Looking up %rm URI %ru from %ct\n\n"); lookup("location"); xlog("L_INFO", "loose_route(): t_relay() %rm to URI %ru\n\n"); route(1); break; }; if ( uri =~ "sip:[0-9]+@(10.1.101.(159|250)|192.168.10.31|64.77.239.90)" ) { if ( method == "INVITE" || method == "ACK" || method == "CANCEL" || method == "REFER" || method == "BYE" ) { xlog("L_INFO", "\n%rm came from a VoIP phone (%ct)\n\n"); # Making an outbound cal, which starts with a "9", to a PSTN phone if( uri =~ "^sip:9[0-9]+@*" ) { append_hf( "P-hint: OUTBOUND\r\n" ); route(2); rewritehostport("10.1.101.152:5060"); # forward( localhost, 5061 ); xlog("L_INFO", "URI is a PSTN phone...sending URI (%ru) to 10.1.101.152\n\n"); # xlog("L_INFO", "URI is a PSTN phone...sending URI (%ru) to Asterisk\n\n"); route(1); break; } # Making an outbound call to another VoIP phone else { # Redirect call to voicemail when destination is not registered append_hf( "P-hint: USRLOC\r\n" ); if( !lookup("location") && !(uri =~ "^sip:11111111111@*") ) { # prefix("*"); xlog("L_INFO", "\n\n%ru is not online (in usrloc)\n"); # route(3); route(4); } else { setflag(5); }; # forward( uri:host, uri:port ); # forward( localhost, 5061 ); if ( isflagset(5) ) { xlog("L_INFO", "URI is a VoIP phone...forwarding URI (%ru) to destination\n\n"); # xlog("L_INFO", "URI is a VoIP phone...forwarding URI (%ru) to Asterisk\n\n"); resetflag(5); route(1); }; break; }; }; route(1); break; }; # route(1); } #--------------------------------------------------------------------- route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP xlog("L_INFO", "\n\n%ru is being t_relay()ed\n"); if (!t_relay()) { xlog("L_INFO", "\n\n%ru t_relay() error\n"); sl_reply_error(); break; }; } #--------------------------------------------------------------------- # All PSTN calls require a prefix of "0010" route[2] { if ( uri =~ "^sip:0010[0-9]+@*" ) { xlog("L_INFO", "URI has 0010 prefix already\n\n"); } else { xlog("L_INFO", "URI does not has 0010 prefix...attaching prefix\n\n"); strip(1); prefix("0010"); }; } #--------------------------------------------------------------------- # Check database for actual user address route[4] { if ( avp_db_load("$to/username","s:") && avp_check("s:username","eq/$to/username/g") ) { xlog("L_INFO", "\n\n%ru is online (in db)\n"); setflag(5); break; }; xlog("L_INFO", "\n\n%ru is not online (in db)\n"); sl_send_reply( "404", "User does not exist" ); } Can someone tell me what i'm doing wrong? Thanks. Jack ____________________________________________________ Sell on Yahoo! Auctions no fees. Bid on great items. http://auctions.yahoo.com/

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Re: 回复： Re: [Serusers] SERWEB 0.9.3 ERROR: bad username or passwd
by Karel Kozlik 07 Jul '05

07 Jul '05

Hi, please (CC) the mailing list in your replies. Try enable logging in serweb. Set $config->enable_loging = true; $config->log_level = "PEAR_LOG_DEBUG"; in config file and see the log file. If it will not help you, send the logfile and content of your subscriber table to me. Karel han yibing napsal(a): > Hi > > I installed SERWEB 0.9.3 version > and i login mysql to check subscriber and > admin_privileges tables ,all status is ok. > > > > --- Karel Kozlik <karel(a)iptel.org>写道: > > >>Hello, >>which version of serweb are you useing? >> >>Please make sure your username, password and domain >>values are exactly >>matched the values of your subscriber table. >> >>regards Karel >> >>han yibing napsal(a): >> >>> "bad username or password" display when i have >>>default username and password (admin/heslo) login >>>SERWEB,my domain is local IP address. >>> >>>would you link to tell why and how to do it ? >>> >>>thanks. >>> >>> >>> >>> >>> >>> >>> >>> >>> >> > ___________________________________________________________ > >>>雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 >>>http://cn.mail.yahoo.com/?id=77071 >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> > > > > > > > > ___________________________________________________________ > 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 > http://cn.mail.yahoo.com/?id=77071 >

1 0

[Serusers] FIFO_RELAY and fifo_server.php
by Daryl Sanders 07 Jul '05

07 Jul '05

I have searched high & low but can't seem to find any info on how to use fifo_server.php for fifo relay. I have setup serweb (0.9.3) on a different box than ser, and it works fine except for the fifo stuff. Can anyone give me a clue on how to configure this properly. Thanks! - Daryl

2 1

[Users] Default Configuration?
by Adam Sherman 07 Jul '05

07 Jul '05

As I anm struggling to build a configuration with OpenSER which operates a multi-domain proxy, sends calls to SEMS for voicemail after a timeout and uses a third-party SIP gateway for PSTN in/out; I wonder if someone has a configuration for this somewhere already. It does seem like a rather "normal" config... Does anyone have such a configuration? It would help me tremendously! Thank you, A. -- Adam Sherman Technologist

2 2

[Users] SEMS AVP
by Adam Sherman 07 Jul '05

07 Jul '05

I'm attempting to configure OpenSER to feed SEMS the correct information for voicemail, namely the email address. SEMS documentation referes to the avp.so module, which I understand is no longer relevant. If all I need to do is pull the email address out of the subscribers table and feed it to SEMS, what openser config do I need? Thanks, A. -- Adam Sherman Technologist

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[Serusers] Configuration example - no register, no auth
by Chris Mason 07 Jul '05

07 Jul '05

I need some newbie help to configure ser so that it forwards all requests to another server. I don't need mysql authentication, or any authentication, just a plain proxy that runs on port 5080. Can someone help me modify the config? All the examples I have seen spend a lot of time on mysql authentication, but little on the proxy routng. Chris

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[Serusers] Nat_uac_test failing with ATA v3.2 behind NAT, ATA v3.1 works
by Jon Mansey 07 Jul '05

07 Jul '05

Hi Serusers, I have 2 ATAs, one is V3.1 and one is V3.2, they are both behind NAT and are behaving differently when registering. With 3.2, the nat_uac_test does not apparently detect it is behind NAT and leaves the flag at 0 in the location table. V3.1 works fine. I can see why nat_uac_test() may be failing, because the ATA seems to have auto-magically detected its public IP and is using it in the via: header, thus receive!=via test fails! Also the contact has the public IP, so the RFC1813 test fails. I have not configured the NAT IP or server in the ATA as the NAT public IP is dynamic. Although the v3.2 ATA seems to have figured out its public IP, it still uses its private IP in the INVITE sdp c= headers, so we get no media. And because the nat flag is 0, the rtpproxy doesn't kick in and fix things. Notice the call-id of the register message includes the private ip. Anyone familiar with ATA firmware versions and features know the magic incarnation to shut off the "auto-detect public IP", ahem, "feature" in v3.2? Or any other suggestions for how to make nat_uac_test() get a positive when it gets one of these smart REGISTERs from behind NAT? Im running the old version of SER 0.8.12 still where nat_uac_test only has 1 2 or 3 options, not the new one that has "19", would that make a difference, upgrading to v0.9? Here are the ngreps of REGISTERs from the 2 versions: (public Ips masked) V3.1 works fine, nat_uac_test results in flag 1 in location table, rtpproxy works fine >From ata v3.1 behind nat1 to ser server: U <nat1 pub ip>:49121 -> <pub ip of ser>:5060 REGISTER sip:sip.mycompany.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.11: 5060..From: <sip:20210@sip.mycompany.com;user=phone>;tag=933299358..To : <sip:20210@sip.mycomapny.com;user=phone>..Call-ID: 4033019825(a)192.16 8.0.11..CSeq: 1 REGISTER..Contact: <sip:20210@192.168.0.11:5060;user=phone; transport=udp>;expires=3600..User-Agent: Cisco ATA 186 v3.1.0 atasip (0402 11A)..Content-Length: 0.... And from ata with v3.2 U <pub ip of nat2>:5060 -> <pub ip of ser>:5060 REGISTER sip:sip.mycompany.com SIP/2.0..Via: SIP/2.0/UDP <pub ip of nat2> :5060;branch=z9hG4bK1c6d78542c385f5e..From: <sip:202111@sip.mycompany.com ;user=phone>;tag=2453464613..To: <sip:202111@sip.mycomapny.com ;user=phone>..Call-ID: 1193202309@172.16.1.100..CSeq: 1 REGISTER..Con tact: <sip:202111@<pub ip of nat2>:5060;user=phone;transport=udp>;expire s=3600..User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A)..Content-Length: 0.... Thx in advance, Jon

2 1

[Serusers] serweb-0.9.3.tgz
by harry gaillac 07 Jul '05

07 Jul '05

Hi all, I downloaded serweb-0.9.3.tgz for ser-0.9.3 I just (re)install serweb however in config/config_data_layer.php i changed. it's ok for me !! Regards Harry $config->data_sql->table_user_preferences_types="preferences_types"; to $config->data_sql->table_user_preferences_types="usr_preferences_types"; I set in sql file : scripts/sql/usr_preferences.sql # $Id: usr_preferences.sql,v 1.2.2.1 2005/06/28 08:50:08 kozlik Exp $ # # Dumping data for table 'usr_preferences_types' # INSERT INTO usr_preferences_types (att_name, att_rich_type, att_raw_type, att_type_spec, default_value) VALUES("fw_voicemail", "boolean", "1", NULL, "0"); INSERT INTO usr_preferences_types (att_name, att_rich_type, att_raw_type, att_type_spec, default_value) VALUES("sw_user_status_visible", "boolean", "1", NULL, "1"); INSERT INTO usr_preferences_types (att_name, att_rich_type, att_raw_type, att_type_spec, default_value) VALUES("send_daily_missed_calls", "boolean", "1", NULL, "0"); ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] URGENT! AS5350 <-> SER incompatibility problem
by Victor Huertas Garcia 07 Jul '05

07 Jul '05

Hi everybody! recently I have been testing calls to/from ISDN with a AS5350. My SER version is 0.8.14 and when I make a call from ISDN towards a SIP UA registered in the SER the problem is that the AS5350 is no aware of Call Release. I explain it in more detail... 1. AS5350 sends an INVITE to the SER with destination number 93222222 2. SER forwards the INVITE to the corresponding SIP UA. 3. UA reponds with a 180 Ringing and it is forwarded to the AS5350 by the SER 4. When I pick up the phone, the SIP UA sends a 200 OK message, which is frowarded by the SER as well. 5. AS5350 replies with a ACK to the 200 OK but beware!! the AS5350 has changed UDP source port!! it is different from the one set in INVITE. 6. The SIP UA hangs up the call by sending a BYE message to the SER (it is Record Routing) 7. The SER forwards the BYE to the AS5350 but with the old UDP port!! 8. AS5350 sends ICMP Destination Unreachable (Cause code: port unreachable) every time that the SER retries the BYE. My guess is that the AS5350 expects the BYE packet through the new UDP port... Has anybody found such a problem? Thanks Victor

2 3

[Serusers] RE: SERWEB error: "Function aggergate_method() doesn't exists."
by Gareth C. Fowler 07 Jul '05

07 Jul '05

I'm currently running Fedora Core 4, SER 0.9.2, and have followed the instruction of running: # pear install -f classkit and when I run: # pear list I receive the following output: Installed packages: =================== Package Version State Archive_Tar 1.1 stable Console_Getopt 1.2 stable DB 1.7.6 stable HTML_Template_IT 1.1 stable HTTP 1.3.5 stable Mail 1.1.4 stable Net_SMTP 1.2.6 stable Net_Socket 1.0.6 stable Net_UserAgent_Detect 2.0.1 stable PEAR 1.3.5 stable XML_Parser 1.2.6 stable XML_RPC 1.2.2 stable classkit 0.4 beta runkit 0.3.0 beta but it's still giving me the "Function aggregate_methods() doesn't exists. Try install Classkit extension. http://pecl.php.net/package/classkit" error. I even installed the runkit which is supposed to be the next version up. Can anyone recommend anything to resolve this error?? I'm more of a perl man, so I'm lost with PHP. Thanks. Gareth

2 2

[Serusers] Nat_uac_test failing with ATA v3.2 behind NAT, v3.1 works
by Jon Mansey 07 Jul '05

07 Jul '05

Hello Serusers, I have 2 ATAs, one is V3.1 and one is V3.2, they are both behind NAT and are behaving differently when registering. With 3.2, the nat_uac_test does not apparently detect it is behind NAT and leaves the flag at 0 in the location table. V3.1 works fine. I can see why nat_uac_test() may be failing, because the ATA seems to have auto-magically detected its public IP and is using it in the via: header, thus receive!=via test fails! Also the contact has the public IP, so the RFC1813 test fails. I have not configured the NAT IP or server in the ATA as the NAT public IP is dynamic. Although the v3.2 ATA seems to have figured out its public IP, it still uses its private IP in the INVITE sdp c= headers, so we get no media. And because the nat flag is 0, the rtpproxy doesn't kick in and fix things. Notice the call-id of the register message includes the private ip. Anyone familiar with ATA firmware versions and features know the magic incarnation to shut off the "auto-detect public IP", ahem, "feature" in v3.2? Or any other suggestions for how to make nat_uac_test() get a positive when it gets one of these smart REGISTERs from behind NAT? Im running the old version of SER 0.8.12 still where nat_uac_test only has 1 2 or 3 options, not the new one that has "19", would that make a difference, upgrading to v0.9? Here are the ngreps of REGISTERs from the 2 versions: (public Ips masked) V3.1 works fine, nat_uac_test results in flag 1 in location table, rtpproxy works fine >From ata v3.1 behind nat1 to ser server: U <nat1 pub ip>:49121 -> <pub ip of ser>:5060 REGISTER sip:sip.mycompany.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.11: 5060..From: <sip:20210@sip.mycompany.com;user=phone>;tag=933299358..To : <sip:20210@sip.mycomapny.com;user=phone>..Call-ID: 4033019825(a)192.16 8.0.11..CSeq: 1 REGISTER..Contact: <sip:20210@192.168.0.11:5060;user=phone; transport=udp>;expires=3600..User-Agent: Cisco ATA 186 v3.1.0 atasip (0402 11A)..Content-Length: 0.... And from ata with v3.2 U <pub ip of nat2>:5060 -> <pub ip of ser>:5060 REGISTER sip:sip.mycompany.com SIP/2.0..Via: SIP/2.0/UDP <pub ip of nat2> :5060;branch=z9hG4bK1c6d78542c385f5e..From: <sip:202111@sip.mycompany.com ;user=phone>;tag=2453464613..To: <sip:202111@sip.mycomapny.com ;user=phone>..Call-ID: 1193202309@172.16.1.100..CSeq: 1 REGISTER..Con tact: <sip:202111@<pub ip of nat2>:5060;user=phone;transport=udp>;expire s=3600..User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A)..Content-Length: 0.... Thx in advance, Jon

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[Serusers] Problems compiling SER 0.9.3 with lcr module
by Alberto Cruz 07 Jul '05

07 Jul '05

Hi I'm trying to compile the lcr backported module from Onsip.org but I'm getting the following error: make[1]: Entering directory `/root/sources/sip_router/modules/lcr' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -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 -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 lcr_mod.c -o lcr_mod.o lcr_mod.c: In function `load_gws': lcr_mod.c:591: error: `HDR_FROM_F' undeclared (first use in this function) lcr_mod.c:591: error: (Each undeclared identifier is reported only once lcr_mod.c:591: error: for each function it appears in.) lcr_mod.c: In function `load_contacts': lcr_mod.c:890: error: too many arguments to function `next_branch' lcr_mod.c:912: error: too many arguments to function `next_branch' make[1]: *** [lcr_mod.o] Error 1 make[1]: Leaving directory `/root/sources/sip_router/modules/lcr' Does anybody know if I'm missing something? Regards Alberto Cruz

4 6

[Serusers] simple ser configurtion
by chris 07 Jul '05

07 Jul '05

hi, i installed ser with mysql support and authentication on solaris9, i have 2 ordinary phones, connected to a linksys 2phone port pap2 device. what i need a simple ser routing configuration where i can call from one of the phones to the other phone on the same pap2 device using ser. pls. help.. thnks in advance. chris.

2 1

[Serusers] session duration accounting
by Ryan Pagquil 07 Jul '05

07 Jul '05

guys, just want to ask if i can make ser to account session durations for billing purposes? 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

6 20

[Serusers] installing sipsak error
by chris 07 Jul '05

07 Jul '05

i'm Installaing sipsak on a solaris 9, when i run ./configure i got this: checking ctype.h presence... yes checking for ctype.h... yes checking arpa/inet.h usability... yes checking arpa/inet.h presence... yes checking for arpa/inet.h... yes checking getopt.h usability... no checking getopt.h presence... no checking for getopt.h... no configure: error: missing required header (see above) where can i get the getopt.h? what package will i install? thnks

2 1

Re: [Serusers] SDP c= field mangled
by zhaomin 07 Jul '05

07 Jul '05

Hello! I want to configure ser+asterisk in a same host, I don't know asterisk if run correct when I watch the asterisk process is run ,but when I used the command astman to handle asterisk ,it's show be connect refuse, cann't open port:5038 .Could you give me some hint to resolve it? and Could you pasted you ser.cfg and asterisk's conf file for a referrence? regard! ZhaoMin ----- Original Message ----- From: "Dave" <ddx66(a)yahoo.com> To: <serusers(a)iptel.org> Sent: Tuesday, July 05, 2005 10:19 PM Subject: [Serusers] SDP c= field mangled > Hi,I am experiencing a strange issue. I am forwarding > calls to Asterisk voicemail upon no-response. So, in > failure route, I simply check the status 408 or 486 > and if true, I jump to a routing-block. In there, I > have the following: > ----- > revert_uri(); > rewritehost(Asterisk-IP); > append_branch(); > if (isflagset(6) || isflagset(7)) use_media_proxy(); > if (!t_relay()...) > ------ > > > Now, Asterisk does receive the invite, except that the > SDP in the SIP invite looks like (PAY ATTENTION TO c= > field). > > ------------------------------- > v=0 > o=208500512 8000 8001 IN IP4 192.168.1.104 > s=SIP Call > c=IN IP4 213.189.X.Y213.189.X.Y > t=0 0 > m=audio 3532235322 RTP/AVP 0 8 4 18 3 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:3 GSM/8000 > a=ptime:20 > ------------------------------------------ > > The c= field has the same IP twice and thus the INVITE > is rejected by Asterisk. > > Is this a bug and did someone have such an issue? > Thanks > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Ser+SEMS, How?
by Antonio Membrives 07 Jul '05

07 Jul '05

Hello i already running Ser 0.9.3 with Mysql and Serweb, but I cant know how to configure Ser with SEMS, I read the doc install, but no work, any basic example of configuration ser+sems for newbye user. Thanks and sorry for my bad english

1 0

[Users] OpenSER & SEMS, fifo's, permissions and versions
by Adam Sherman 07 Jul '05

07 Jul '05

I'm running OpenSER 0.10.x and SEMS HEAD. A few questions: 1. What permissions to the fifo's normally need and how does that affect the uid/gid you run openser & sems as? 2. Will those versions work together? Thanks, A. -- Adam Sherman Technologist

2 6

[Serusers] SERWEB 0.9.3 ERROR: bad username or passwd
by han yibing 07 Jul '05

07 Jul '05

"bad username or password" display when i have default username and password (admin/heslo) login SERWEB,my domain is local IP address. would you link to tell why and how to do it ? thanks. ___________________________________________________________ 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 http://cn.mail.yahoo.com/?id=77071

2 1

[Serusers] listening on more than one port
by jeff kwong 07 Jul '05

07 Jul '05

Hi! Is there a way for SER to listen in more than one port? Thanks! _jeff

2 2

[Serusers] RE: mySQL PSTN Gateway
by Abdul Lateef 07 Jul '05

07 Jul '05

HI, Dear Iqbal, i checked in /usr/local/lib/ser/modules But i am not able to find lcr module. If it is second party module let me please know from where i can get it? Thank You Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

1 0

[Serusers] multiple SER with 1 mysql server
by Chia Huey Lim 07 Jul '05

07 Jul '05

Hi. I have a centralized mysql server for SER database, and have 2 SER server connecting to the database. I failed to bridge UA 1 that is connected to SER A and the other UA 2 that is connected to SER B. Quite confused with the domain and alias actually, I wish to differentiate UA that register to different SER, thus I configure such that SER A has its IP as domain name and SER B has its IP as its domain name. Is it inappropriate to do that? What is the proper way to do so? Any guideline that I can bridge UA that is connected to different proxy to talk to each other? Thanks in advance. =) Regards, Chia

1 0

[Serusers] PSTN call receives a bye after 5 seconds using SER+B2BUA+MEDIAPROXY
by Alberto Cruz 06 Jul '05

06 Jul '05

Hi all I have been looking at the debugs in order to try to figure out If I'm doing something wrong at my SER.CFG I have the following: UA ----> SER -----> B2BUA -----> SER ------> PSTN-GW 1. The user agent place a call to the SER proxy. 2. SER Proxy receives the call and forwads to B2BUA (VOCAL at the same server using UDP port 5065) 3. B2BUA receives the calls activates the Billing supervision and Sends an invite for a new call to the SER. 4. SER receives the call from the B2BUA and rewrite host and port to the PSTN-GW. 5. PSTN-GW receives the invite from the SER and sends back the ACK. 6. Inmediatly from some reason SER is sending a BYE to the call. I'm attaching a debug from the SER and my SER.CFG file Could anybody help me please? Regards Alberto Cruz Jun 25 08:43:31 sipsrv ser[1936]: SIP Request: Jun 25 08:43:31 sipsrv ser[1936]: method: <INVITE> Jun 25 08:43:31 sipsrv ser[1936]: uri: <sip:018183324166@sipsrv.conecta-te.com> Jun 25 08:43:31 sipsrv ser[1936]: version: <SIP/2.0> Jun 25 08:43:31 sipsrv ser[1936]: parse_headers: flags=1 Jun 25 08:43:31 sipsrv ser[1936]: Found param type 232, <branch> = <z9hG4bK-46d68929>; state=16 Jun 25 08:43:31 sipsrv ser[1936]: end of header reached, state=5 Jun 25 08:43:31 sipsrv ser[1936]: parse_headers: Via found, flags=1 Jun 25 08:43:31 sipsrv ser[1936]: parse_headers: this is the first via Jun 25 08:43:31 sipsrv ser[1936]: After parse_msg... Jun 25 08:43:31 sipsrv ser[1936]: preparing to run routing scripts... Jun 25 08:43:31 sipsrv ser[1936]: parse_headers: flags=128 Jun 25 08:43:31 sipsrv ser[1936]: end of header reached, state=9 Jun 25 08:43:31 sipsrv ser[1936]: DEBUG: get_hdr_field: <To> [42]; uri=[sip:018183324166@sipsrv.conecta-te.com] Jun 25 08:43:31 sipsrv ser[1936]: DEBUG: to body [<sip:018183324166@sipsrv.conecta-te.com>^M ] Jun 25 08:43:31 sipsrv ser[1936]: get_hdr_field: cseq <CSeq>: <101> <INVITE> Jun 25 08:43:31 sipsrv ser[1936]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Jun 25 08:43:31 sipsrv ser[1936]: DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 Jun 25 08:43:31 sipsrv ser[1936]: parse_headers: flags=64 Jun 25 08:43:31 sipsrv ser[1936]: DEBUG: add_param: tag=7a83401cfe54b69o0 Jun 25 08:43:31 sipsrv ser[1936]: end of header reached, state=29 Jun 25 08:43:31 sipsrv ser[1936]: parse_headers: flags=256 Jun 25 08:43:31 sipsrv ser[1936]: DEBUG: get_hdr_body : content_length=316 Jun 25 08:43:31 sipsrv ser[1936]: found end of header Jun 25 08:43:31 sipsrv ser[1936]: find_first_route: No Route headers found Jun 25 08:43:31 sipsrv ser[1936]: loose_route: There is no Route HF Jun 25 08:43:31 sipsrv ser[1936]: grep_sock_info - checking if host==us: 21==11 && [sipsrv.conecta-te.com] == [200.94.75.6] Jun 25 08:43:31 sipsrv ser[1936]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:31 sipsrv ser[1936]: grep_sock_info - checking if host==us: 21==11 && [sipsrv.conecta-te.com] == [200.94.75.6] Jun 25 08:43:31 sipsrv ser[1936]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:31 sipsrv ser[1936]: grep_sock_info - checking if host==us: 21==11 && [sipsrv.conecta-te.com] == [200.94.75.6] Jun 25 08:43:31 sipsrv ser[1936]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:31 sipsrv ser[1936]: grep_sock_info - checking if host==us: 21==11 && [sipsrv.conecta-te.com] == [200.94.75.6] Jun 25 08:43:31 sipsrv ser[1936]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:31 sipsrv ser[1936]: Hey this request is NOT from B2BUA Jun 25 08:43:31 sipsrv ser[1936]: FORWARDING TO B2BUA Jun 25 08:43:31 sipsrv ser[1936]: parse_headers: flags=-1 Jun 25 08:43:31 sipsrv ser[1936]: parse_headers: flags=64 Jun 25 08:43:31 sipsrv ser[1936]: parse_headers: flags=64 Jun 25 08:43:31 sipsrv ser[1936]: parse_headers: flags=-1 Jun 25 08:43:31 sipsrv ser[1936]: check_via_address(200.67.33.247, 172.31.253.127, 0) Jun 25 08:43:31 sipsrv ser[1936]: Sending: INVITE sip:018183324166@sipsrv.conecta-te.com SIP/2.0^M Record-Route: <sip:018183324166@200.94.75.6:5060;nat=yes;ftag=7a83401cfe54b69o0;lr=on>^M Via: SIP/2.0/UDP 200.94.75.6;branch=0^M Via: SIP/2.0/UDP 172.31.253.127:5060;rport=60425;received=200.67.33.247;branch=z9hG4bK-46d68929^M From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=7a83401cfe54b69o0^M To: <sip:018183324166@sipsrv.conecta-te.com>^M Call-ID: b5fcf1e1-baf4a709(a)172.31.253.127^M CSeq: 101 INVITE^M Max-Forwards: 16^M Contact: 8183651723 <sip:8183651723@200.67.33.247:60425>^M Expires: 240^M User-Agent: Linksys/PAP2-2.0.12(LS)^M Content-Length: 316^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: x-sipura^M Content-Type: application/sdp^M P-hint: PASANDO POR EL B2BUA^M P-hint: APLICO NAT AL B2BUA^M ^M v=0^M o=- 23012599 23012599 IN IP4 172.31.253.127^M s=-^M c=IN IP4 172.31.253.127^M t=0 0^M m=audio 16458 RTP/AVP 18 0 4 8 100 101^M a=rtpmap:18 G729a/8000^M a=rtpmap:0 PCMU/800 Jun 25 08:43:31 sipsrv ser[1936]: orig. len=884, new_len=1105, proto=1 Jun 25 08:43:31 sipsrv ser[1936]: DEBUG:destroy_avp_list: destroying list (nil) Jun 25 08:43:31 sipsrv ser[1936]: receive_msg: cleaning up Jun 25 08:43:31 sipsrv ser[1929]: SIP Request: Jun 25 08:43:31 sipsrv ser[1929]: method: <INVITE> Jun 25 08:43:31 sipsrv ser[1929]: uri: <sip:018183324166@200.94.75.6:5060> Jun 25 08:43:31 sipsrv ser[1929]: version: <SIP/2.0> Jun 25 08:43:31 sipsrv ser[1929]: parse_headers: flags=1 Jun 25 08:43:31 sipsrv ser[1929]: end of header reached, state=5 Jun 25 08:43:31 sipsrv ser[1929]: parse_headers: Via found, flags=1 Jun 25 08:43:31 sipsrv ser[1929]: parse_headers: this is the first via Jun 25 08:43:31 sipsrv ser[1929]: After parse_msg... Jun 25 08:43:31 sipsrv ser[1929]: preparing to run routing scripts... Jun 25 08:43:31 sipsrv ser[1929]: parse_headers: flags=128 Jun 25 08:43:31 sipsrv ser[1929]: end of header reached, state=9 Jun 25 08:43:31 sipsrv ser[1929]: DEBUG: get_hdr_field: <To> [42]; uri=[sip:018183324166@sipsrv.conecta-te.com] Jun 25 08:43:31 sipsrv ser[1929]: DEBUG: to body [<sip:018183324166@sipsrv.conecta-te.com>^M ] Jun 25 08:43:31 sipsrv ser[1929]: get_hdr_field: cseq <CSeq>: <101> <INVITE> Jun 25 08:43:31 sipsrv ser[1929]: DEBUG:maxfwd:is_maxfwd_present: value = 16 Jun 25 08:43:31 sipsrv ser[1929]: parse_headers: flags=64 Jun 25 08:43:31 sipsrv ser[1929]: check_via_address(200.94.75.6, 200.94.75.6, 0) Jun 25 08:43:31 sipsrv ser[1929]: parse_headers: flags=33554432 Jun 25 08:43:31 sipsrv ser[1929]: DEBUG: add_param: tag=7a83401cfe54b69o0 Jun 25 08:43:31 sipsrv ser[1929]: end of header reached, state=29 Jun 25 08:43:31 sipsrv ser[1929]: parse_headers: flags=256 Jun 25 08:43:31 sipsrv ser[1929]: DEBUG: get_hdr_body : content_length=316 Jun 25 08:43:31 sipsrv ser[1929]: found end of header Jun 25 08:43:31 sipsrv ser[1929]: find_first_route: No Route headers found Jun 25 08:43:31 sipsrv ser[1929]: loose_route: There is no Route HF Jun 25 08:43:31 sipsrv ser[1929]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:31 sipsrv ser[1929]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:31 sipsrv ser[1929]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:31 sipsrv ser[1929]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:31 sipsrv ser[1929]: parse_headers: flags=64 Jun 25 08:43:31 sipsrv ser[1929]: check_via_address(200.94.75.6, 200.94.75.6, 0) Jun 25 08:43:31 sipsrv ser[1929]: parse_headers: flags=33554432 Jun 25 08:43:31 sipsrv ser[1929]: lookup(): '018183324166' Not found in usrloc Jun 25 08:43:31 sipsrv ser[1929]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:31 sipsrv ser[1929]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:31 sipsrv ser[1929]: lookup(): '018183324166' Not found in usrloc Jun 25 08:43:31 sipsrv ser[1929]: DEBUG: t_newtran: msg id=9 , global msg id=1 , T on entrance=0xffffffff Jun 25 08:43:31 sipsrv ser[1929]: parse_headers: flags=-1 Jun 25 08:43:31 sipsrv ser[1929]: parse_headers: flags=60 Jun 25 08:43:31 sipsrv ser[1929]: t_lookup_request: start searching: hash=21902, isACK=0 Jun 25 08:43:31 sipsrv ser[1929]: DEBUG: proceeding to pre-RFC3261 transaction matching Jun 25 08:43:31 sipsrv ser[1929]: DEBUG: t_lookup_request: no transaction found Jun 25 08:43:31 sipsrv ser[1929]: DBG: trans=0x4057c048, callback type 1, id 0 entered Jun 25 08:43:31 sipsrv ser[1929]: SER: new INVITE Jun 25 08:43:31 sipsrv ser[1929]: parse_headers: flags=-1 Jun 25 08:43:31 sipsrv ser[1929]: check_via_address(200.94.75.6, 200.94.75.6, 0) Jun 25 08:43:31 sipsrv ser[1929]: WARNING:vqm_resize: resize(0) called Jun 25 08:43:31 sipsrv ser[1929]: DEBUG: reply sent out. buf=0x810fe78: SIP/2.0 1..., shmem=0x4057bcd8: SIP/2.0 1 Jun 25 08:43:31 sipsrv ser[1929]: DEBUG: _reply_light: finished Jun 25 08:43:31 sipsrv ser[1929]: DEBUG: mk_proxy: doing DNS lookup... Jun 25 08:43:31 sipsrv ser[1929]: check_via_address(200.94.75.6, 200.94.75.6, 0) Jun 25 08:43:31 sipsrv ser[1929]: DEBUG: add_to_tail_of_timer[4]: 0x4057c160 Jun 25 08:43:31 sipsrv ser[1929]: DEBUG: add_to_tail_of_timer[0]: 0x4057c170 Jun 25 08:43:31 sipsrv ser[1929]: SER: new transaction fwd'ed Jun 25 08:43:31 sipsrv ser[1929]: DEBUG:destroy_avp_list: destroying list (nil) Jun 25 08:43:31 sipsrv ser[1929]: receive_msg: cleaning up Jun 25 08:43:31 sipsrv ser[1931]: SIP Reply (status): Jun 25 08:43:31 sipsrv ser[1931]: version: <SIP/2.0> Jun 25 08:43:31 sipsrv ser[1931]: status: <100> Jun 25 08:43:31 sipsrv ser[1931]: reason: <Trying> Jun 25 08:43:31 sipsrv ser[1931]: parse_headers: flags=1 Jun 25 08:43:31 sipsrv ser[1931]: Found param type 232, <branch> = <z9hG4bKe855.757a9ce3.0>; state=9 Jun 25 08:43:31 sipsrv ser[1931]: parse_via: next_via Jun 25 08:43:31 sipsrv ser[1931]: end of header reached, state=5 Jun 25 08:43:31 sipsrv ser[1931]: parse_headers: Via found, flags=1 Jun 25 08:43:31 sipsrv ser[1931]: parse_headers: this is the first via Jun 25 08:43:31 sipsrv ser[1931]: After parse_msg... Jun 25 08:43:31 sipsrv ser[1931]: forward_reply: found module tm, passing reply to it Jun 25 08:43:31 sipsrv ser[1931]: DEBUG: t_check: msg id=12 global id=2 T start=(nil) Jun 25 08:43:31 sipsrv ser[1931]: parse_headers: flags=17 Jun 25 08:43:31 sipsrv ser[1931]: DEBUG: add_param: tag=2BBC1008-1340 Jun 25 08:43:31 sipsrv ser[1931]: end of header reached, state=29 Jun 25 08:43:31 sipsrv ser[1931]: DEBUG: get_hdr_field: <To> [60]; uri=[sip:018183324166@sipsrv.conecta-te.com] Jun 25 08:43:31 sipsrv ser[1931]: DEBUG: to body [<sip:018183324166@sipsrv.conecta-te.com>] Jun 25 08:43:31 sipsrv ser[1931]: get_hdr_field: cseq <CSeq>: <101> <INVITE> Jun 25 08:43:31 sipsrv ser[1931]: parse_headers: flags=4 Jun 25 08:43:31 sipsrv ser[1931]: DEBUG: t_reply_matching: hash 21902 label 1053402967 branch 0 Jun 25 08:43:31 sipsrv ser[1931]: DEBUG: t_reply_matching: reply matched (T=0x4057c048)! Jun 25 08:43:31 sipsrv ser[1931]: DEBUG: t_check: msg id=12 global id=12 T end=0x4057c048 Jun 25 08:43:31 sipsrv ser[1931]: DEBUG: reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1) Jun 25 08:43:31 sipsrv ser[1931]: ->>>>>>>>> T_code=100, new_code=100 Jun 25 08:43:31 sipsrv ser[1931]: DEBUG: relay_reply: branch=0, save=0, relay=-1 Jun 25 08:43:31 sipsrv ser[1931]: DEBUG: add_to_tail_of_timer[1]: 0x4057c170 Jun 25 08:43:31 sipsrv ser[1931]: DEBUG:destroy_avp_list: destroying list (nil) Jun 25 08:43:31 sipsrv ser[1931]: receive_msg: cleaning up Jun 25 08:43:32 sipsrv ser[1943]: DEBUG: timer routine:4,tl=0x4057c160 next=(nil) Jun 25 08:43:36 sipsrv ser[1930]: SIP Reply (status): Jun 25 08:43:36 sipsrv ser[1930]: version: <SIP/2.0> Jun 25 08:43:36 sipsrv ser[1930]: status: <183> Jun 25 08:43:36 sipsrv ser[1930]: reason: <Session Progress> Jun 25 08:43:36 sipsrv ser[1930]: parse_headers: flags=1 Jun 25 08:43:36 sipsrv ser[1930]: Found param type 232, <branch> = <z9hG4bKe855.757a9ce3.0>; state=9 Jun 25 08:43:36 sipsrv ser[1930]: parse_via: next_via Jun 25 08:43:36 sipsrv ser[1930]: end of header reached, state=5 Jun 25 08:43:36 sipsrv ser[1930]: parse_headers: Via found, flags=1 Jun 25 08:43:36 sipsrv ser[1930]: parse_headers: this is the first via Jun 25 08:43:36 sipsrv ser[1930]: After parse_msg... Jun 25 08:43:36 sipsrv ser[1930]: forward_reply: found module tm, passing reply to it Jun 25 08:43:36 sipsrv ser[1930]: DEBUG: t_check: msg id=6 global id=2 T start=0xffffffff Jun 25 08:43:36 sipsrv ser[1930]: parse_headers: flags=17 Jun 25 08:43:36 sipsrv ser[1930]: DEBUG: add_param: tag=2BBC1008-1340 Jun 25 08:43:36 sipsrv ser[1930]: end of header reached, state=29 Jun 25 08:43:36 sipsrv ser[1930]: DEBUG: get_hdr_field: <To> [60]; uri=[sip:018183324166@sipsrv.conecta-te.com] Jun 25 08:43:36 sipsrv ser[1930]: DEBUG: to body [<sip:018183324166@sipsrv.conecta-te.com>] Jun 25 08:43:36 sipsrv ser[1930]: get_hdr_field: cseq <CSeq>: <101> <INVITE> Jun 25 08:43:36 sipsrv ser[1930]: parse_headers: flags=4 Jun 25 08:43:36 sipsrv ser[1930]: DEBUG: t_reply_matching: hash 21902 label 1053402967 branch 0 Jun 25 08:43:36 sipsrv ser[1930]: DEBUG: t_reply_matching: reply matched (T=0x4057c048)! Jun 25 08:43:36 sipsrv ser[1930]: DEBUG: t_check: msg id=6 global id=6 T end=0x4057c048 Jun 25 08:43:36 sipsrv ser[1930]: DEBUG: reply_received: org. status uas=100, uac[0]=100 local=0 is_invite=1) Jun 25 08:43:36 sipsrv ser[1930]: ->>>>>>>>> T_code=100, new_code=183 Jun 25 08:43:36 sipsrv ser[1930]: DEBUG: relay_reply: branch=0, save=0, relay=0 Jun 25 08:43:36 sipsrv ser[1930]: old size: 752, new size: 698 Jun 25 08:43:36 sipsrv ser[1930]: build_res_from_sip_res: copied size: orig:89, new: 35, rest: 663 msg= SIP/2.0 183 Session Progress^M Via: SIP/2.0/UDP 200.94.75.6:5065^M From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=7a83401cfe54b69o0^M To: <sip:018183324166@sipsrv.conecta-te.com>;tag=2BBC1008-1340^M Date: Sat, 25 Jun 2005 13:43:56 GMT^M Call-ID: 5c2d7341271e8b55e0cb066402224168(a)200.94.75.6^M Server: Cisco-SIPGateway/IOS-12.x^M CSeq: 101 INVITE^M Allow-Events: telephone-event^M Content-Type: application/sdp^M Content-Disposition: session;handling=required^M Content-Length: 212^M ^M v=0^M o=CiscoSystemsSIP-GW-UserAgent 375 1794 IN IP4 200.94.75.1^M s=SIP Call^M c=IN IP4 200.94.27.30^M t=0 0^M m=audio 16960 RTP/AVP 18^M c=IN IP4 200.94.27.30^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=ptime:20^M Jun 25 08:43:36 sipsrv ser[1930]: DEBUG: reply relayed. buf=0x810f708: SIP/2.0 1..., shmem=0x4057bcd8: SIP/2.0 1 Jun 25 08:43:36 sipsrv ser[1930]: DEBUG: add_to_tail_of_timer[1]: 0x4057c170 Jun 25 08:43:36 sipsrv ser[1930]: DEBUG:destroy_avp_list: destroying list (nil) Jun 25 08:43:36 sipsrv ser[1930]: receive_msg: cleaning up Jun 25 08:43:41 sipsrv ser[1935]: SIP Reply (status): Jun 25 08:43:41 sipsrv ser[1935]: version: <SIP/2.0> Jun 25 08:43:41 sipsrv ser[1935]: status: <200> Jun 25 08:43:41 sipsrv ser[1935]: reason: <OK> Jun 25 08:43:41 sipsrv ser[1935]: parse_headers: flags=1 Jun 25 08:43:41 sipsrv ser[1935]: Found param type 232, <branch> = <z9hG4bKe855.757a9ce3.0>; state=9 Jun 25 08:43:41 sipsrv ser[1935]: parse_via: next_via Jun 25 08:43:41 sipsrv ser[1935]: end of header reached, state=5 Jun 25 08:43:41 sipsrv ser[1935]: parse_headers: Via found, flags=1 Jun 25 08:43:41 sipsrv ser[1935]: parse_headers: this is the first via Jun 25 08:43:41 sipsrv ser[1935]: After parse_msg... Jun 25 08:43:41 sipsrv ser[1935]: forward_reply: found module tm, passing reply to it Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: t_check: msg id=11 global id=0 T start=0xffffffff Jun 25 08:43:41 sipsrv ser[1935]: parse_headers: flags=17 Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: add_param: tag=2BBC1008-1340 Jun 25 08:43:41 sipsrv ser[1935]: end of header reached, state=29 Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: get_hdr_field: <To> [60]; uri=[sip:018183324166@sipsrv.conecta-te.com] Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: to body [<sip:018183324166@sipsrv.conecta-te.com>] Jun 25 08:43:41 sipsrv ser[1935]: get_hdr_field: cseq <CSeq>: <101> <INVITE> Jun 25 08:43:41 sipsrv ser[1935]: parse_headers: flags=4 Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: t_reply_matching: hash 21902 label 1053402967 branch 0 Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: t_reply_matching: reply matched (T=0x4057c048)! Jun 25 08:43:41 sipsrv ser[1935]: parse_headers: flags=4 Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: t_check: msg id=11 global id=11 T end=0x4057c048 Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: reply_received: org. status uas=183, uac[0]=183 local=0 is_invite=1) Jun 25 08:43:41 sipsrv ser[1935]: ->>>>>>>>> T_code=183, new_code=200 Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: relay_reply: branch=0, save=0, relay=0 Jun 25 08:43:41 sipsrv ser[1935]: old size: 886, new size: 832 Jun 25 08:43:41 sipsrv ser[1935]: build_res_from_sip_res: copied size: orig:75, new: 21, rest: 811 msg= SIP/2.0 200 OK^M Via: SIP/2.0/UDP 200.94.75.6:5065^M From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=7a83401cfe54b69o0^M To: <sip:018183324166@sipsrv.conecta-te.com>;tag=2BBC1008-1340^M Date: Sat, 25 Jun 2005 13:43:56 GMT^M Call-ID: 5c2d7341271e8b55e0cb066402224168(a)200.94.75.6^M Server: Cisco-SIPGateway/IOS-12.x^M CSeq: 101 INVITE^M Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO^M Allow-Events: telephone-event^M Contact: <sip:018183324166@200.94.75.1:5060>^M Record-Route: <sip:200.94.75.6;ftag=7a83401cfe54b69o0;lr=on>^M Content-Type: application/sdp^M Content-Length: 212^M ^M v=0^M o=CiscoSystemsSIP-GW-UserAgent 375 1794 IN IP4 200.94.75.1^M s=SIP Call^M c=IN IP4 200.94.27.30^M t=0 0^M m=audio 16960 RTP/AVP 18^M c=IN IP4 200.94.27.30^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=ptime:20^M Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: update_totag_set: new totag Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: add_to_tail_of_timer[2]: 0x4057c090 Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: reply relayed. buf=0x8110388: SIP/2.0 2..., shmem=0x4057bcd8: SIP/2.0 2 Jun 25 08:43:41 sipsrv ser[1935]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Jun 25 08:43:41 sipsrv ser[1935]: DEBUG:destroy_avp_list: destroying list (nil) Jun 25 08:43:41 sipsrv ser[1935]: receive_msg: cleaning up Jun 25 08:43:41 sipsrv ser[1924]: SIP Request: Jun 25 08:43:41 sipsrv ser[1924]: method: <ACK> Jun 25 08:43:41 sipsrv ser[1924]: uri: <sip:018183324166@200.94.75.6:5060> Jun 25 08:43:41 sipsrv ser[1924]: version: <SIP/2.0> Jun 25 08:43:41 sipsrv ser[1924]: parse_headers: flags=1 Jun 25 08:43:41 sipsrv ser[1924]: Found param type 232, <branch> = <z9hG4bK-76d2bc25>; state=16 Jun 25 08:43:41 sipsrv ser[1924]: end of header reached, state=5 Jun 25 08:43:41 sipsrv ser[1924]: parse_headers: Via found, flags=1 Jun 25 08:43:41 sipsrv ser[1924]: parse_headers: this is the first via Jun 25 08:43:41 sipsrv ser[1924]: After parse_msg... Jun 25 08:43:41 sipsrv ser[1924]: preparing to run routing scripts... Jun 25 08:43:41 sipsrv ser[1924]: DEBUG : sl_filter_ACK: to late to be a local ACK! Jun 25 08:43:41 sipsrv ser[1924]: parse_headers: flags=128 Jun 25 08:43:41 sipsrv ser[1924]: DEBUG: add_param: tag=2cbe59e2 Jun 25 08:43:41 sipsrv ser[1924]: end of header reached, state=29 Jun 25 08:43:41 sipsrv ser[1924]: DEBUG: get_hdr_field: <To> [55]; uri=[sip:018183324166@sipsrv.conecta-te.com] Jun 25 08:43:41 sipsrv ser[1924]: DEBUG: to body [<sip:018183324166@sipsrv.conecta-te.com>] Jun 25 08:43:41 sipsrv ser[1924]: get_hdr_field: cseq <CSeq>: <101> <ACK> Jun 25 08:43:41 sipsrv ser[1924]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Jun 25 08:43:41 sipsrv ser[1924]: DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 Jun 25 08:43:41 sipsrv ser[1924]: DEBUG: add_param: tag=7a83401cfe54b69o0 Jun 25 08:43:41 sipsrv ser[1924]: end of header reached, state=29 Jun 25 08:43:41 sipsrv ser[1924]: parse_headers: flags=256 Jun 25 08:43:41 sipsrv ser[1924]: is_preloaded: No Jun 25 08:43:41 sipsrv ser[1924]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:41 sipsrv ser[1924]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:41 sipsrv ser[1924]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:41 sipsrv ser[1924]: grep_sock_info - checking if port 5060 matches port 5065 Jun 25 08:43:41 sipsrv ser[1924]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:41 sipsrv ser[1924]: grep_sock_info - checking if port 5060 matches port 5065 Jun 25 08:43:41 sipsrv ser[1924]: parse_headers: flags=256 Jun 25 08:43:41 sipsrv ser[1924]: DEBUG: get_hdr_body : content_length=0 Jun 25 08:43:41 sipsrv ser[1924]: found end of header Jun 25 08:43:41 sipsrv ser[1924]: find_next_route: No next Route HF found Jun 25 08:43:41 sipsrv ser[1924]: after_strict: No next URI found Jun 25 08:43:41 sipsrv ser[1924]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:41 sipsrv ser[1924]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:41 sipsrv ser[1924]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:41 sipsrv ser[1924]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:41 sipsrv ser[1924]: lookup(): '018183324166' Not found in usrloc Jun 25 08:43:41 sipsrv ser[1924]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:41 sipsrv ser[1924]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:41 sipsrv ser[1924]: PASE POR AQUI POR QUE NO SOY CANCEL, INVITE O REGISTER Jun 25 08:43:41 sipsrv ser[1924]: parse_headers: flags=-1 Jun 25 08:43:41 sipsrv ser[1924]: lookup(): '018183324166' Not found in usrloc Jun 25 08:43:41 sipsrv ser[1924]: LOOKUP LOCATION PARA CUALQUIER COSA Jun 25 08:43:41 sipsrv ser[1924]: DEBUG:destroy_avp_list: destroying list (nil) Jun 25 08:43:41 sipsrv ser[1924]: receive_msg: cleaning up Jun 25 08:43:41 sipsrv ser[1943]: DEBUG: timer routine:1,tl=0x4057c170 next=(nil) Jun 25 08:43:41 sipsrv ser[1942]: SIP Reply (status): Jun 25 08:43:41 sipsrv ser[1942]: version: <SIP/2.0> Jun 25 08:43:41 sipsrv ser[1942]: status: <200> Jun 25 08:43:41 sipsrv ser[1942]: reason: <OK> Jun 25 08:43:41 sipsrv ser[1942]: parse_headers: flags=1 Jun 25 08:43:41 sipsrv ser[1942]: Found param type 232, <branch> = <z9hG4bKe855.757a9ce3.0>; state=9 Jun 25 08:43:41 sipsrv ser[1942]: parse_via: next_via Jun 25 08:43:41 sipsrv ser[1942]: end of header reached, state=5 Jun 25 08:43:41 sipsrv ser[1942]: parse_headers: Via found, flags=1 Jun 25 08:43:41 sipsrv ser[1942]: parse_headers: this is the first via Jun 25 08:43:41 sipsrv ser[1942]: After parse_msg... Jun 25 08:43:41 sipsrv ser[1942]: forward_reply: found module tm, passing reply to it Jun 25 08:43:41 sipsrv ser[1942]: DEBUG: t_check: msg id=14 global id=2 T start=0xffffffff Jun 25 08:43:41 sipsrv ser[1942]: parse_headers: flags=17 Jun 25 08:43:41 sipsrv ser[1942]: DEBUG: add_param: tag=2BBC1008-1340 Jun 25 08:43:41 sipsrv ser[1942]: end of header reached, state=29 Jun 25 08:43:41 sipsrv ser[1942]: DEBUG: get_hdr_field: <To> [60]; uri=[sip:018183324166@sipsrv.conecta-te.com] Jun 25 08:43:41 sipsrv ser[1942]: DEBUG: to body [<sip:018183324166@sipsrv.conecta-te.com>] Jun 25 08:43:41 sipsrv ser[1942]: get_hdr_field: cseq <CSeq>: <101> <INVITE> Jun 25 08:43:41 sipsrv ser[1942]: parse_headers: flags=4 Jun 25 08:43:41 sipsrv ser[1942]: DEBUG: t_reply_matching: hash 21902 label 1053402967 branch 0 Jun 25 08:43:41 sipsrv ser[1942]: DEBUG: t_reply_matching: reply matched (T=0x4057c048)! Jun 25 08:43:41 sipsrv ser[1942]: parse_headers: flags=4 Jun 25 08:43:41 sipsrv ser[1942]: DEBUG: t_check: msg id=14 global id=14 T end=0x4057c048 Jun 25 08:43:41 sipsrv ser[1942]: DEBUG: reply_received: org. status uas=200, uac[0]=200 local=0 is_invite=1) Jun 25 08:43:41 sipsrv ser[1942]: ->>>>>>>>> T_code=200, new_code=200 Jun 25 08:43:41 sipsrv ser[1942]: DBG: t_should_relay_response: 200 INV after final sent Jun 25 08:43:41 sipsrv ser[1942]: DEBUG: relay_reply: branch=0, save=0, relay=0 Jun 25 08:43:41 sipsrv ser[1942]: old size: 886, new size: 832 Jun 25 08:43:41 sipsrv ser[1942]: build_res_from_sip_res: copied size: orig:75, new: 21, rest: 811 msg= SIP/2.0 200 OK^M Via: SIP/2.0/UDP 200.94.75.6:5065^M From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=7a83401cfe54b69o0^M To: <sip:018183324166@sipsrv.conecta-te.com>;tag=2BBC1008-1340^M Date: Sat, 25 Jun 2005 13:43:56 GMT^M Call-ID: 5c2d7341271e8b55e0cb066402224168(a)200.94.75.6^M Server: Cisco-SIPGateway/IOS-12.x^M CSeq: 101 INVITE^M Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO^M Allow-Events: telephone-event^M Contact: <sip:018183324166@200.94.75.1:5060>^M Record-Route: <sip:200.94.75.6;ftag=7a83401cfe54b69o0;lr=on>^M Content-Type: application/sdp^M Content-Length: 212^M ^M v=0^M o=CiscoSystemsSIP-GW-UserAgent 375 1794 IN IP4 200.94.75.1^M s=SIP Call^M c=IN IP4 200.94.27.30^M t=0 0^M m=audio 16960 RTP/AVP 18^M c=IN IP4 200.94.27.30^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=ptime:20^M Jun 25 08:43:41 sipsrv ser[1942]: DEBUG: update_totag_set: totag retransmission Jun 25 08:43:41 sipsrv ser[1942]: DEBUG: reply relayed. buf=0x8110218: SIP/2.0 2..., shmem=0x4057bcd8: SIP/2.0 2 Jun 25 08:43:41 sipsrv ser[1942]: DEBUG:destroy_avp_list: destroying list (nil) Jun 25 08:43:41 sipsrv ser[1942]: receive_msg: cleaning up Jun 25 08:43:42 sipsrv ser[1931]: SIP Reply (status): Jun 25 08:43:42 sipsrv ser[1931]: version: <SIP/2.0> Jun 25 08:43:42 sipsrv ser[1931]: status: <200> Jun 25 08:43:42 sipsrv ser[1931]: reason: <OK> Jun 25 08:43:42 sipsrv ser[1931]: parse_headers: flags=1 Jun 25 08:43:42 sipsrv ser[1931]: Found param type 232, <branch> = <z9hG4bKe855.757a9ce3.0>; state=9 Jun 25 08:43:42 sipsrv ser[1931]: parse_via: next_via Jun 25 08:43:42 sipsrv ser[1931]: end of header reached, state=5 Jun 25 08:43:42 sipsrv ser[1931]: parse_headers: Via found, flags=1 Jun 25 08:43:42 sipsrv ser[1931]: parse_headers: this is the first via Jun 25 08:43:42 sipsrv ser[1931]: After parse_msg... Jun 25 08:43:42 sipsrv ser[1931]: forward_reply: found module tm, passing reply to it Jun 25 08:43:42 sipsrv ser[1931]: DEBUG: t_check: msg id=13 global id=12 T start=0xffffffff Jun 25 08:43:42 sipsrv ser[1931]: parse_headers: flags=17 Jun 25 08:43:42 sipsrv ser[1931]: DEBUG: add_param: tag=2BBC1008-1340 Jun 25 08:43:42 sipsrv ser[1931]: end of header reached, state=29 Jun 25 08:43:42 sipsrv ser[1931]: DEBUG: get_hdr_field: <To> [60]; uri=[sip:018183324166@sipsrv.conecta-te.com] Jun 25 08:43:42 sipsrv ser[1931]: DEBUG: to body [<sip:018183324166@sipsrv.conecta-te.com>] Jun 25 08:43:42 sipsrv ser[1931]: get_hdr_field: cseq <CSeq>: <101> <INVITE> Jun 25 08:43:42 sipsrv ser[1931]: parse_headers: flags=4 Jun 25 08:43:42 sipsrv ser[1931]: DEBUG: t_reply_matching: hash 21902 label 1053402967 branch 0 Jun 25 08:43:42 sipsrv ser[1931]: DEBUG: t_reply_matching: reply matched (T=0x4057c048)! Jun 25 08:43:42 sipsrv ser[1931]: parse_headers: flags=4 Jun 25 08:43:42 sipsrv ser[1931]: DEBUG: t_check: msg id=13 global id=13 T end=0x4057c048 Jun 25 08:43:42 sipsrv ser[1931]: DEBUG: reply_received: org. status uas=200, uac[0]=200 local=0 is_invite=1) Jun 25 08:43:42 sipsrv ser[1931]: ->>>>>>>>> T_code=200, new_code=200 Jun 25 08:43:42 sipsrv ser[1931]: DBG: t_should_relay_response: 200 INV after final sent Jun 25 08:43:42 sipsrv ser[1931]: DEBUG: relay_reply: branch=0, save=0, relay=0 Jun 25 08:43:42 sipsrv ser[1931]: old size: 886, new size: 832 Jun 25 08:43:42 sipsrv ser[1931]: build_res_from_sip_res: copied size: orig:75, new: 21, rest: 811 msg= SIP/2.0 200 OK^M Via: SIP/2.0/UDP 200.94.75.6:5065^M From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=7a83401cfe54b69o0^M To: <sip:018183324166@sipsrv.conecta-te.com>;tag=2BBC1008-1340^M Date: Sat, 25 Jun 2005 13:43:56 GMT^M Call-ID: 5c2d7341271e8b55e0cb066402224168(a)200.94.75.6^M Server: Cisco-SIPGateway/IOS-12.x^M CSeq: 101 INVITE^M Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO^M Allow-Events: telephone-event^M Contact: <sip:018183324166@200.94.75.1:5060>^M Record-Route: <sip:200.94.75.6;ftag=7a83401cfe54b69o0;lr=on>^M Content-Type: application/sdp^M Content-Length: 212^M ^M v=0^M o=CiscoSystemsSIP-GW-UserAgent 375 1794 IN IP4 200.94.75.1^M s=SIP Call^M c=IN IP4 200.94.27.30^M t=0 0^M m=audio 16960 RTP/AVP 18^M c=IN IP4 200.94.27.30^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=ptime:20^M Jun 25 08:43:42 sipsrv ser[1931]: DEBUG: update_totag_set: totag retransmission Jun 25 08:43:42 sipsrv ser[1931]: DEBUG: reply relayed. buf=0x81102a8: SIP/2.0 2..., shmem=0x4057bcd8: SIP/2.0 2 Jun 25 08:43:42 sipsrv ser[1931]: DEBUG:destroy_avp_list: destroying list (nil) Jun 25 08:43:42 sipsrv ser[1931]: receive_msg: cleaning up Jun 25 08:43:50 sipsrv ser[1936]: SIP Request: Jun 25 08:43:50 sipsrv ser[1936]: method: <BYE> Jun 25 08:43:50 sipsrv ser[1936]: uri: <sip:018183324166@200.94.75.6:5060> Jun 25 08:43:50 sipsrv ser[1936]: version: <SIP/2.0> Jun 25 08:43:50 sipsrv ser[1936]: parse_headers: flags=1 Jun 25 08:43:50 sipsrv ser[1936]: Found param type 232, <branch> = <z9hG4bK-c79dc8ef>; state=16 Jun 25 08:43:50 sipsrv ser[1936]: end of header reached, state=5 Jun 25 08:43:50 sipsrv ser[1936]: parse_headers: Via found, flags=1 Jun 25 08:43:50 sipsrv ser[1936]: parse_headers: this is the first via Jun 25 08:43:50 sipsrv ser[1936]: After parse_msg... Jun 25 08:43:50 sipsrv ser[1936]: preparing to run routing scripts... Jun 25 08:43:50 sipsrv ser[1936]: parse_headers: flags=128 Jun 25 08:43:50 sipsrv ser[1936]: DEBUG: add_param: tag=2cbe59e2 Jun 25 08:43:50 sipsrv ser[1936]: end of header reached, state=29 Jun 25 08:43:50 sipsrv ser[1936]: DEBUG: get_hdr_field: <To> [55]; uri=[sip:018183324166@sipsrv.conecta-te.com] Jun 25 08:43:50 sipsrv ser[1936]: DEBUG: to body [<sip:018183324166@sipsrv.conecta-te.com>] Jun 25 08:43:50 sipsrv ser[1936]: get_hdr_field: cseq <CSeq>: <102> <BYE> Jun 25 08:43:50 sipsrv ser[1936]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Jun 25 08:43:50 sipsrv ser[1936]: DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 Jun 25 08:43:50 sipsrv ser[1936]: DEBUG: add_param: tag=7a83401cfe54b69o0 Jun 25 08:43:50 sipsrv ser[1936]: end of header reached, state=29 Jun 25 08:43:50 sipsrv ser[1936]: parse_headers: flags=-1 Jun 25 08:43:50 sipsrv ser[1936]: DEBUG: get_hdr_body : content_length=0 Jun 25 08:43:50 sipsrv ser[1936]: found end of header Jun 25 08:43:50 sipsrv ser[1936]: parse_headers: flags=256 Jun 25 08:43:50 sipsrv ser[1936]: is_preloaded: No Jun 25 08:43:50 sipsrv ser[1936]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:50 sipsrv ser[1936]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:50 sipsrv ser[1936]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:50 sipsrv ser[1936]: grep_sock_info - checking if port 5060 matches port 5065 Jun 25 08:43:50 sipsrv ser[1936]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:50 sipsrv ser[1936]: grep_sock_info - checking if port 5060 matches port 5065 Jun 25 08:43:50 sipsrv ser[1936]: parse_headers: flags=256 Jun 25 08:43:50 sipsrv ser[1936]: found end of header Jun 25 08:43:50 sipsrv ser[1936]: find_next_route: No next Route HF found Jun 25 08:43:50 sipsrv ser[1936]: after_strict: No next URI found Jun 25 08:43:50 sipsrv ser[1936]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:50 sipsrv ser[1936]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:50 sipsrv ser[1936]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:50 sipsrv ser[1936]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:50 sipsrv ser[1936]: Hey this request is NOT from B2BUA Jun 25 08:43:50 sipsrv ser[1936]: FORWARDING TO B2BUA Jun 25 08:43:50 sipsrv ser[1936]: parse_headers: flags=-1 Jun 25 08:43:50 sipsrv ser[1936]: parse_headers: flags=64 Jun 25 08:43:50 sipsrv ser[1936]: check_via_address(200.67.33.247, 172.31.253.127, 0) Jun 25 08:43:50 sipsrv ser[1936]: parse_headers: flags=33554432 Jun 25 08:43:50 sipsrv ser[1936]: parse_headers: flags=64 Jun 25 08:43:50 sipsrv ser[1936]: parse_headers: flags=-1 Jun 25 08:43:50 sipsrv ser[1936]: check_via_address(200.67.33.247, 172.31.253.127, 0) Jun 25 08:43:50 sipsrv ser[1936]: Sending: BYE sip:018183324166@200.94.75.6:5060 SIP/2.0^M Record-Route: <sip:200.94.75.6;ftag=7a83401cfe54b69o0;lr=on>^M Via: SIP/2.0/UDP 200.94.75.6;branch=0^M Via: SIP/2.0/UDP 172.31.253.127:5060;rport=60425;received=200.67.33.247;branch=z9hG4bK-c79dc8ef^M From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=7a83401cfe54b69o0^M To: <sip:018183324166@sipsrv.conecta-te.com>;tag=2cbe59e2^M Call-ID: b5fcf1e1-baf4a709(a)172.31.253.127^M CSeq: 102 BYE^M Max-Forwards: 16^M User-Agent: Linksys/PAP2-2.0.12(LS)^M Content-Length: 0^M P-hint: finishing call^M P-hint: PASANDO POR EL B2BUA^M P-hint: APLICO NAT AL B2BUA^M ^M . Jun 25 08:43:50 sipsrv ser[1936]: orig. len=425, new_len=600, proto=1 Jun 25 08:43:50 sipsrv ser[1936]: DEBUG:destroy_avp_list: destroying list (nil) Jun 25 08:43:50 sipsrv ser[1936]: receive_msg: cleaning up Jun 25 08:43:50 sipsrv ser[1924]: SIP Reply (status): Jun 25 08:43:50 sipsrv ser[1924]: version: <SIP/2.0> Jun 25 08:43:50 sipsrv ser[1924]: status: <200> Jun 25 08:43:50 sipsrv ser[1924]: reason: <OK> Jun 25 08:43:50 sipsrv ser[1924]: parse_headers: flags=1 Jun 25 08:43:50 sipsrv ser[1924]: Found param type 232, <branch> = <0>; state=16 Jun 25 08:43:50 sipsrv ser[1924]: end of header reached, state=5 Jun 25 08:43:50 sipsrv ser[1924]: parse_headers: Via found, flags=1 Jun 25 08:43:50 sipsrv ser[1924]: parse_headers: this is the first via Jun 25 08:43:50 sipsrv ser[1924]: After parse_msg... Jun 25 08:43:50 sipsrv ser[1924]: forward_reply: found module tm, passing reply to it Jun 25 08:43:50 sipsrv ser[1924]: DEBUG: t_check: msg id=8 global id=2 T start=0xffffffff Jun 25 08:43:50 sipsrv ser[1924]: parse_headers: flags=17 Jun 25 08:43:50 sipsrv ser[1924]: Found param type 235, <rport> = <60425>; state=6 Jun 25 08:43:50 sipsrv ser[1924]: Found param type 234, <received> = <200.67.33.247>; state=6 Jun 25 08:43:50 sipsrv ser[1924]: Found param type 232, <branch> = <z9hG4bK-c79dc8ef>; state=16 Jun 25 08:43:50 sipsrv ser[1924]: end of header reached, state=5 Jun 25 08:43:50 sipsrv ser[1924]: parse_headers: Via found, flags=17 Jun 25 08:43:50 sipsrv ser[1924]: parse_headers: this is the second via Jun 25 08:43:50 sipsrv ser[1924]: DEBUG: add_param: tag=2cbe59e2 Jun 25 08:43:50 sipsrv ser[1924]: end of header reached, state=29 Jun 25 08:43:50 sipsrv ser[1924]: DEBUG: get_hdr_field: <To> [55]; uri=[sip:018183324166@sipsrv.conecta-te.com] Jun 25 08:43:50 sipsrv ser[1924]: DEBUG: to body [<sip:018183324166@sipsrv.conecta-te.com>] Jun 25 08:43:50 sipsrv ser[1924]: get_hdr_field: cseq <CSeq>: <102> <BYE> Jun 25 08:43:50 sipsrv ser[1924]: DEBUG: t_reply_matching: failure to match a transaction Jun 25 08:43:50 sipsrv ser[1924]: DEBUG: t_check: msg id=8 global id=8 T end=(nil) Jun 25 08:43:50 sipsrv ser[1924]: parse_headers: flags=2 Jun 25 08:43:50 sipsrv ser[1924]: old size: 436, new size: 392 Jun 25 08:43:50 sipsrv ser[1924]: build_res_from_sip_res: copied size: orig:60, new: 16, rest: 376 msg= SIP/2.0 200 OK^M Via: SIP/2.0/UDP 172.31.253.127:5060;rport=60425;received=200.67.33.247;branch=z9hG4bK-c79dc8ef^M To: <sip:018183324166@sipsrv.conecta-te.com>;tag=2cbe59e2^M From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=7a83401cfe54b69o0^M Call-ID: b5fcf1e1-baf4a709(a)172.31.253.127^M CSeq: 102 BYE^M Record-Route: <sip:200.94.75.6;ftag=7a83401cfe54b69o0;lr=on>^M Content-Length: 0^M ^M Jun 25 08:43:50 sipsrv ser[1924]: update_sock_struct_from_via: using 'rport' Jun 25 08:43:50 sipsrv ser[1924]: update_sock_struct_from_via: using 'received' Jun 25 08:43:50 sipsrv ser[1924]: update_sock_struct_from_via: trying SRV lookup Jun 25 08:43:50 sipsrv ser[1924]: reply forwarded to 172.31.253.127:5060 Jun 25 08:43:50 sipsrv ser[1924]: DEBUG:destroy_avp_list: destroying list (nil) Jun 25 08:43:50 sipsrv ser[1924]: receive_msg: cleaning up Jun 25 08:43:50 sipsrv ser[1942]: SIP Request: Jun 25 08:43:50 sipsrv ser[1942]: method: <BYE> Jun 25 08:43:50 sipsrv ser[1942]: uri: <sip:200.94.75.6;lr=on> Jun 25 08:43:50 sipsrv ser[1942]: version: <SIP/2.0> Jun 25 08:43:50 sipsrv ser[1942]: parse_headers: flags=1 Jun 25 08:43:50 sipsrv ser[1942]: end of header reached, state=5 Jun 25 08:43:50 sipsrv ser[1942]: parse_headers: Via found, flags=1 Jun 25 08:43:50 sipsrv ser[1942]: parse_headers: this is the first via Jun 25 08:43:50 sipsrv ser[1942]: After parse_msg... Jun 25 08:43:50 sipsrv ser[1942]: preparing to run routing scripts... Jun 25 08:43:50 sipsrv ser[1942]: parse_headers: flags=128 Jun 25 08:43:50 sipsrv ser[1942]: DEBUG: add_param: tag=2BBC1008-1340 Jun 25 08:43:50 sipsrv ser[1942]: end of header reached, state=29 Jun 25 08:43:50 sipsrv ser[1942]: DEBUG: get_hdr_field: <To> [60]; uri=[sip:018183324166@sipsrv.conecta-te.com] Jun 25 08:43:50 sipsrv ser[1942]: DEBUG: to body [<sip:018183324166@sipsrv.conecta-te.com>] Jun 25 08:43:50 sipsrv ser[1942]: get_hdr_field: cseq <CSeq>: <102> <BYE> Jun 25 08:43:50 sipsrv ser[1942]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Jun 25 08:43:50 sipsrv ser[1942]: DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 Jun 25 08:43:50 sipsrv ser[1942]: DEBUG: add_param: tag=7a83401cfe54b69o0 Jun 25 08:43:50 sipsrv ser[1942]: end of header reached, state=29 Jun 25 08:43:50 sipsrv ser[1942]: parse_headers: flags=-1 Jun 25 08:43:50 sipsrv ser[1942]: DEBUG: get_hdr_body : content_length=0 Jun 25 08:43:50 sipsrv ser[1942]: found end of header Jun 25 08:43:50 sipsrv ser[1942]: parse_headers: flags=256 Jun 25 08:43:50 sipsrv ser[1942]: is_preloaded: No Jun 25 08:43:50 sipsrv ser[1942]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.6] == [200.94.75.6] Jun 25 08:43:50 sipsrv ser[1942]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:50 sipsrv ser[1942]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.1] == [200.94.75.6] Jun 25 08:43:50 sipsrv ser[1942]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:50 sipsrv ser[1942]: grep_sock_info - checking if host==us: 11==11 && [200.94.75.1] == [200.94.75.6] Jun 25 08:43:50 sipsrv ser[1942]: grep_sock_info - checking if port 5060 matches port 5060 Jun 25 08:43:50 sipsrv ser[1942]: check_self: host != me Jun 25 08:43:50 sipsrv ser[1942]: after_strict: Next hop: 'sip:018183324166@200.94.75.1:5060' is strict router Jun 25 08:43:50 sipsrv ser[1942]: TOI: rewrite_uri: Rewriting Request-URI with 'sip:018183324166@200.94.75.1:5060' Jun 25 08:43:50 sipsrv ser[1942]: parse_headers: flags=-1 Jun 25 08:43:50 sipsrv ser[1942]: DEBUG: has_totag: totag found Jun 25 08:43:50 sipsrv ser[1942]: DEBUG: t_newtran: msg id=15 , global msg id=14 , T on entrance=0xffffffff Jun 25 08:43:50 sipsrv ser[1942]: parse_headers: flags=-1 Jun 25 08:43:50 sipsrv ser[1942]: parse_headers: flags=60 Jun 25 08:43:50 sipsrv ser[1942]: t_lookup_request: start searching: hash=21899, isACK=0 Jun 25 08:43:50 sipsrv ser[1942]: DEBUG: proceeding to pre-RFC3261 transaction matching Jun 25 08:43:50 sipsrv ser[1942]: DEBUG: t_lookup_request: no transaction found Jun 25 08:43:50 sipsrv ser[1942]: DBG: trans=0x40579e98, callback type 1, id 0 entered Jun 25 08:43:50 sipsrv ser[1942]: DEBUG: mk_proxy: doing DNS lookup... Jun 25 08:43:50 sipsrv ser[1942]: check_via_address(200.94.75.6, 200.94.75.6, 0) Jun 25 08:43:50 sipsrv ser[1942]: DEBUG: add_to_tail_of_timer[4]: 0x40579fb0 Jun 25 08:43:50 sipsrv ser[1942]: DEBUG: add_to_tail_of_timer[0]: 0x40579fc0 Jun 25 08:43:50 sipsrv ser[1942]: SER: new transaction fwd'ed Jun 25 08:43:50 sipsrv ser[1942]: DEBUG:destroy_avp_list: destroying list (nil) Jun 25 08:43:50 sipsrv ser[1942]: receive_msg: cleaning up Jun 25 08:43:50 sipsrv ser[1931]: SIP Reply (status): Jun 25 08:43:50 sipsrv ser[1931]: version: <SIP/2.0> Jun 25 08:43:50 sipsrv ser[1931]: status: <481> Jun 25 08:43:50 sipsrv ser[1931]: reason: <Call Leg/Transaction Does Not Exist> Jun 25 08:43:50 sipsrv ser[1931]: parse_headers: flags=1 Jun 25 08:43:50 sipsrv ser[1931]: Found param type 232, <branch> = <z9hG4bKb855.107310d2.0>; state=9 Jun 25 08:43:50 sipsrv ser[1931]: parse_via: next_via Jun 25 08:43:50 sipsrv ser[1931]: end of header reached, state=5 Jun 25 08:43:50 sipsrv ser[1931]: parse_headers: Via found, flags=1 Jun 25 08:43:50 sipsrv ser[1931]: parse_headers: this is the first via Jun 25 08:43:50 sipsrv ser[1931]: After parse_msg... Jun 25 08:43:50 sipsrv ser[1931]: forward_reply: found module tm, passing reply to it Jun 25 08:43:50 sipsrv ser[1931]: DEBUG: t_check: msg id=14 global id=13 T start=0xffffffff Jun 25 08:43:50 sipsrv ser[1931]: parse_headers: flags=17 Jun 25 08:43:50 sipsrv ser[1931]: DEBUG: add_param: tag=2BBC1008-1340 Jun 25 08:43:50 sipsrv ser[1931]: end of header reached, state=29 Jun 25 08:43:50 sipsrv ser[1931]: DEBUG: get_hdr_field: <To> [60]; uri=[sip:018183324166@sipsrv.conecta-te.com] Jun 25 08:43:50 sipsrv ser[1931]: DEBUG: to body [<sip:018183324166@sipsrv.conecta-te.com>] Jun 25 08:43:50 sipsrv ser[1931]: get_hdr_field: cseq <CSeq>: <102> <BYE> Jun 25 08:43:50 sipsrv ser[1931]: DEBUG: t_reply_matching: hash 21899 label 755054337 branch 0 Jun 25 08:43:50 sipsrv ser[1931]: DEBUG: t_reply_matching: reply matched (T=0x40579e98)! Jun 25 08:43:50 sipsrv ser[1931]: DEBUG: t_check: msg id=14 global id=14 T end=0x40579e98 Jun 25 08:43:50 sipsrv ser[1931]: DEBUG: reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=0) Jun 25 08:43:50 sipsrv ser[1931]: ->>>>>>>>> T_code=0, new_code=481 Jun 25 08:43:50 sipsrv ser[1931]: DEBUG: relay_reply: branch=0, save=0, relay=0 Jun 25 08:43:50 sipsrv ser[1931]: old size: 372, new size: 318 Jun 25 08:43:50 sipsrv ser[1931]: build_res_from_sip_res: copied size: orig:108, new: 54, rest: 264 msg= SIP/2.0 481 Call Leg/Transaction Does Not Exist^M Via: SIP/2.0/UDP 200.94.75.6:5065^M From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=7a83401cfe54b69o0^M To: <sip:018183324166@sipsrv.conecta-te.com>;tag=2BBC1008-1340^M Call-ID: 5c2d7341271e8b55e0cb066402224168(a)200.94.75.6^M CSeq: 102 BYE^M Content-Length: 0^M ^M Jun 25 08:43:50 sipsrv ser[1931]: WARNING:vqm_resize: resize(0) called Jun 25 08:43:50 sipsrv ser[1931]: DEBUG: add_to_tail_of_timer[2]: 0x40579ee0 Jun 25 08:43:50 sipsrv ser[1931]: DEBUG: reply relayed. buf=0x810f708: SIP/2.0 4..., shmem=0x4057d8d8: SIP/2.0 4 Jun 25 08:43:50 sipsrv ser[1931]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Jun 25 08:43:50 sipsrv ser[1931]: DEBUG:destroy_avp_list: destroying list (nil) Jun 25 08:43:50 sipsrv ser[1931]: receive_msg: cleaning up Jun 25 08:43:51 sipsrv ser[1943]: DEBUG: timer routine:0,tl=0x40579fc0 next=(nil) Jun 25 08:43:51 sipsrv ser[1943]: DEBUG: timer routine:4,tl=0x40579fb0 next=(nil) Jun 25 08:44:00 sipsrv ser[1943]: DEBUG: timer routine:2,tl=0x4057c090 next=(nil) Jun 25 08:44:00 sipsrv ser[1943]: DEBUG: wait_handler : removing 0x4057c048 from table Jun 25 08:44:00 sipsrv ser[1943]: DEBUG: delete transaction 0x4057c048 Jun 25 08:44:00 sipsrv ser[1943]: DEBUG: wait_handler : done Jun 25 08:44:10 sipsrv ser[1943]: DEBUG: timer routine:2,tl=0x40579ee0 next=(nil) Jun 25 08:44:10 sipsrv ser[1943]: DEBUG: wait_handler : removing 0x40579e98 from table Jun 25 08:44:10 sipsrv ser[1943]: DEBUG: delete transaction 0x40579e98 Jun 25 08:44:10 sipsrv ser[1943]: DEBUG: wait_handler : done # # $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") # -- 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) # -- mediaproxy 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")) { 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 #---------------------------------------------------- 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(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=="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; }; #------------------------------------- #RTP Proxy handler #------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); append_hf("P-hint: are you nated?\r\n"); }; route(1); } route[4] { #--------------- #we need to check for the alias #--------------- lookup("aliases"); if (uri!=myself) { route(1); break; }; } 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"); }; }; }

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RE: [Serusers] NAT considerations...
by Lucas Aimaretto 06 Jul '05

06 Jul '05

Sure. I'm using the mystun server. I used the vovida's one, but it does not work as expected. Regards, Lucas > Hi Lucas. > May I ask you which STUN server are you using? > TNX > > Juan > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Lucas > Aimaretto > Sent: Wednesday, July 06, 2005 12:44 PM > To: andres(a)telesip.net; 'Giovanni Balasso' > Cc: serusers(a)lists.iptel.org > Subject: RE: [Serusers] NAT considerations... > > > > Why don't people use stun? > > > Has it some major drawbacks I still haven't found? What are main > > > advantages of rtpproxy-mediaproxy solutions? > > > I'm really curious to know serusers opinions about this issue. > > > > > > thank you all for your two cents ;) > > > > STUN does not work if your NAT is Symmetric. For example all Linux > > NATs or routers with Linux OS like the Linksys ones. > Unless you have > > full control on what type of NAT your customer will deploy, > it will be > > very hard to stick to an all STUN solution. > > > > I have many linuxes around there, and many linksys routers > too, and had no > problems with STUN at all. In fact, when I run my stun-client > application > behind a linux-like OS, this is what I'm told: > > Port restricted NAT detected - VoIP will work with STUN Preserves port > number Does not supports hairpin of media > > Regards, > > Lucas > > -- > No virus found in this outgoing message. > Checked by AVG Anti-Virus. > Version: 7.0.323 / Virus Database: 267.8.9/42 - Release Date: > 06/07/2005 > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > -- > No virus found in this incoming message. > Checked by AVG Anti-Virus. > Version: 7.0.323 / Virus Database: 267.8.9/42 - Release Date: > 06/07/2005 > > -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.9/42 - Release Date: 06/07/2005

1 0

[Serusers] serweb problem (many sessions)
by Sebastian Kühner 06 Jul '05

06 Jul '05

Hello, I have a big problem with serweb. I'm searching for the solution since hours... but I didn't find it!! I installed serweb according to in the installation guide and the HOWTOs that I found in the internet. The problem is that if I want to open the index.php-page (of admin), always appears the Explorer Window "The page cannot be displayed" and when I look at the access_log of apache2 there appears something like that: GET /serweb/index.php?phplib_Session=d69f77e7ccb0d3b6c7d684bcbc368063&phplib_Ses sion=4c8a0c40e09c1bfa01a91ef5832a026a&phplib_Session=5f354e9a405ade8796a54a3 88b0353db&phplib_Session=769c3fa2b669f1dcdd2291c24227ef5d&phplib_Session=4ca 670bdd9a1f4f20441eac5f96c7b8e&phplib_Session=da194f7b1855c60daf515de13b14dab 2&phplib_Session=9bba333123ff119f94f924aeb0bc4710&phplib_Session=f6c9842d7cf 429c632608ad4ee7645c7&phplib_Session=bf06f602d09a1d313297a00f4b293133&phplib _Session=6d798a6dcc84dac416415312debe47d3&phplib_Session=4d1991912b23d4a59ee 1f29f40830317&phplib_Session=31268d0242558745860dbacf41a1be15&phplib_Session =5255e631f1cde5c708cb985db5ebfcaa&phplib_Session=7020b63c06bd983fb5eb054ac4e f627f&phplib_Session=9651b78d2f391f94f23c8b46c603d0a5&phplib_Session=466c61f 55a87cb7ac4661e2816a3ec9d&phplib_Session=9515dbc324cc048f72fb7e1149166f18&ph plib_Session=e0940531f81730793c23a703d772e738&phplib_Session=8e67ddbe3e69eec 9a952b2a59eed48ea&phplib_Session=621fc501c7590d413a79c58278956677&phplib_Ses sion=a4f2af4b0d2e757e0568ee76f8563262&phplib_Session=70960d0cd7a32896b7fd80a 0a4c7f806&phplib_Session=5b78a5a53455042a992def2ab3012430&phplib_Session=0c8 fb10161429bca54375b4dda5aec51&phplib_Session=b1ddb14986db2740a239b5f3494ae08 3&phplib_Session=e281788f6b7533869817cce01a1ebaeb&phplib_Session=d16de271da5 e37b5de9f81268975330b&phplib_Session=45a453bdc8d3969f392c2a4e230c7e0a&phplib _Session=8cd36450b9f6d0592bfe9864bb4e3782&phplib_Session=db9b2e8ca0094fafc2b 923cd377c558c&phplib_Session=dcd92ee19095fba1c4a1475183b27352&phplib_Session =38e1ab2e6f6a65a1bce8d7602db10014&phplib_Session=b074fa09bae847e186be4c22ed8 280a5&phplib_Session=27e2d0c9e4172bed9752fe20a744195c&phplib_Session=d817e3e 3639c105042228e083f9081dd&phplib_Session=f9ac5312dcfcdf3bade2f80a7d6d98bf&ph plib_Session=5f15600e879327263d4d83f489ec6422&phplib_Session=dde588458177dde 8907b932a4ee8a0fe&phplib_Session=d55285a056e6e5a873735a673d80a7cc&phplib_Ses sion=1c51e86702ef7004638d9488cd967304&phplib_Session=bf5b51775d6208711b6904f eeddf5d7b&phplib_Session=a7940c6ed31b41c83b60174ae05b023d HTTP/1.1" 302 - ... that means that serweb wants to make "thousands" of sessions. I didn't find a solution for that in the internet and I hope that someone of you can help me! Thanks! Sebastian

2 1

[Serusers] mySQL PSTN Gateway
by Abdul Lateef 06 Jul '05

06 Jul '05

Hi all, i configured ser.cfg to support mysql db. I wanted to know in wchich table i should put the PSTN Gateways and dailpeers. Because i have more than 5 PSTN Gateways i wanted to use it in ser with some dialpeer. Is it possible to take the gateway ip and port from mysql table according to dialpeer route? Thank You Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail

2 1

[Serusers] Not getting ACK to Mysql when calling SIP to PSTN
by Pablo Bertuzzi 06 Jul '05

06 Jul '05

Hello, i am runing ser 0.10.99-dev8 and 0.8.14 on a different machine. When calling from a sip device (Linksys RT31P2) to PSTN (Cisco 5300), the ACK message is not logged to mysql. Only INVITEs (without 200 status) and BYEs are logged. The other scenarios sip to sip and pstn to sip, log all the messages i need. (INVITE 200, ACK and BYE) I am sure there is something missing in ser.cfg. Any clue on what i am doing wrong?? Kind regards, Pablo. # ------------------ 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/textops.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" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- acc params -- modparam("acc", "db_flag", 1) modparam("acc", "report_cancels", 1) modparam("acc", "report_ack", 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 if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (method=="INVITE" || method=="BYE" || method=="ACK") { setflag(1); }; 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 (src_ip==66.100.100.70 and method=="INVITE") { ## Calls from PSTN forward(uri:host, uri:port); ## break; ## }; if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("66.100.100.73", "subscriber")) { www_challenge("66.100.100.73", "0"); break; }; save("location"); break; }; if (method=="INVITE" || method=="BYE" || method=="CANCEL" || method=="ACK") { setflag(1); acc_db_request("", "acc"); }; if (uri=~"^sip:1786[0-9]*@66.100.100.73") { prefix("5000#"); rewritehostport("66.100.100.70:5060"); forward(uri:host, uri:port); 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(); }; }

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