- sr-users - kamailio.org

Re: [Serusers] lookup_contacts() error after sd_lookup() from the speeddial module- latest cvs ottendorf
by TZieleniewski 23 Feb '07

23 Feb '07

2 5

[Users] ACK transaction
by Cseke Tamas 23 Feb '07

23 Feb '07

Hello, I'd like to relay sip calls from openserto another sip proxy (asterisk), and i have a problem with radius accounting, i 'd like to report acks, but openser doesn't send accounting packet on it. As i understand ack requests must match a transaction flaged to accounting. Config look like this, so i think all transaction is flaged. modparam("acc", "report_ack", 1) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) route{ ... setflag(1); setflag(2); ... } Asterisk retransmit the 200 ok, so i think there is something mistake in the ack. as i read in rfc 3261, ACK must match the dialog, callid, from, to, fromtag, to, tag, and sequence number is relevant. and ack on a 200 ok, is always a new transaction, however i see branch parameter is 0 in the ack, it must start with the magic cookie z9hG4..., isn't it? I dont understand what could be the problem, why 200 ok is retransmitted, and why not openser send acctpacket to radius. Please someone take a look at the sip messages openser->asterisk INVITE sip:0619997400@127.0.0.1:5061 SIP/2.0 Record-Route: <sip:217.116.32.20;lr=on;ftag=LAWYzDCZnyMjZxIp> Via: SIP/2.0/UDP 217.116.32.20;branch=z9hG4bKde6a.a93de1c7.0 Via: SIP/2.0/UDP 192.168.1.150:1720;received=217.116.36.22;branch=z9hG4bKvDD2OX3SfnAE6cR5;rport=1720 Max-Forwards: 69 User-Agent: PA168S V1.53.006 CFG0 From: "222" <sip:222@siptest>;tag=LAWYzDCZnyMjZxIp To: "0619997400" <sip:0619997400@siptest> Call-ID: x3Gd6ZKUB9IeRuPO(a)192.168.1.150 Contact: <sip:222@217.116.36.2:1720> CSeq: 2 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 275 [sdp] asterisk->openser SIP/2.0 200 OK Via: SIP/2.0/UDP 217.116.32.20;branch=z9hG4bKde6a.a93de1c7.0;received=217.116.32.20 Via: SIP/2.0/UDP 192.168.1.150:1720;received=217.116.36.2;branch=z9hG4bKvDD2OX3SfnAE6cR5;rport=1720 Record-Route: <sip:217.116.32.20;lr=on;ftag=LAWYzDCZnyMjZxIp> From: "222" <sip:222@siptest>;tag=LAWYzDCZnyMjZxIp To: "0619997400" <sip:0619997400@siptest>;tag=as37b57c9f Call-ID: x3Gd6ZKUB9IeRuPO(a)192.168.1.150 CSeq: 2 INVITE User-Agent: eWorld Com Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:0619997400@217.116.32.20:5061> Content-Type: application/sdp Content-Length: 206 [sdp] openser->asterisk <-- SIP read from 217.116.32.20:5060: ACK sip:0619997400@127.0.0.1:5061 SIP/2.0 Record-Route: <sip:217.116.32.20;lr=on;ftag=LAWYzDCZnyMjZxIp> Via: SIP/2.0/UDP 217.116.32.20;branch=0 Via: SIP/2.0/UDP 192.168.1.150:1720;received=217.116.36.2;branch=z9hG4bKKysauOpwGdpnsmlS;rport=1720 Route: <sip:217.116.32.20;lr=on;ftag=LAWYzDCZnyMjZxIp> Max-Forwards: 69 User-Agent: PA168S V1.53.006 CFG0 From: "222" <sip:222@siptest>;tag=LAWYzDCZnyMjZxIp To: "0619997400" <sip:0619997400@siptest>;tag=as37b57c9f Call-ID: x3Gd6ZKUB9IeRuPO(a)192.168.1.150 Contact: <sip:222@217.116.36.2:1720> Authorization: Digest username="222", realm="siptest", nonce="45dea111f92e1d1f9bf408a3996413168c304ee9", uri="sip:0619997400@217.116.32.20:5061", response="50419464515ca2771ec4b2bde7147219", algorithm=MD5 CSeq: 2 ACK Content-Length: 0 Thanks any help, Tamas

2 1

[Users] Authenticate INVITEs
by Octavarium 23 Feb '07

23 Feb '07

I have enabled mysql authentication of REGISTER messagess, but i need that only INVITE mesagges from devices properly registered in OpenSER can go through and make calls, and that INVITE mesagges from devices NOT registered are dropped. How can i do this? do you have a sample script? Thanx a lot!

2 1

[Users] Error Handling with Parallel Forking
by Chris Heiser 23 Feb '07

23 Feb '07

Hello gurus, I'm attempting to do some rather crazy parallel forking and have found a few issues I'm hoping to solve. I thought I'd solicit some advice from the experts. Some background: Each device registers with its own unique information. No two devices share the same credentials. Since a random user might want their number sent to multiple phones, and maybe even a cell phone, I've done the following: if(method=="INVITE" && !isflagset(5)) { if(avp_db_load("$ru/username","fork")) { t_on_branch("1"); avp_pushto("$ru/username","$avp(s:fork)/g"); }; }; Then the branch route looks like: branch_route[1] { setflag(5); route(3); } The idea here is the fork avp contains one or more possible destinations, but we don't store a complete URI here. The branch route sends us back through the normal routing logic to lookup local users on the system, and maybe route something to an external PSTN gateway, or another proxy. The problem: I haven't found a way to gracefully handle the event when one of the possible destinations is a device that hasn't registered. Say we allow the user to have a soft client, so we add the username to the list of fork avps. If the device isn't registered, I have 2 options: 1) Use a default route 2) Detect that I should know who this is, and error out Either way, an error is produced, but I don't see a good way of detecting if we should push the 404, or generate a 404 back to the client. The polite method would be: Are there other branches still going? If yes, just exit this branch If no, relay, or generate an error back to the client So my question is how to do this last part. Am I crazy for even thinking about doing this in OpenSER? BTW, I'm using v1.1.1 --Chris

2 1

[Users] uri module question
by Klaus Darilion 23 Feb '07

23 Feb '07

Hi! I have problems understanding the uri module. As far as I understand I can map aliases to subscribers - e.g. my main identity is klaus@mydomain, but I also want to use nickname@cooldomain in From: header with a single SIP account (single subscriber) Thus I can use uri_db module and uri table to check if klaus@mydomain is allowed to use nickname@cooldomain. Thus, what do I enter in the uri table? The definition is: id | username | domain | uri_user | last_modified ----+----------+--------+----------+--------------- In which column do I have to put which values? thanks klaus -- Klaus Darilion nic.at

2 1

[Serusers] Call from Asterisk to SER to a foreign subscriber
by karim basraoui 23 Feb '07

23 Feb '07

Hi, I've set up a small architecture in which an ASterisk proxy is connected to a SER proxy in two different domains. I have sip phones connected to both of them. when an asterisk subscriber "travels" to SER domain it tries to register to SER. I've configured SER to accept foreign subscribers and to REGISTER them to their initial domain by the command "t_replicate". Till now everything works good, the UA is registered and Asterisk is aware of its new location, it displays in peers list the UA and its new IP address and port. The problem is when i'm trying to call the foreign UA from an Asterisk subscriber, it doesn't work. i have a TRYING response but no call establishement. how can i overcome this problem please? -- BASRAOUI Karim www.basraouik.afrikart.net

1 0

[Users] No logging. OpenSER on OpenBSD
by Ben Ferguson - backcountry.com 23 Feb '07

23 Feb '07

I can't get any logging out of openSER for some reason. I've tried using the log_facility setup via /etc/syslog.conf and i still cannot get any output. What am I missing? Here's the top of my openser.cfg file: debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 sip_warning=yes #log_facility=LOG_LOCAL7 I've tried un-commenting that last line to correspond to my /etc/syslog.conf: # don't log messages with LOG_LOCAL7 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL7 in /var/log/ser.log local7.* -/var/log/ser.log What am i doing wrong? Any ideas? Thanks, Ben

3 5

[Serusers] authenticate INVITEs
by Octavarium 23 Feb '07

23 Feb '07

I have enabled mysql authentication of REGISTER messagess, but i need that only INVITE mesagges from devices properly registered in SER can go through, and that INVITE mesagges from devices NOT registered are dropped. How can i do this? do you have a sample script? Thanx a lot!

3 3

[Users] How to reduce load of mediaproxy
by Alex 23 Feb '07

23 Feb '07

Hello, I'm sorry if this question was asked before, I googled and did not found references to it. I'm interested in reducing load of mediaproxy (using mediaproxy 1.8.0, openser-devel). Is it possible to connect callers not behind NAT directly and how configuration should look like? I want to direct traffic by this scheme: Caller Proxy Called NAT <- mediaproxy -> NAT NAT <- mediaproxy -> not NAT not NAT <- mediaproxy -> NAT not NAT <- direct connect -> not NAT I tried configuration something like this: route { if (client_nat_test(3)) { use_media_proxy(); t_on_reply("1"); } } onreply_route[1] { if (client_nat_test("2")) { use_media_proxy(); } } And it works fine then caller is behind NAT and called is not (NAT <-> not NAT) - traffic was directed via mediaproxy. But, then caller is not behind NAT and called is (not NAT <-> NAT), this configuration does not work. Any suggestions? Thanks.

1 0

[Serusers] Polycom IP 601 help needed
by Steve Blair 23 Feb '07

23 Feb '07

I have been given the task of getting a Polycom IP 601 running SIP v2.0.1.0291 to register with our SER proxy and be able to interact with our Asterisk server for voice mail. The Asterisk server currently sends unsolicited NOTIFY messages to turn on/off the message waiting light. Most of the configuration is working however I cannot get the phone to register with SER if I set the voIpProt.server.1.address to the SRV name of our SIP domain. The only way the phone will register is if I set this parameter to the IP address of our SER server which is something we do not want to do. Is there any way to make this phone perform a SRV lookup for the server address? Thanks,Steve

3 5

[Users] tr_inv_timer
by Ibrahim Hamouda 23 Feb '07

23 Feb '07

Is there a way to change the value of the timer from inside the script Ibrahim Hamouda Sphinx Information Technologies Inc. IT Canada International Tel: 403-668-6880 Fax: 403-229-0407 N.A.: 1-877-500-7664 Mobile: 403-714-3336

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[Users] CISCO & OpenSER & more ...
by Daniel-Constantin Mierla 23 Feb '07

23 Feb '07

Hello, OpenSER was chosen to be the SIP proxy of Cisco Service Node for Linksys One. That shows the very good level of stability and flexibility as well as wide set of capabilities in OpenSER: http://www.openser.org/index.php?option=com_content&task=view&id=90&Itemid=9 I would like to express my thanks to all supporting the project by any means (coding, testing, community support ...). Release 1.2.0 is knocking at the door -- it is a huge step forward since 1.1.0 adding lot of new stuff, plus a handful of improvements to existing features, just to remember some of news in 1.2.0: - SIMPLE Presence support - SNMP - PERL API - JAVA SIP Servlet Application Server (http://wesip.eu) - script variables - DNS failover - IP blacklists - XMPP IM gateway - XMLRPC - SIP session timers - secure federation peering - retransmission timer accuracy - automatic error handling So, here I wanted to get: help with testing 1.2.0 is very appreciated these days :-) For those looking forward to meet other OpenSER folks, the next two major events are: - Fosdem, Brussels, Belgium, this weekend - Von Spring, San Jose, USA, in March Cheers, Daniel

5 5

[Serusers] Rewriting To: field
by Stéphane Alnet 23 Feb '07

23 Feb '07

Hello, I am doing interop testing with Level3, and need to be able to rewrite the To: field in requests to match their requirements. I can fake the rewrites inside route[], but although the packets sent out look like they use the new "To:", the header is never parsed again, so the internal structures of SER still hold the former (incorrect) value. When a request hit failure_route[] (for example when a CANCEL is sent when an INVITE times out) the original, incorrect To: header is then sent (and there's no way to modify the content of the CANCEL from within failure_route[]). Is there a standard way to alter the internally-stored value for To: (and From:, since I'm asking)? If not, could someone point me to the right place in the code to start adding this kind of feature? Thanks, Stéphane Alnet -- http://carrierclass.net/ http://www.linkedin.com/in/stephalnet

2 1

RE: [Users] OpenSER Configuration Generator
by Watkins, Bradley 22 Feb '07

22 Feb '07

Hrmmm... I consistently get a Rails error when I get to the config_create stage. This is on both a Linux and Windows machine (a dedicated T1 and a proxied DS3, respectively for connection) using Firefox. Any ideas? - Brad > -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Andreas Granig > Sent: Thursday, February 22, 2007 9:46 AM > To: users(a)openser.org > Subject: [Users] OpenSER Configuration Generator > > All, > > I've finished the first version of sip:wizard, an online > SER/OpenSER configuration generator: http://www.sipwise.com/wizard > > Currently, support for OpenSER 1.1 is available. In the > future, also new OpenSER releases as well as SER will be supported. > > At the moment, we have config features for basic routing, > far-end NAT traversal, ENUM and PSTN in place. This will also > be extended, depending on suggestions from you, the users. > > I hope you'll find it useful. Feedback is of course very much > appreciated. > > Cheers, > Andreas > > > This e-mail is confidential and may well also be legally > privileged. If you have received it in error, you are on > notice of its status. Please notify us immediately by reply > e-mail and then delete this message from your system. Please > do not copy it or use it for any purposes, or disclose its > contents to any other person: to do so could be a breach of > confidence. Thank you for your cooperation. > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

2 1

[Users] OpenSER Configuration Generator
by Andreas Granig 22 Feb '07

22 Feb '07

All, I've finished the first version of sip:wizard, an online SER/OpenSER configuration generator: http://www.sipwise.com/wizard Currently, support for OpenSER 1.1 is available. In the future, also new OpenSER releases as well as SER will be supported. At the moment, we have config features for basic routing, far-end NAT traversal, ENUM and PSTN in place. This will also be extended, depending on suggestions from you, the users. I hope you'll find it useful. Feedback is of course very much appreciated. Cheers, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation.

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[Users] status codes
by Ibrahim Hamouda 22 Feb '07

22 Feb '07

Running openser 1.2 failure_route[1] xlog("L_INFO", "status codes are [$rs] [$rr]\n"); gives "status codes are [<null>] [<null>] any ideas? Ibrahim Hamouda Sphinx Information Technologies Inc. IT Canada International Tel: 403-668-6880 Fax: 403-229-0407 N.A.: 1-877-500-7664 Mobile: 403-714-3336

2 1

[Serusers] Bug in 0.9.6? International characters in Contact header
by Roger Lewau 22 Feb '07

22 Feb '07

Today I discovered that SER does not respond well to international chars in contact header of at least 200 OK. The 200 OK message had the below contact header. The two dots are the swedish chars ä and ö. Contact: Hampus H.ggl.f <sip:330039@217.168.167.xyz:5061>. The error logged from SER was: Feb 22 09:22:27 sip /usr/local/sbin/ser[40800]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Feb 22 09:22:30 sip /usr/local/sbin/ser[40798]: ERROR: mk_proxy: could not resolve hostname: "Hampus" Feb 22 09:22:30 sip /usr/local/sbin/ser[40798]: ERROR: uri2proxy: bad host name in URI <sip:Hampus> Feb 22 09:22:30 sip /usr/local/sbin/ser[40798]: ERROR: t_forward_nonack: failure to add branches The UA is a PAP2-EU device running latest firmware. The text Hampus H.ggl.f was entered in the display name field of PAP2. Is this an error with SER or is it illegal to use international characters in Contact header? Kind regards Roger

2 1

[Users] Active sessions - dialog
by Michel Bensoussan 22 Feb '07

22 Feb '07

Hello I need to write a CAC (Call Admission Control) module for an 802.11 AP (Access Point). The idea is to use a SIP Proxy to monitoring bandwith utilization according to codec, and allow or disallow new sessions, depending on resources. On a first stage I just want to limit the session to a fix number. For that, for each new session request (INVITE), I need to know how many active session there is already. I think using the module dialog but it is experimental. What is exactly its status? Is it usable for now? Is there another way to get information about the number of active sessions? Thanks. Regards, Michel

2 3

[Users] moved temporarily - different port, same host
by Benko 22 Feb '07

22 Feb '07

Hello! I'm having a issue with a 302 moved temporarily from a UA i want to attach to my proxy. The UA replies to a INVITE from the proxy with a 302 MOVED TEMPORARILY and i'm using "get_redirects" from the uac_redirect-module to extract the contact. However, i don't know how to continue from there as the domain part in the redirect-contact is the hostname of the proxy but the port is 5065 and transport=TCP (see debug for details) and if i simply relay it, it would be relayed back to the proxy on the wrong port 5065. I guess i could use the request branch pseudo variable($br) and extract the port from it(or change the domain-part to the actual domain/ip of the UA and relay it), still i don't really know how to do it. This is the sip-trace of the relevant dialog part: 2xx.xxx.xxx.x06 is the openser-host, 6x.xxx.xxx.x93 is the useragent ###################################### #### T 2xx.xxx.xxx.x06:56669 -> 6x.xxx.xxx.x93:5060 [AP] INVITE sip:4989111111@proxydomain.org SIP/2.0. Record-Route: <sip:2xx.xxx.xxx.x06;transport=tcp;r2=on;lr;ftag=as784d1ec2>. Record-Route: <sip:2xx.xxx.xxx.x06;r2=on;lr;ftag=as784d1ec2>. Via: SIP/2.0/TCP 2xx.xxx.xxx.x06;branch=z9hG4bK96db.2c3c0006.0. Via: SIP/2.0/UDP 2xx.xxx.xxx.97:5060;branch=z9hG4bK52fef574;rport=5060. From: "SIPPHONE1" <sip:089222222@otherproxydomain.org>;tag=as784d1ec2. To: <sip:4989111111@proxydomain.org>. Contact:<sip:089222222@21x.xxx.xxx.97>. Call-ID:7fc122590665124f347fb2d956ba4c07@sip.at.telgo.cc. CSeq: 102 INVITE. User-Agent: MyMediaGW1. Max-Forwards: 69. Date: Wed, 21 Feb 2007 13:02:26 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 492. . v=0. o=root 14334 14334 IN IP4 2xx.xxx.xxx.97. s=session. c=IN IP4 2xx.xxx.xxx.97. t=0 0. m=audio 10380 RTP/AVP 0 8 18 4 3 97 111 5 10 7 110 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:111 G726-32/8000. a=rtpmap:5 DVI4/8000. a=rtpmap:10 L16/8000. a=rtpmap:7 LPC/8000. a=rtpmap:110 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - - . # T 6x.xxx.xxx.x93:5060 -> 2xx.xxx.xxx.x06:56669 [AP] SIP/2.0 100 Trying. FROM: "SIPPHONE1"<sip:089222222@otherproxydomain.org>;tag=as784d1ec2. TO: <sip:4989111111@proxydomain.org>. CSEQ: 102 INVITE. CALL-ID: 7fc122590665124f347fb2d956ba4c07(a)otherproxydomain.org. MAX-FORWARDS: 70. VIA: SIP/2.0/TCP 2xx.xxx.xxx.x6;branch=z9hG4bK96db.2c3c0006.0,SIP/2.0/UDP 2xx.xxx.xxx.97:5060;branch=z9hG4bK52fef574;rport=5060. CONTENT-LENGTH: 0. . ## T 6x.xxx.xxx.x93:5060 -> 2xx.xxx.xxx.x06:56669 [AP] SIP/2.0 302 Moved Temporarily. FROM: "SIPPHONE1"<sip:089222222@otherproxydomain.org>;tag=as784d1ec2. TO: <sip:4989111111@proxydomain.org>;tag=b3addacf4f. CSEQ: 102 INVITE. CALL-ID: 7fc122590665124f347fb2d956ba4c07(a)otherproxydomain.org. MAX-FORWARDS: 70. VIA: SIP/2.0/TCP 2xx.xxx.xxx.x06;branch=z9hG4bK96db.2c3c0006.0,SIP/2.0/UDP 213.208.4.97:5060;branch=z9hG4bK52fef574;rport=5060. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. CONTACT: <sip:4989111111@proxydomain.org:5065;transport=TCP>. . ## T 2xx.xxx.xxx.x06:56669 -> 6x.xxx.xxx.x93:5060 [AP] ACK sip:4989111111@proxydomain.org SIP/2.0. Via: SIP/2.0/TCP 213.208.4.106;branch=z9hG4bK96db.2c3c0006.0. From: "SIPPHONE1" <sip:089222222@otherproxydomain.org>;tag=as784d1ec2. Call-ID: 7fc122590665124f347fb2d956ba4c07(a)otherproxydomain.org. TO:<sip:4989111111@proxydomain.org>;tag=b3addacf4f. CSeq: 102 ACK. User-Agent: OpenSer (1.1.0-tls (i386/linux)). Content-Length: 0. . ###################################### Either i'm lacking understanding of the openser functionalities and sip redirect in general and simply don't see how this could be accomplished easier or i guess i need to grab "sip:4989111111@proxydomain.org:5065;transport=TCP" and extract the right port and protocol from it to relay it correctly(don't know how to do that though), i'm not sure. Currently i'm using a simple t_relay("tcp:6x.xxx.xxx.x93:5065"); to get it working, but i'd like to implement it in a more generic way. thx for all answers! best regards christian

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[Serusers] URI Pattern Matching
by Jon Farmer 22 Feb '07

22 Feb '07

Hi I am trying to find a way to detect if the Invite URI starts with a *. How do I match this as ser normally sees * as a wildcard. Regards Jon -- Jon Farmer Telford, Shropshire, UK

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[Users] Install on OpenBSD: cannot start openser
by Ben Ferguson - backcountry.com 22 Feb '07

22 Feb '07

Hello all, I've just installed openSER on OpenBSD and I'm now trying to get openSER to start and am having trouble. I try openserctl start and here's what i get: # openserctl start database engine 'MYSQL' loaded Control engine 'FIFO' loaded Starting OpenSER : \E[37;31mERROR: PID file /var/run/openser.pid does not exist -- OpenSER start failed # After a bunch of reading it seems as if openserctl is now obsolete (???) and that you should use the init script. I can't find reference for anyone running it like this on OpenBSD. I have found openser.init file in /<src directory>/packaging/gentoo and /<src directory>/packaging/rpm, but i am not sure how to use them on OpenBSD (I'm rather new to OpenBSD... I'm used to the rpm style of things.) I've searched and searched but can't find any info... So. anybody out there using openSER on OpenBSD? Or anybody know howto (or whereto find the info to) set up the start script for openSER on OpenBSD? Any help is greatly appreciated! Thanks, Ben

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RE: [Users] call never reaches the failure_route - Please help
by vallimamod abdullah 22 Feb '07

22 Feb '07

On 2/22/07, "Ibrahim Hamouda" <ihamouda(a)itcanint.net> wrote: > > There is already a xlog entry in the beginning of the failure route, and no > it doesn't reach it. > > > Any other ideas? Have you tried to put the 't_on_failure("1")' statement to the main route (instead of the invite route) ? -- Regards, -vma .

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[Serusers] What is the max. length of an IM ?
by Killer 22 Feb '07

22 Feb '07

Hi ! Does someone know the maximum length of an Instant Message I can send with SER ? When I send an IM wich is longer, I get the message by SER : 2(9833) ERROR: read_line: request line too long 2(9833) ERROR: fifo_server: line expected 2(9833) ERROR: fifo_uac_error: body expected How can I solve this problem ? Can I set the maximum length of an IM in any file or something like that ? Best regards ___________________________________________________________ Der frühe Vogel fängt den Wurm. Hier gelangen Sie zum neuen Yahoo! Mail: http://mail.yahoo.de

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[Serusers] fifo tutorial
by Steve Blair 22 Feb '07

22 Feb '07

I lost my link to the only ser fifo tutorial I ever saw.I'd like to write some scripts that talk to the fifo but I do not know what commands are available. Does anyone have a tutorial? Thanks,Steve

2 1

[Users] Adding redirect support to my openser.cfg.
by Shane Burrell 22 Feb '07

22 Feb '07

I want to support 302 type moves. The sip devices we had before supported reinvites but the new linksys units send a 302 message telling me that the user has temporarly moved. Is there a good example of the redirects modules use in the cookbook example? My cfg file is based off that and I would like to add this to a live working system with little interruption or suprises. Thanks, Shane

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[Users] maddr parameter
by Watkins, Bradley 22 Feb '07

22 Feb '07

I see that there was some discussion back in March 2006 (http://www.openser.org/pipermail/users/2006-March/003339.html) regarding this parameter, but nothing since. Daniel mentioned that perhaps it might be put on the to-do list, but I see no mention of it anywhere else. Was this implemented in 1.2? I see that maddr is available using pseudo-var transformations as {uri.maddr} in 1.2, but not mention of any automatic behavior if the parameter exists. I realize that I can work around it by using subst and subst_uri, but I really shouldn't have to. A simple t_relay should send it on its merry way to the value of maddr. My reading of both RFC3261 and 3263 seem to imply that the existence of this parameter in the request URI mandates forwarding the request to this address. I see in the code (for 1.1.1, I haven't looked at 1.2 yet) that the parameter is parsed, but then it appears (correct me if I'm wrong) as though nothing further is done with it. Is there some relatively simple way to modify the code to so that the maddr is used directly? As you can probably tell, I have a gateway with which I have to interoperate which uses this parameter extensively, hence my interest. :) Any advice (even if it's "use subst and subst_uri you whiner" ;) ) is appreciated. Thanks in advance, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

4 3

[Users] call never reaches the failure_route - Please help
by Ibrahim Hamouda 22 Feb '07

22 Feb '07

Hi Guys Could somebody kindly review my openser configuration file below, and tell me why the call never reaches the failure_route. I'm using openser 1.2 Thanks debug=3 fork=no log_stderror=yes listen=10.10.10.10 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "domain.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "permissions.so" loadmodule "mi_fifo.so" loadmodule "xlog.so" modparam("auth_db|permissions|uri_db|usrloc|domain", "db_url", "mysql://openser:ngtgmn@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) #modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "i:30") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://openser:ngtgmn@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return(0); }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return(0); }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("10.10.10.10:5060;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 ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); return(0); }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return(0); } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return(0); }; consume_credentials(); }; if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); return(0); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; return(0); }; if (method=="ACK") { route(1); return(0); } if (method=="CANCEL") { route(1); return(0); } else if (method=="INVITE") { route(3); return(0); } else if (method=="REGISTER") { route(2); return(0); }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); return(0); }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return(0); }; 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"); return(0); }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return(0); }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return(0); } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return(0); }; consume_credentials(); }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); return(0); }; if (uri=~"^sip:011[0-9]*@") { route(4); route(5); return(0); }; if (avp_db_query("SELECT value FROM usr_preferences WHERE username = '$rU' and attribute = 'callfwd'")) { if (is_avp_set("$avp(i:1)")) { setflag(22); avp_pushto("$ru", "$avp(i:1)"); avp_delete("$avp(i:1)"); route(6); return(0); }; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); route(5); return(0); }; sl_send_reply("404", "User Not Found"); return(0); }; if (avp_db_query("SELECT value FROM usr_preferences WHERE username = '$rU' and attribute = 'fwdnoanswer'")) { if (is_avp_set("$avp(i:1)")) { xlog("L_INFO", "Found no answer attribute [$avp(i:1)]\n"); avp_delete("$avp(i:1)"); setflag(27); }; }; if (avp_db_query("SELECT value FROM usr_preferences WHERE username = '$rU' and attribute = 'fwdbusy'")) { if (is_avp_set("$avp(i:1)")) { setflag(26); }; }; t_on_failure("1"); route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("10.10.10.11"); # INSERT YOUR PSTN GATEWAY IP ADDRESS # avp_printf("i:5", "s:inv_timeout"); t_on_failure("1"); route(1); } route[6] { # ------------------------------------------------------------------------ # Call Forwarding Reply Route Handler # # This must be done as a route block because sl_send_reply() cannot be # called from the failure_route block # ------------------------------------------------------------------------ lookup("aliases"); if (!is_uri_host_local()) { if (!isflagset(22)) { append_branch(); }; route(4); route(1); return(0); }; if (uri=~"^sip:011[0-9]*@") { route(4); route(5); return(0); }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); route(1); return(0); }; sl_send_reply("404", "User Not Found"); }; route(4); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } failure_route[1] { xlog("L_INFO", "Reached failure route\n"); if (t_check_status("487")) { return(0); }; if (isflagset(26) && t_check_status("486")) { if (avp_db_query("SELECT value FROM usr_preferences WHERE username = '$rU' and attribute = 'fwdbusy'")) { if (is_avp_set("$avp(i:1)")) { avp_pushto("$ru", "$avp(i:1)"); avp_delete("$avp(i:1)"); resetflag(26); route(6); return(0); }; }; }; if (isflagset(27) && t_check_status("408")) { xlog("L_INFO", "Reached no answer route\n"); if (avp_db_query("SELECT value FROM usr_preferences WHERE username = '$rU' and attribute = 'fwdnoanswer'")) { if (is_avp_set("$avp(i:1)")) { xlog("L_INFO", "Processing answer route\n"); avp_pushto("$ru", "$avp(i:1)"); avp_delete("$avp(i:1)"); resetflag(27); route(6); return(0); }; }; }; end_media_session(); } Ibrahim Hamouda

4 6

[Users] CDR tools feed back
by ram 22 Feb '07

22 Feb '07

Hi Does any one successfully deployed CDR Tools with prepaid and post paid iam looking mainly call disconnect when the balance of the user go negative any sugestions ram

3 3

[Users] Rewriting To: field
by Stéphane Alnet 21 Feb '07

21 Feb '07

Hello, I am doing interop testing with Level3, and need to be able to rewrite the To: field in requests to match their requirements. I can fake the rewrites inside route[], but although the packets sent out look like they use the new "To:", the header is never parsed again, so the internal structures of SER still hold the former (incorrect) value. When a request hit failure_route[] (for example when a CANCEL is sent when an INVITE times out) the original, incorrect To: header is then sent (and there's no way to modify the content of the CANCEL from within failure_route[]). Is there a standard way to alter the internally-stored value for To: (and From:, since I'm asking)? If not, could someone point me to the right place in the code to start adding this kind of feature? Thanks, Stéphane Alnet -- http://carrierclass.net/ http://www.linkedin.com/in/stephalnet

6 9

Re: [Users] Re: openSER on OpenBSD: cannot start openser
by Ben Ferguson - backcountry.com 21 Feb '07

21 Feb '07

This line? modparam("auth_db", "password_column", "password") If so, yes, we did try altering that line to reflect the password for the 'openser' user. Is that what you mean?--If not, can you explain? Thanks for the help. -Ben- ----- Original Message ----- From: "Ovidiu Sas" <sip.nslu(a)gmail.com> To: "Ben Ferguson - backcountry.com" <bferguson(a)backcountry.com> Cc: users(a)openser.org Sent: Tuesday, February 20, 2007 6:36:50 PM (GMT-0700) US/Mountain Subject: Re: [Users] Re: openSER on OpenBSD: cannot start openser Hi Ben, Have you changed the password inside the openser.cfg file? Regards, Ovidiu Sas On 2/20/07, Ben Ferguson - backcountry.com <bferguson(a)backcountry.com> wrote: > After further investigation, it looks as though this is actually caused by changing the default passwords for the openser and openserro users. We tried changing the passwords in openserctlrc to correspond with the mysql user password changes. We granted the rights accordingly in mysql as well. We also found /usr/local/lib/openser/openserctl/openserctl.sqlbase and /usr/local/sbin/openser_mysql.sh that referenced the passwords and we changed them accordingly there and still had no luck. We could not find any other files that seemed to reference the actual passwords. So we then removed the database and tried a fresh install and got the same result. So again the database was removed and this time we altered /usr/local/lib/openser/openserctl/openserctl.sqlbase and /usr/local/sbin/openser_mysql.sh to reflect the passwords we wanted to use, and ran openser_mysql.sh create. This resulted in the same error as before. So, we set all the files back to default with the default passwords, removed all mysql database info and again ran operser_mysql.sh create. Then ran openserctl start again: > > # openserctl start > database engine 'MYSQL' loaded > Control engine 'FIFO' loaded > > Starting OpenSER : \E[37;33mINFO: started (pid: 3947) > # > > Sweet! We then stopped openser and went and changed the passwords again in mysql and in openserctlrc and got the same error as before. It is impossible to change the passwords for openser and openserro to something other than the default! Crazy! We do not want to run with the default passwords... > > Any ideas? > > We are using OpenBSD 4.0 i386, openSER-1.1.1-tls, mysql5.0. > > Thanks, > Ben > > > ----- Original Message ----- > From: "Ben Ferguson - backcountry.com" <bferguson(a)backcountry.com> > To: users(a)openser.org > Sent: Tuesday, February 20, 2007 2:04:15 PM (GMT-0700) US/Mountain > Subject: openSER on OpenBSD: cannot start openser > > > > Hello all, > > I've just installed openSER on OpenBSD and I'm now trying to get openSER to start and am having trouble. I try openserctl start and here's what i get: > > > # openserctl start > database engine 'MYSQL' loaded > Control engine 'FIFO' loaded > > Starting OpenSER : > \E[37;31mERROR: PID file /var/run/openser.pid does not exist -- OpenSER start failed > # > > After a bunch of reading it seems as if openserctl is now obsolete (???) and that you should use the init script. I can't find reference for anyone running it like this on OpenBSD. I have found openser.init file in /<src directory>/packaging/gentoo and /<src directory>/packaging/rpm, but i am not sure how to use them on OpenBSD (I'm rather new to OpenBSD... I'm used to the rpm style of things.) > > I've searched and searched but can't find any info... So. anybody out there using openSER on OpenBSD? Or anybody know howto (or whereto find the info to) set up the start script for openSER on OpenBSD? > > Any help is greatly appreciated! > > Thanks, > Ben > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

2 1

[Users] Problem with voice
by Andrew Nowrot 21 Feb '07

21 Feb '07

Hi I am new to all openser related things. I managed to install it, combine with postgres and do simple accounting, but now I am facing a slightly problem. The situation looks like this: I have two sip phones both on public IP and both registered to openser (on public IP). I am able to ring between this two phones (so I guess that the sip signalization works OK), but I can't hear any voice. For testing purposes I disabled the firewall on openser so I am sure that all ports are "free". My openser.cfg looks like this: #------------------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes # Set to no to enter debugging mode log_stderror=no # (cmd line: -E) Set to yes to enter debugging mode check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/build/buildd/openser-1.1.0 /debian/openser/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/build/buildd/openser-1.1.0 /debian/openser/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/build/buildd/openser-1.1.0 /debian/openser/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/openser/modules/postgres.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/dbtext.so" #loadmodule "/usr/lib/openser/modules/mediaproxy.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", no) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column", "ha1") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) #usrloc|auth_db| modparam("usrloc|auth_db", "db_url", " postgres://openser:openserrw@XXX.XXX.XXX.XXX:5432/openser") modparam("acc", "db_url", "postgres://openser:openserrw@XXX.XXX.XXX.XXX :5432/openser") #modparam("acc", "db_url", "postgres://openser:openserrw@XXX.XXX.XXX.XXX :5432/openser") #modparam("acc", "failed_transaction_flag", 1) #modparam("acc", "report_cancels", 1) modparam("acc", "db_flag", 1) modparam("acc", "log_level", 1) #modparam("acc", "log_fmt", "mfs") #modparam("acc", "log_flag", 1 ) #modparam("acc", "log_fmt", "miocfs") #modparam("acc", "db_flag", 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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") { # mark the INVITEs for accounting setflag(1); # enforce record-routing so the BYEs will come through this server record_route(); }; #if (!method=="REGISTER") # record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request if(method=="BYE") { setflag(1); } t_relay(); return; #append_hf("P-hint: rr-enforced\r\n"); #route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); # 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) } else { if (method=="REGISTER") { #Uncomment this if you want to use digest authentication if (!www_authorize("XXX.XXX.XXX.XXX", "subscriber")) { www_challenge("XXX.XXX.XXX.XXX", "0"); exit; } else { save("location"); exit; }; }; #lookup("aliases"); #if (!uri==myself) { # append_hf("P-hint: outbound alias\r\n"); # route(1); #}; # native SIP destinations are handled using our USRLOC DB if (uri=~"sip:003275566*") { if (!lookup("location")) { sl_send_reply("444", "Not Found"); exit; } append_hf("P-hint: usrloc applied\r\n"); } else { rewritehost("YYY.YYY.YYY.YYY"); #t_relay(); }; }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; #exit; } What could cause this silence? I was wondering maybe mediaproxy or rtpproxy could help, but there is no nat so I can find a reason why I should use them. Could you check my conf file and point me into appropriate direction. Any kind of help would be gladly expected. Best wishes to all. Andrew

2 1

[Users] CDR extraction
by Nick De Cristofaro 21 Feb '07

21 Feb '07

Anyone can recommend a good CDR extraction module that would work with old ser-0.8.14 ? I want it to parse all our calls that are sitting in our old logs. Thanks Nick

2 1

Re: [Users] Cisco 7960 phones
by Scott Yagel 21 Feb '07

21 Feb '07

Ok, I figured out how to unlock the config, using the new menu item called (doh!) "unlock config". One 7960 phone now a functioning SIP phone on the OpenSER. Last thing to figure out is why I cannot access it by browser. Thanks again for the help. Scott Yagel PacketCall, Inc. syagel(a)packetcall.net >-----Original Message----- >From: Stephen Varga [mailto:varga@zbzoom.net] >Sent: Wednesday, February 21, 2007 11:38 AM >To: syagel(a)packetcall.net >Cc: users(a)openser.org >Subject: Re: [Users] Cisco 7960 phones > >Scott, > >Put the attached files into your tftp server that the phones are pulling >their configs and image from. You should also have the following image >files in the same tftp server: > > P003-08-2-00.bin > P0S3-08-2-00.loads > P0S3-08-2-00.zip > P003-08-2-00.sbn > P0S3-08-2-00.sb2 > >I have used these files to successfully upgrade phones. > >If you have SIPxxxxxxxxxxxx.cnf file created for the phones, make sure >the image_version line is commented out. > >The biggest problem I have with this version is using that it reads the >dialplan.xml file but it does not work; I am guessing there is a syntax >change but can not been able to find SIP documentation related to the >8.2 code on Cisco's website. I have been staying with the 7.5 version >of the code because of this problem. > >I found watching which files are being requested by phone a big help in >getting the loads to work properly > >Let me know what happens. > >Steve >

2 1

[Serusers] Accounting with Stateless
by Gustavo Passos Tourinho 21 Feb '07

21 Feb '07

Hi All. Is it possible to use radius accounting with a stateless proxy? For example: if (uri=~"^sip:0001@test.org") { acc_rad_request ("Start"); rewritehostport("127.0.0.1"); forward (uri:host); } When I use this code, the radius request appears without Sip-Translated-Request-URI, why? Changing the above code for this, it works fine (include Sip-Translated-Request-URI): if (uri=~"^sip:0001@test.org") { rewritehostport("127.0.0.1"); t_relay(); } Thanks, Tourinho

1 0

[Users] Re: Problem with voice
by vallimamod abdullah 21 Feb '07

21 Feb '07

Hi Andrew, On 2/20/07, "Andrew Nowrot" <andrew.nowrot(a)gmail.com> wrote: > > Hi > > [...] > I have two sip phones both on public IP and both registered to openser (on > public IP). I am able to ring between this two phones (so I guess that the > sip signalization works OK), but I can't hear any voice. For testing > purposes I disabled the firewall on openser so I am sure that all ports are > "free". My openser.cfg looks like this: > I faced a similar problem some time ago. Here are some tips based on what I have tried to solve this: - Capture the sip messages using ngrep: 'ngrep -qt -d eth0 -W byline port 5060' (of course, replace eth0 with your real iface) - Check the sdp part to see what media IP adress is advertised by each party ('c=IN IP4' line in the INVITE request and OK response.) You may have private IPs here if your sip phones are behind a NAT. - Sniff udp traffic in the port range where the RTP traffic is likely to be (I use '-s0 -X' options on tcpdump to get the payload in hex.) - If you don't see any incoming our outgoing packet then, there ay be a firewall drop issue. Hope this will help. -- Regards, -vma .

1 0

[Users] Audio problem in NAT
by raviprakash sunkara 21 Feb '07

21 Feb '07

Hello Users, I posted so many mail to users but no one reply my issue please help me openSER proxy is mysipdomain.com , and its private_ ip is 192.168.2.60 and SIP server and Proxy is also in Behind UAC's are Behind the NATs --------------------- INVITE --------------- U 59.144.88.7:5060 -> 192.168.2.60:5060 INVITE sip:9002@mysipdomain.com;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKe2a540a8170eb12a. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>. Call-ID: 1685867393(a)192.168.1.2. CSeq: 1 INVITE. Contact: Indian-2 <sip:8002@192.168.1.2:5060;user=phone;transport=udp>. User-Agent: Cisco ATA 188 v3.2.1 atasip (050616A). Expires: 300. Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: 100rel,replaces. Content-Length: 245. Content-Type: application/sdp. v=0. o=8002 14279 14279 IN IP4 192.168.1.2. s=ATA186 Call. c=IN IP4 192.168.1.2. t=0 0. m=audio 16386 RTP/AVP 0 4 8 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:4 G723/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 192.168.2.60:5060 -> 61.17.248.68:3186 INVITE sip:9002@192.168.2.7:5060;user=phone;transport=udp SIP/2.0. Max-Forwards: 10. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: Indian-2 <sip:8002@59.144.88.7:5060;user=phone;transport=udp>. User-Agent: Cisco ATA 188 v3.2.1 atasip (050616A). Expires: 300. Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: 100rel,replaces. Content-Length: 265. Content-Type: application/sdp. v=0. o=8002 14329 14329 IN IP4 192.168.1.2. s=ATA186 Call. c=IN IP4 192.168.1.2. t=0 0. m=audio 16386 RTP/AVP 0 4 8 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:4 G723/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=direction:active. ------------------- RINGING------------ U 61.17.248.68:3186 -> 192.168.2.60:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Require: 100rel. RSeq: 1. Contact: 9002 <sip:9002@192.168.2.7:5060;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Content-Length: 0. . # U 192.168.2.60:5060 -> 59.144.88.7:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Require: 100rel. RSeq: 1. Contact: 9002 <sip:9002@61.17.248.68:3186;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Content-Length: 0. U 61.17.248.68:3186 -> 192.168.2.60:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: 9002 <sip:9002@192.168.2.7:5060;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: replaces. Content-Length: 193. Content-Type: application/sdp. . v=0. o=9002 27865 27865 IN IP4 192.168.2.7. s=ATA186 Call. c=IN IP4 192.168.2.7. t=0 0. m=audio 16386 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 192.168.2.60:5060 -> 59.144.88.7:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: 9002 <sip:9002@61.17.248.68:3186;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: replaces. Content-Length: 193. Content-Type: application/sdp. . v=0. o=9002 27865 27865 IN IP4 192.168.2.7. s=ATA186 Call. c=IN IP4 192.168.2.7. t=0 0. m=audio 16386 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 61.17.248.68:3186 -> 192.168.2.60:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: 9002 <sip:9002@192.168.2.7:5060;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: replaces. Content-Length: 193. Content-Type: application/sdp. . v=0. o=9002 27865 27865 IN IP4 192.168.2.7. s=ATA186 Call. c=IN IP4 192.168.2.7. t=0 0. m=audio 16386 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 192.168.2.60:5060 -> 59.144.88.7:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: 9002 <sip:9002@61.17.248.68:3186;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: replaces. Content-Length: 193. Content-Type: application/sdp. v=0. o=9002 27865 27865 IN IP4 192.168.2.7. s=ATA186 Call. c=IN IP4 192.168.2.7. t=0 0. m=audio 16386 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

2 1

[Users] SDP Parser
by Michel Bensoussan 21 Feb '07

21 Feb '07

Hello For each voice session I need to know the used codec (for bandwith calculation). For that I need to parse the SIP message body. I didn't find in OpenSER such a functionality. Is there a module that doing that? Or maybe someone is working on it? A suggestion for an open source? Thanks. Regards, Michel.

6 15

[Serusers] SER 2.0 TLS module documentation
by Greger V. Teigre 21 Feb '07

21 Feb '07

FYI. This doc is available at http://www.iptel.org/ser/doc/modules/tls It is auto-updated every night from the latest CVS. See http://www.iptel.org/views/tableofmodules for all available module docs. g-) Andrei Pelinescu-Onciul wrote: > andrei 2007/02/21 01:54:45 CET > > SER CVS Repository > > Modified files: > modules/tls README.TLS > Added files: > modules/tls README > modules/tls/doc Makefile certs_howto.xml functions.xml > history.xml params.xml tls.xml > Log: > - added tls module documentation (not yet complete, still missing select, rpc and better tls.cfg description). > > Test case: try to read the generated README and see if it makes sense and you uderstand at least 88,73% from it. Prerequisites: well rested and a blood alcohol level within legal driving limits. WARNING: known to induce strong drowsiness. > > Revision Changes Path > 1.1 +483 -0 sip_router/modules/tls/README (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/README… > 1.2 +2 -0 sip_router/modules/tls/README.TLS > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/README… > 1.1 +29 -0 sip_router/modules/tls/doc/Makefile (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/doc/Ma… > 1.1 +94 -0 sip_router/modules/tls/doc/certs_howto.xml (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/doc/ce… > 1.1 +17 -0 sip_router/modules/tls/doc/functions.xml (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/doc/fu… > 1.1 +22 -0 sip_router/modules/tls/doc/history.xml (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/doc/hi… > 1.1 +395 -0 sip_router/modules/tls/doc/params.xml (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/doc/pa… > 1.1 +103 -0 sip_router/modules/tls/doc/tls.xml (new) > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/tls/doc/tl… > _______________________________________________ > Serdev mailing list > Serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev > > >

1 0

[Users] "noisy feedback"
by Peter Gradwell 21 Feb '07

21 Feb '07

Hi What does "noisy feedback" mean? Quite often we see it in sip traces: SIP/2.0 200 OK To: "Andy Abraham"<sip:2036367@lon-sip-1.gradwell.net>;tag=20c56ba31bbe85b9bc9d0e0 9f876aff7.93cb From: "Andy Abraham"<sip:2036367@lon-sip-1.gradwell.net>;tag=wgu8z0auky Via: SIP/2.0/UDP 192.168.153.100:2085;branch=z9hG4bK-shuf9rjbthup Call-ID: 3c26700aecd1-45fzt2ryqvy3@snom190 CSeq: 2 REGISTER Contact: <sip:2036367@192.168.153.100:2085;line=oru1hcdy>;q=1;expires=25 Server: OpenSer (1.2.0-pre5-notls (i386/linux)) Warning: 392 193.111.201.74:5060 "Noisy feedback tells: pid=6112 req_src_ip=193.111.200.11 req_src_port=5082 in_uri=sip:lon-sip-1.gradwell.net out_uri=sip:lon-sip-1.gradwell.net via_cnt==1" Content-Length: 0 Cheers peter -- peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/ -- engineering & hosting services for email, web and voip -- -- http://www.peter.me.uk/ -- http://www.voip.org.uk/ --

2 1

[Serusers] Radius + Route
by Zeeshan Sajid 21 Feb '07

21 Feb '07

i am trying to configure radius with SER and I have succeeded in doing authentication and accounting for SER using Radius. Now my question is there any way that we can get/load the address of the next gateway and route our call to that. i mean i dont want to use the LCR module instead i would like to use some other LCR code that communicates via radius. thanks in advance regards Zeeshan Awan --------------------------------- Looking for earth-friendly autos? Browse Top Cars by "Green Rating" at Yahoo! Autos' Green Center.

2 1

[Serusers] xlite and ser
by zahra bahar 21 Feb '07

21 Feb '07

Hi all. I am running xlite and ser server (172.16.4.208)on a computer and another xlite phone on the other computer(172.16.4.118). the xlite on 172.16.4.118 could call to the 172.16.4.208 but xlite on 172.16.4.208 can't initiall a call to 172.16.4.118. I had set the proxy and domain address:172.16.4.208. and the xlite on 172.16.4.118 register as: 1001(a)172.16.4.208.is it true?what is the wrong? ____________________________________________________________________________________ Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front

1 0

[Users] adding SDP lines
by jh＠tutpro.com 20 Feb '07

20 Feb '07

i noticed that if i try to add in openser.cfg file a direction=active sdp line using fix_nated_sdp("1") just before i call t_write_unix to send the request to sems, sems does not get that added line. is it so that currently there is no way to add sdp lines in the same openser.cfg that calls t_write_sems and that adding sdp lines needs to be done in earlier openser instance? -- juha

3 2

[Serusers] radius check_from/check_to howto?
by Bruce Bauman 20 Feb '07

20 Feb '07

In the SER issue tracker, someone raised the question of implementing check_from & check_to for radius (http://tracker.iptel.org/browse/SER-35). The issue was closed saying that selects could be used to achieve this. Does anyone want to clue me in as to how you would do this? I can't for the life of me figure this out from the documentation. Thanks. -- Bruce

1 0

RE: [Users] Error while compiling "mysql" module
by Karthik Kamaraj 20 Feb '07

20 Feb '07

Hi All, I need to start openser with Digest Authentication enabled. I have installed libmysqlclient-dev package and modified the openser.cfg file to enable digest authentication feature. When I try to run the openser it gives following error. Looks like it searching for mysql server. Please clarify me how to enable the Digest authentication feature in openser. 0(3004) WARNING: fix_socket_list: could not rev. resolve 192.168.127.151 0(3004) WARNING: fix_socket_list: could not rev. resolve 192.168.127.151 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.127.151 [192.168.127.151]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.127.151 [192.168.127.151]:5060 Aliases: tcp: my.firewall:5060 udp: my.firewall:5060 WARNING: no fork mode and more than one listen address found(will use only the the first one) 0(3004) init_tcp: using epoll_lt as the io watch method (auto detected) 0(0) INFO: statistics manager successfully initialized 0(0) StateLess module - initializing 0(0) TM - initializing... 0(0) Maxfwd module- initializing 0(0) TextOPS - initializing 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing 0(0) new_connection: Can't connect to local MySQL server through socket '/var/l ib/mysql/mysql.sock' (2) 0(0) ERROR:auth_db:auth_fixup: Unable to open database connection 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 134 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration Thanks, Karthik -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Karthik Kamaraj Sent: Monday, February 19, 2007 8:38 PM To: Bogdan-Andrei Iancu Cc: users(a)openser.org Subject: RE: [Users] Error while compiling "mysql" module Hi Bogdan, You mean to say that libmysqlclient-dev package will provide the required *.h files? Thanks, Karthik -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Monday, February 19, 2007 8:34 PM To: Karthik Kamaraj Cc: users(a)openser.org Subject: Re: [Users] Error while compiling "mysql" module Hi Karthik, you need to install the libmysqlclient-dev package containing the development files for mysql client library. regards, bogdan Karthik Kamaraj wrote: > > Hi All, > > I was trying to install Openser from source openser-1.1.1-notls and I > am getting followinr error while compiling "mysql" module. > > Compiling dbase.c > > gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall > -Wno-strict-aliasing -m > > inline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 > -DNAME='"op > > enser"' -DVERSION='"1.1.1-notls"' -DARCH='"i386"' -DOS='"linux"' > -DCOMPILER='"gc > > c 4.1.1"' -D__CPU_i386 -D__OS_linux -D__SMP_yes > -DCFG_DIR='"/usr/local/etc/opens > > er/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST > -DUSE_TCP -DDISA > > BLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DFAST_LOCK > -DADAPTIVE_WAIT > > -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN > -DHAVE_SCH > > ED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H > -DHAVE_TI > > MEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include > -I/usr/loca > > l/include/mysql -I/usr/local/mysql/include/mysql > -I/usr/local/mysql/include -I/u > > sr/include/mysql -c dbase.c -o dbase.o > > dbase.c:29:25: error: mysql/mysql.h: No such file or directory > > dbase.c:30:26: error: mysql/errmsg.h: No such file or directory > > Looks like mysql.h and errmsg.h file is not available. > > I need your comments to resolve the issue. > > Thanks, > > Karthik > > ======================================================================== ==================================================== > > Tech Mahindra, formerly Mahindra-British Telecom. > > Disclaimer: > > This message and the information contained herein is proprietary and > confidential and subject to the Tech Mahindra policy statement, you > may review at http://www.techmahindra.com/Disclaimer.html externally > and http://tim.techmahindra.com/Disclaimer.html internally within Tech > Mahindra. > > ======================================================================== ==================================================== > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > ======================================================================== ==================================================== Tech Mahindra, formerly Mahindra-British Telecom. Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review at <a href="http://www.techmahindra.com/Disclaimer.html">http://www.techmahind ra.com/Disclaimer.html</a> externally and <a href="http://tim.techmahindra.com/Disclaimer.html">http://tim.techmahind ra.com/Disclaimer.html</a> internally within Tech Mahindra. ======================================================================== ==================================================== _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users ============================================================================================================================ Tech Mahindra, formerly Mahindra-British Telecom. Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review at <a href="http://www.techmahindra.com/Disclaimer.html">http://www.techmahindra.com/Disclaimer.html</a> externally and <a href="http://tim.techmahindra.com/Disclaimer.html">http://tim.techmahindra.com/Disclaimer.html</a> internally within Tech Mahindra. ============================================================================================================================

2 1

[Users] fosdem 2007
by Daniel-Constantin Mierla 20 Feb '07

20 Feb '07

Hello, if some of you are joining Fosdem 2007, we can schedule short meeting around. Just send me an email. http://www.fosdem.org/2007/ I will be a lot in Jabber DevRoom, focusing around current SIMPLE-XMPP gateway and interoperability issues. In short and medium terms, events where OpenSER representatives plan to go are listed at (not finished yet): http://www.openser.org/index.php?option=com_content&task=view&id=87&Itemid=… The page should let community members to learn quicker where OpenSER is present, so they can schedule travelings if they want to attend. Also, if any of you is going to speak about some solution or services which use OpenSER, please contact us to list you there. I am sure some people will be happy to learn about and attend the speech/event. If you have old/new slides, and you want to make them public, we can cache and list in openser web page as well. Cheers, Daniel

1 0

[Serusers] cannot allocate memory error
by Su Kerry 20 Feb '07

20 Feb '07

3 4

Re: [Serusers] SER Nokia CANCEL Problem
by Jiri Kuthan 20 Feb '07

20 Feb '07

At 10:07 16/02/2007, samuel wrote: >The 100 Trying is hardcoded and you'll have to look for it in the >code, remove it, and recompile. > >After that, you should send back from your config file the 100 Trying >response in the convenient places to stop retransmissions. > >Samuel. > >2007/2/15, Abdul Qadir <ablqadir(a)yahoo.com>: >>Hi , >> >>I tried to call from one nokia sip (E61 and other models )phone to another >>nokia sip phone. The call works fine. The problem comes only when I call >>from Phone A to Phone B and then immediately cancel the call(from Phone A). >>The Phone A will hangup the call as it sent CANCEL but the SER will ignore >>this CANCEL and still send INVITE to Phone B resulting in a ghost call >>situation. >> >>I tried to capture a log of message and found that Phone A "CANCEL" message >>is received on SER even before any provisional response from Phone B. >>Therefor SER doesnot relay this CANCEL request to Phone B. I even checked >>RFC which clearly says that UAC should not send CANCEL untill it receives >>any provisional response. I talked to Nokia expert and they said the 100 >>Trying message from your server is considered as provisional response, >>therefor behaviour of client is absolutely correct. It appears that there are two things mixed together. The 100 is sent upstream whereas the problem appears downstream. Not sending 100 downstream doesn't solve the race conditions located downstream. The problem is that SER doesn't delay sending the CANCEL, it just gives up on it. The issues is being bug-tracked now, there is no ETA yet. Thank you for reporting! -jiri >>Is there any way I can stop 100 Trying message and still run statefull SER, >>so that I can verify what nokia said. Any ideas suggestions are welcome. >> >>Thanking you all in advance. >> >>Best Regards, >>Abdul Qadir >> >> >> ________________________________ >>Don't be flakey. Get Yahoo! Mail for Mobile and >>always stay connected to friends. >> >> >>_______________________________________________ >>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 -- Jiri Kuthan http://iptel.org/~jiri/

2 3

[Serusers] Error in routing
by GMario 19 Feb '07

19 Feb '07

Hi all, I'm trying to set-up a server ser. I have a basic installation with mysql. My script is a basic script, but when i try to make a phone between two user (or send a text message), the call isn't go and in the standard output i read this error: ERROR: parse_uri: uri too short: <NULL> (4) ERROR: uri2proxy: bad_uri: NULL With google i don't found anything. (in attachment i put my config file) Anyone can help me? Tanks in advance. -- GMario PS: Sorry for my English, this one is my first email write in this language.

2 4

RE: [Users] Error while compiling "mysql" module
by Karthik Kamaraj 19 Feb '07

19 Feb '07

Hi Bogdan, You mean to say that libmysqlclient-dev package will provide the required *.h files? Thanks, Karthik -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Monday, February 19, 2007 8:34 PM To: Karthik Kamaraj Cc: users(a)openser.org Subject: Re: [Users] Error while compiling "mysql" module Hi Karthik, you need to install the libmysqlclient-dev package containing the development files for mysql client library. regards, bogdan Karthik Kamaraj wrote: > > Hi All, > > I was trying to install Openser from source openser-1.1.1-notls and I > am getting followinr error while compiling "mysql" module. > > Compiling dbase.c > > gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall > -Wno-strict-aliasing -m > > inline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 > -DNAME='"op > > enser"' -DVERSION='"1.1.1-notls"' -DARCH='"i386"' -DOS='"linux"' > -DCOMPILER='"gc > > c 4.1.1"' -D__CPU_i386 -D__OS_linux -D__SMP_yes > -DCFG_DIR='"/usr/local/etc/opens > > er/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST > -DUSE_TCP -DDISA > > BLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DFAST_LOCK > -DADAPTIVE_WAIT > > -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN > -DHAVE_SCH > > ED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H > -DHAVE_TI > > MEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include > -I/usr/loca > > l/include/mysql -I/usr/local/mysql/include/mysql > -I/usr/local/mysql/include -I/u > > sr/include/mysql -c dbase.c -o dbase.o > > dbase.c:29:25: error: mysql/mysql.h: No such file or directory > > dbase.c:30:26: error: mysql/errmsg.h: No such file or directory > > Looks like mysql.h and errmsg.h file is not available. > > I need your comments to resolve the issue. > > Thanks, > > Karthik > > ======================================================================== ==================================================== > > Tech Mahindra, formerly Mahindra-British Telecom. > > Disclaimer: > > This message and the information contained herein is proprietary and > confidential and subject to the Tech Mahindra policy statement, you > may review at http://www.techmahindra.com/Disclaimer.html externally > and http://tim.techmahindra.com/Disclaimer.html internally within Tech > Mahindra. > > ======================================================================== ==================================================== > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > ============================================================================================================================ Tech Mahindra, formerly Mahindra-British Telecom. Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review at <a href="http://www.techmahindra.com/Disclaimer.html">http://www.techmahindra.com/Disclaimer.html</a> externally and <a href="http://tim.techmahindra.com/Disclaimer.html">http://tim.techmahindra.com/Disclaimer.html</a> internally within Tech Mahindra. ============================================================================================================================

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[Users] Error while compiling "mysql" module
by Karthik Kamaraj 19 Feb '07

19 Feb '07

Hi All, I was trying to install Openser from source openser-1.1.1-notls and I am getting followinr error while compiling "mysql" module. Compiling dbase.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -Wno-strict-aliasing -m inline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -DNAME='"op enser"' -DVERSION='"1.1.1-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gc c 4.1.1"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/opens er/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISA BLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCH ED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TI MEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -I/usr/loca l/include/mysql -I/usr/local/mysql/include/mysql -I/usr/local/mysql/include -I/u sr/include/mysql -c dbase.c -o dbase.o dbase.c:29:25: error: mysql/mysql.h: No such file or directory dbase.c:30:26: error: mysql/errmsg.h: No such file or directory Looks like mysql.h and errmsg.h file is not available. I need your comments to resolve the issue. Thanks, Karthik ============================================================================================================================ Tech Mahindra, formerly Mahindra-British Telecom. Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review at <a href="http://www.techmahindra.com/Disclaimer.html">http://www.techmahindra.com/Disclaimer.html</a> externally and <a href="http://tim.techmahindra.com/Disclaimer.html">http://tim.techmahindra.com/Disclaimer.html</a> internally within Tech Mahindra. ============================================================================================================================

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[Users] presence messages
by Helge Waastad 19 Feb '07

19 Feb '07

Hi, I'm running kphone and presence on the openser proxy. I see: Feb 16 17:21:55 proxy-02 /usr/sbin/openser[7740]: PRESENCE:query_db_notify: Could not get subs_dialog from database Feb 16 17:21:55 proxy-02 /usr/sbin/openser[7740]: PRESENCE:update_subscribtion:Could not send notify for presence.winfo does anyone know what these messages mean? br hw

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[Users] NO voice, When UAC are in Different Network
by raviprakash sunkara 19 Feb '07

19 Feb '07

Hello Users, I tried a lot , But I can't find the Solution of my Problem Here is My Problem, Please solve my issues When UAC's are in Behind the NAT with Different network and OpenSER is also inside the NAT with Firewalls supported, At this Time is No Voice at , HangUP and Accounting is taking place clearly, But When UAC's are Behind the NAT with Same NetWok and OpenSER is Also inside the NAT, At this time , Voice and Accounting also clear openser,cfg # NAT Registering route[2] { if(nat_uac_test("19") { setflag(6); force_rtport() fix_nated_register(); fix_nated_contact(); }; } # Invite Section with NAT Haple route[3] { if(nat_uac_test("19") { fix_nated_sdp("1") force_rport() fix_nated_contact(); } force_rtp_proxy(); } on_route_reply { if(status==200 || status == 180 ) { forece_rtp_poxy(); } force_nated_contact(); } Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

1 0

Re: [Users] Sip User Agent and IPv6 with OpenSER 1.1.1
by Jiri Kuthan 19 Feb '07

19 Feb '07

At 19:39 15/02/2007, pat.kush(a)verizon.com wrote: >Hello, > >I am setting up an end-to-end IPv6 testbed at Verizon which includes PCs >with IPv6-related applications, Servers, Edge Routers, and Backbone >Routers. > >Currenly I am looking for a SIP client (SIP User Agent) for IPv6 that works >on VISTA, XP, or Linux , and interoperates with OpenSER 1.1.1. > >Does anyone know of or has a SIP client that works on IPv6 networks with >OpenSER 1.1.1? > >Has anyone used OpenSER 1.1.1 for IPv6? Is there any documentation for >this, or bugs ? Well, SER (www.iptel.org/ser/) with v6 has been extensivelly used in PanEuropean v6 trial (http://www.6net.org/), couple of our customers' trials and the most recent sipit (www.sipit.net/). Obviously, signaling in IPv6 was the easy part, sending media between IPv4 and v6 was the harder part. On the client-side, in the recent sipits we have seen an apparently solid implementation from Hellosoft (not sure if GA) and CounterPath (http://www.xten.com/index.php?menu=Support&smenu=compare). At least some version of kphone are v6 enabled but this is I guess very prototypical (http://old.iptel.org/products/kphone/) -jiri -- Jiri Kuthan http://iptel.org/~jiri/

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[Users] Openser Database Migration
by Hansi Lao 19 Feb '07

19 Feb '07

Hello All, I just recently migrated to openser 1.2 from 1.1.x. how can i migrate my database from mysql? do i have to recreate the whole database again? or is there another alternative? Openser halts since its encounters this error which is obviously related to the database backend: 0(0) StateLess module - initializing 0(0) TM - initializing... 0(0) Maxfwd module- initializing 0(0) INFO:ul_init_locks: locks array size 512 0(0) TextOPS - initializing 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing 0(0) ERROR:auth_db:auth_fixup: Invalid table version (use openser_mysql.sh reinstall) 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 122 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration 0(0) INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill Any help wil be greatly appreciated. Thanks!

2 3

[Users] 1.2.0 and retransmissions
by Helge Waastad 19 Feb '07

19 Feb '07

2 7

[Serusers] Prefix from the To user to the from user
by Ibrahim Hamouda 19 Feb '07

19 Feb '07

Hi all This is my first post to the list, so I hope I can get some help. Here is the scenario. I'm building a SER server 0.9.6 with the following characteristics. I'll be supporting different companies with multiple locations each. The phone number will be 7 digits. First 2 digits are the company, the following 2 digits are the branch or location and the last 3 digits are the user number. So, let's say companyA digits are 21, and companyB digits are 22. CompanyA has two branches 10 and 11. Branch 10 has a user 100 (full number in subscriber table is 2110100), and a user 101 (full number in subscriber table is 2110101) Branch 11 has a user 100 (full number in subscriber table is 2111100). CompanyB has a branch 10 and user within the branch 100 (full number in subscriber table is 2210100) I want user when dialing within the same branch to be able to dial the other user number only. User 2110100 can dial 100 to reach User 2110101. But if he wants to dial User 2111100 (Same company different branch) he can dial 11100. And if he wants to dial a user within another company he has to dial the full 7 digits. No to SER config If ser receives a 3 digits number in the $ruri, it should extract the first 4 digits from the From URI and prefix the $ruri/username with them. If ser receives a 5 digits number in the $ruri, it should extract the first 2 digits from the From URI and prefix the $ruri/username with them. My problem is how to extract the first 2 or first 4 digits from the from URI. Any ideas? Thanks in advanced for your help, I have spent 5 days now trying to figure this one out. Ibrahim Hamouda

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[Users] NAT Problem
by raviprakash sunkara 17 Feb '07

17 Feb '07

Hello Users, Good Morning, What are Type of the NAT is used for OpenSER for production , OpenSER is inside the NAT (router/firewall ) 1) I tested with 2 SJ phones and 2 cisco ATA phones, UAC's are in Behind the NAT, UACs are in the Same N/w with one public_address, SJ phone show the NAT type is Symmetric NAT this case is Working Fine, For example : [Behind the NAT1 ] [Behind the NAT2] [Behind the NAT1 ] SJphone --------| firewall]-----------------------> OpenSER-------[firewall]--------------> ATA phone 2) When SJ phones or HardPhones are in Different network behind the NAT of the OpenSER server, here SJ phone shows the Port Restricted core NAT Here media is not is signaling at all , in this Case.... For example : [Behind the NAT1 ] [Behind the NAT2] [Behind the NAT3 ] SJphone ----------[firewall]---------------------> OpenSER--------[firewall ]-------------> ATA phone SIP NAT Traversal support only Symmetric NAT , ? Then How to solve this issue ? Please Help ? -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

2 1

[Serusers] Linksys WRT54GL w/DD-WRT voip (SIPatH) & external registration
by Bruce 17 Feb '07

17 Feb '07

Does anybody has the experience with the Linksys router WRT54GL running DD-WRT v23 SP2 (09/15/06) voip (SIPatH) firmware? I would like to register one external phone at another registrar through the router (using Outbound proxy). I have modified the ser.cfg to process external requests: if (method=="REGISTER") { if (src_ip==192.168.0.0/16 or src_ip==172.16.0.0/12 or src_ip==10.0.0.0/8 or src_ip==X.Y.0.0/16) where X.Y.z.t is the WAN IP address of the SIP Phone on the Internet but unfortunately I have discovered the router is dropping incoming external SIP packets and not passing them to the SIPatH server for processing. Processing of local LAN SIP requests works fine. The router has the LAN address 10.0.0.138. Forwarding the port 5060 to 10.0.0.138 does not help. Any advice would be highly appreciated. Main goal is to allow external phone, registered at another registrar trough the router, to call local LAN SIP phones (and back) trough the SIPatH.

1 0

[Users] IRC channel: #openser
by Daniel-Constantin Mierla 17 Feb '07

17 Feb '07

Hello, this should be interesting for IRC fans. Many openser folks are hanging around #openser channel at irc.freenode.net. It should be a good place to shoot quick questions, answers may get back instantly. Some details: http://www.openser.org/index.php?option=com_content&task=view&id=86&Itemid=… I strongly recommend that important issues/solutions should be summarized to mailing lists. The lists are archived online, improving the knowledge base publicly available -- new comers will have access to it and therefore the learning process is speed up. The channel is in the process of getting officially reserved and assigned to OpenSER project. Many thanks to Adam Linford and all others spending time to get this done. Have fun there, Daniel

2 1

[Users] Sip User Agent and IPv6 with OpenSER 1.1.1
by pat.kush＠verizon.com 16 Feb '07

16 Feb '07

Hello, I am setting up an end-to-end IPv6 testbed at Verizon which includes PCs with IPv6-related applications, Servers, Edge Routers, and Backbone Routers. Currenly I am looking for a SIP client (SIP User Agent) for IPv6 that works on VISTA, XP, or Linux , and interoperates with OpenSER 1.1.1. Does anyone know of or has a SIP client that works on IPv6 networks with OpenSER 1.1.1? Has anyone used OpenSER 1.1.1 for IPv6? Is there any documentation for this, or bugs ? Thanks. Pat Kush Pat Kush Distinguished Member Technical Staff Verizon Technologies 40 Sylvan Road Waltham, MA 02451

3 2

[Serusers] uac module in 9.6
by Stephen Helfen 16 Feb '07

16 Feb '07

I am under the impression that 9.6 can be run including the uac module in order to implement the uac_replace_from function. Have I been misinformed?Thanks,Steve _________________________________________________________________ Get the most out of tax shelters and other breaks. http://articles.moneycentral.msn.com/Taxes/TaxShelters/TaxShelters.aspx?ici…

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[Users] Re: [Devel] Sip User Agent and IPv6 with OpenSER 1.1.1
by pat.kush＠verizon.com 16 Feb '07

16 Feb '07

Thank you. We were in contact with Counterpath about 2 months ago. At that time Eyebeam did not support IPv6. We have used Eyebeam in the IPv4 portion of our IPv6 NGNLAB. Pat "Klaus Darilion" <klaus.mailinglists To: Patricia J. Kush/EMPL/MA/Verizon@VZNotes @pernau.at> cc: users(a)openser.org, team(a)openser.org, devel(a)openser.org Subject: Re: [Devel] Sip User Agent and IPv6 with OpenSER 1.1.1 02/16/2007 06:57 AM pat.kush(a)verizon.com wrote: > Hello, > > I am setting up an end-to-end IPv6 testbed at Verizon which includes PCs > with IPv6-related applications, Servers, Edge Routers, and Backbone > Routers. > > Currenly I am looking for a SIP client (SIP User Agent) for IPv6 that works > on VISTA, XP, or Linux , and interoperates with OpenSER 1.1.1. Counterpath says that eyebeam supports IPv6. regards klaus > > Does anyone know of or has a SIP client that works on IPv6 networks with > OpenSER 1.1.1? > > Has anyone used OpenSER 1.1.1 for IPv6? Is there any documentation for > this, or bugs ? > > Thanks. > > Pat Kush > > Pat Kush > Distinguished Member Technical Staff > Verizon Technologies > 40 Sylvan Road > Waltham, MA 02451 > > > > > _______________________________________________ > Devel mailing list > Devel(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/devel -- Klaus Darilion nic.at

1 0

[Users] Openser and NATed users
by Victor Gamov 16 Feb '07

16 Feb '07

Hi guys! I try to use OpenSER with NATed-users and now I have some problem. No I'll try to explain it. I have the following network scheme: mySIP-phone_behind_NAT --> NAT --> OpenSER --> OtherProxy --> other_SIP_phone mySIP-phone_behind_NAT have some knowledges about external NAT_ip so it successfully registered on OpenSER using nathelper module and location table have new record looks like: user@mydomain Contact: user@NAT_ip:5060 Received: NAT_ip:NAT_port Then mySIP-phone_behind_NAT send new INVITE with Contact=user@NAT_ip:5060 OpenSER fixup this Contact using fix_nated_contact() and send INVITE to OtherProxy with Contact=user@NAT_ip:NAT_port;nat=yes Then 200OK and ACK successfully traversed via OpenSER and dialog esteblished. Then other_SIP_phone sends BYE request using R-URI learned from previous INVITE Contact-header=ser@NAT_ip:NAT_port;nat=yes OpenSER rewrite this R-URI to user@NAT_ip:NAT_port and send it using t_relay(). The packet goes to NAT_ip:NAT_port but mySIP-phone_behind_NAT sends "404 Not found" because it does not know anything about user@NAT_ip:NAT_port -- it expected user@NAT_ip:5060 as sended in first INVITE. So, my question is: is this my misconfiguration or incorrect logic understanding ? I have some ideas how to fix this problem, for eaxample: when I got INVITE from mySIP-phone_behind_NAT I must rewrite Contact to user@mydomain (assuming that requests to mydomain goes to OpenSER always). Then if OpenSER got BYE from other_SIP_phone it will lookup() and restore Contact and Received info. But I have problem again: If I rewrite R-URI to Contact and t_relay() it -- packet will be discarded by NAT because it does not anything about port 5060. If I rewrite R-URI using Received info I will get problem discribed early. Feature request "Allow AVP in t_relay()" (published in Tracker) will solve this problem Any suggestion? -- CU, Victor Gamov

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[Serusers] XLOG buffer overflow help
by Nuno Ribeiro 16 Feb '07

16 Feb '07

Hello! I'm trying to add an character in the final length of an avp. The avp that I need to modify contain all SDP message. The script code that I'm using is the following: xlset_attr("$sdpnew","%$sdp;") XLOG:xl_print_log: buffer overflow -- increase the buffer size... 2(8682) Pos: 0, Add: 664, Len: 256, Buf: 2(8682) xl_printstr: Error while formating result 2(8682) ERROR: avp.c:807: xlset_attr:Error while expanding xl_format How can I overcome this problem? If there is another way to do this, please tell me Best regards, -- Nuno Ribeiro

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Re: Re: [Serusers] lookup_contacts() error after sd_lookup() from the speeddial mo dule- latest cvs ottendorf
by tzieleniewski 16 Feb '07

16 Feb '07

> Hi, > > seems related to the lookup session we had on beginning of this week. > sd_lookup puts the uid into TO / USER, which is not working with the > head version (lookup_contacts need it in TO / URI) > > You can use this workaround: > if (uri =~ "sip:[0-9]{2}@.*") { > sd_lookup("speed_dial"); > $tr.uid=$tu.uid; > } Yes it seemed to me too but I just wasn't sure:) I checked it if it would help and unfortunately nothing changed. I checked it more deeply to get some more view on this and found strange results. I did the following: dump_attrs(); xlog("L_INFO", " route[INBOUND]: (uri =~ sip:[0-9]{2}@.*) <%rm> <%ru> \n"); if (uri =~ "sip:[0-9]{2}@.*") { xlog("L_INFO", " route[INBOUND]: sd_lookup(speed_dial) \n"); sd_lookup("speed_dial"); }; dump_attrs(); and result showed that there is no uid put into TO / USER so generally putting $tr.uid=$tu.uid does nothing, it is visible below in the output: Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:540: class=GLOBAL Feb 16 09:14:10 rd ser[16874]: AVP["lang"]="en" Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 16 09:14:10 rd ser[16874]: AVP["did"]="voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: AVP["digest_realm"]="voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:560: track=TO class=DOMAIN Feb 16 09:14:10 rd ser[16874]: AVP["did"]="voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: AVP["digest_realm"]="voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:570: track=FROM class=USER Feb 16 09:14:10 rd ser[16874]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: AVP["email"]="tzl(a)touk.pl" Feb 16 09:14:10 rd ser[16874]: AVP["lang"]="en" Feb 16 09:14:10 rd ser[16874]: AVP["tyk"]="tyk" Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:580: track=TO class=USER Feb 16 09:14:10 rd ser[16874]: INFO: No AVP present Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:590: track=FROM class=URI Feb 16 09:14:10 rd ser[16874]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:600: track=TO class=URI Feb 16 09:14:10 rd ser[16874]: INFO: No AVP present Feb 16 09:14:10 rd ser[16874]: route[INBOUND]: (uri =~ sip:[0-9]{2}@.*) <INVITE> <sip:01@voip.touk.pl> Feb 16 09:14:10 rd ser[16874]: route[INBOUND]: sd_lookup(speed_dial) Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:540: class=GLOBAL Feb 16 09:14:10 rd ser[16874]: AVP["lang"]="en" Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 16 09:14:10 rd ser[16874]: AVP["did"]="voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: AVP["digest_realm"]="voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:560: track=TO class=DOMAIN Feb 16 09:14:10 rd ser[16874]: AVP["did"]="voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: AVP["digest_realm"]="voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:570: track=FROM class=USER Feb 16 09:14:10 rd ser[16874]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: AVP["email"]="tzl(a)touk.pl" Feb 16 09:14:10 rd ser[16874]: AVP["lang"]="en" Feb 16 09:14:10 rd ser[16874]: AVP["tyk"]="tyk" Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:580: track=TO class=USER Feb 16 09:14:10 rd ser[16874]: INFO: No AVP present Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:590: track=FROM class=URI Feb 16 09:14:10 rd ser[16874]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 16 09:14:10 rd ser[16874]: INFO: avp.c:600: track=TO class=URI Feb 16 09:14:10 rd ser[16874]: INFO: No AVP present And there are still many questions about the lookup_user() function. When I call a subscriber directly not by speeddial number and I invoke it this way: lookup_user("$t.uid", "@ruri"); then the invocation of he lookup_contacts("location") function returns positive result. Any other form of lookup_user() causes the failure of the lookup_contacts() function. I tried to do $tr.uid = $t.uid as well as $tu.uid = $t.uid after lookup_user() but it always didn't cause any change and those avp values where always empty. Please share any ideas:) Cheers -tomasz > > it will set the AVP in the URI track and should be then reachable. > > Michal > > > On Thu, 2007-02-15 at 14:20 +0100, tzieleniewski wrote: > > Hi! > > > > I have encountered the following situation after I make the translation from the speeddial number into the proper uri the user cannot be found by the lookup_contacts() function. Calling a user directly works without any problem: > > > > My logic is the following: > > > > dump_attrs(); > > if (uri =~ "sip:[0-9]{2}@.*") { > > sd_lookup("speed_dial"); > > } > > dump_attrs(); > > if (lookup_user("$t.uid", "@ruri")) { > > dump_attrs(); > > load_attrs("$tu", "@ruri"); > > load_attrs("$tr", "@ruri"); > > dump_attrs(); > > if (lookup_contacts("location")) { > > append_hf("P-hint: usrloc applied\r\n"); > > route(FORWARD); > > drop; > > } else { > > route(CALL_FORWARD_VOICEMAIL); > > drop; > > } > > } > > > > Here is the debug output: > > > > Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: <INVITE> <sip:01@voip.touk.pl> > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:540: class=GLOBAL > > Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:550: track=FROM class=DOMAIN > > Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:560: track=TO class=DOMAIN > > Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:570: track=FROM class=USER > > Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["email"]="tzl(a)touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" > > Feb 15 14:15:23 rd ser[14469]: AVP["tyk"]="tyk" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:580: track=TO class=USER > > Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:590: track=FROM class=URI > > Feb 15 14:15:23 rd ser[14469]: AVP["authuid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:600: track=TO class=URI > > Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present > > Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: (uri =~ sip:[0-9]{2}@.*) <INVITE> <sip:01@voip.touk.pl> > > Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: sd_lookup(speed_dial) > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:540: class=GLOBAL > > Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:550: track=FROM class=DOMAIN > > Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:560: track=TO class=DOMAIN > > Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:570: track=FROM class=USER > > Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["email"]="tzl(a)touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" > > Feb 15 14:15:23 rd ser[14469]: AVP["tyk"]="tyk" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:580: track=TO class=USER > > Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:590: track=FROM class=URI > > Feb 15 14:15:23 rd ser[14469]: AVP["authuid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:600: track=TO class=URI > > Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present > > Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: (lookup_user($t.uid, @ruri)) <INVITE> <sip:hellboy@voip.touk.pl> > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:540: class=GLOBAL > > Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:550: track=FROM class=DOMAIN > > Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:560: track=TO class=DOMAIN > > Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:570: track=FROM class=USER > > Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["email"]="tzl(a)touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" > > Feb 15 14:15:23 rd ser[14469]: AVP["tyk"]="tyk" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:580: track=TO class=USER > > Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:590: track=FROM class=URI > > Feb 15 14:15:23 rd ser[14469]: AVP["authuid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:600: track=TO class=URI > > Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: load_attrs($tu,@ruri) <INVITE> <sip:hellboy@voip.touk.pl> > > Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: load_attrs($tr,@ruri) <INVITE> <sip:hellboy@voip.touk.pl> > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:540: class=GLOBAL > > Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:550: track=FROM class=DOMAIN > > Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:560: track=TO class=DOMAIN > > Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:570: track=FROM class=USER > > Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["email"]="tzl(a)touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" > > Feb 15 14:15:23 rd ser[14469]: AVP["tyk"]="tyk" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:580: track=TO class=USER > > Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:590: track=FROM class=URI > > Feb 15 14:15:23 rd ser[14469]: AVP["authuid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:600: track=TO class=URI > > Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: AVP["forward_blindd"]="sip:evn@voip.touk.pl" > > Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: (lookup_contacts(location)) > > Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: route(CALL_FORWARD_VOICEMAIL) <INVITE> <sip:hellboy@voip.touk.pl> > > > > Bests > > -tomasz > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] Forwarding and mediaproxy
by Shaun Hofer 16 Feb '07

16 Feb '07

Hi, I wish to forward busy and no answer calls to Asterisk, and have the RTP stream go through mediaproxy. At the moment, some calls use mediaproxy and some don't. If the call is not using mediaproxy to get it too. I have noticed that I can't call use_media_proxy() from failed route. I don't want to use mediaproxy for every call between UA's, if not needed. Any suggestions on how I might be able to get calls to use mediaproxy if forwarded ? current fail route: failure_route[1] { if (t_check_status("487")) { break; }; if (isflagset(26) && t_check_status("486")) { avp_delete("s:fwdbusy"); resetflag(26); revert_uri(); rewritehostport("202.168.41.218:5060"); append_branch(); t_relay_to_udp("202.168.41.218", "5060"); break; }; if (isflagset(27) && t_check_status("408")) { avp_delete("s:fwdnoanswer"); resetflag(27); revert_uri(); rewritehostport("202.168.41.218:5060"); append_branch(); t_relay_to_udp("202.168.41.218", "5060"); break; }; end_media_session(); } Thanks -- Shaun Hofer

4 11

[Serusers] SER doesn't create TCP registrations
by Andrew T Gin 16 Feb '07

16 Feb '07

Hi I'm having some trouble getting SER working. I am using example Hello World example ser.cfg (in Ser-GettingStarted.pdf). My client can REGISTER using UDP successfully, but when I to REGISTER with TCP, it fails. I performed a wireshark trace, and the problem is when the client sends a SYN, SER responds with an RST, so the TCP 3 way handshake doesn't finish. The output of SER shows: Listening on udp: 132.181.9.24 [132.181.9.24]:5060 tcp: 132.181.9.24 [132.181.9.24]:5060 Aliases: tcp: cosc4101.cosc.canterbury.ac.nz:5060 udp: cosc4101.cosc.canterbury.ac.nz:5060 What have I done wrong? Thanks Andrew

2 1

[Serusers] Can SER do this?
by David Good 16 Feb '07

16 Feb '07

I have a problem that I *think* SER can help with, but looking through the docs and samples, I don't see anything quite like what I need to do. Here's my situation: I have an Asterisk box setup internally with a few PSTN trunks. We have another box in a colo that we're using to communicate to a SIP trunk from bandwidth.com. Currently we have Asterisk running there, too. What we want to do is: * Have all outbound calls from Asterisk go to the SIP trunk at the colo. We have this working now using Asterisk at the colo. * Have all incoming calls from the SIP trunk (it has several DID numbers) be routed to the internal Asterisk box just like the existing PSTN trunks. I haven't been able to get this part working using Asterisk (which is why I'm looking at SER now) There is NAT involved between the colo server and bandwidth.com. Is this possible?

2 1

[Users] Open SER for SIP NNI
by konstantinos nikou 16 Feb '07

16 Feb '07

Dear all, I am trying to use the Open SER for SIP NNI. I would like all calls with the prefix 077 to be routed to another SIP server and add a prefix (+49). I am using the following code: if (uri=~"^sip:077[0-9]*@*") { strip(1); prefix("+49"); rewritehost("172.17.2.35"); # forward( 172.17.2.35, 5060 ); route(1); } Normally, i believe that when i run this code and i am sending an INVITE from my user to a number e.g. 0771234567, the SER server should send this invite to the IP 172.17.2.35. But i don't see this. The SER responsds directly with a 404 NOT FOUND message. Do you have any idea of how i could solve the problem? I am using Suse Linux and attached you can find my configuration file. Thank you all, K.N. ____________________________________________________________________________________ Looking for earth-friendly autos? Browse Top Cars by "Green Rating" at Yahoo! Autos' Green Center. http://autos.yahoo.com/green_center/

2 1

[Users] Openser- parser error for openser.cfg
by K Biju 16 Feb '07

16 Feb '07

Hi All, I have installed openser version 1.1.1-notls(i386/linux). I am trying to configure the openser to route all calls to a gateway. For this I have added the following rules in openser.cfg route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; } record_route(); if (src_ip==1.1.1.1) { forward_tcp(4.4.4.4, 5060); } else { forward_udp(1.1.1.1, 5060); } } Now when I bring up the server I get the following error [root@localhost openser-1.1.1-notls]# /usr/local/sbin/openser -n 1 -ddddd -E -l tcp:47.100.105.195:5060 -l udp:47.100.105.195:5060 0(18381) read 3133282016 from /dev/urandom 0(18381) seeding PRNG with 9917133 0(18381) test random number 521145638 0(18381) parse error (27,46-50): parse error 0(18381) parse error (27,50-51): bad arguments 0(18381) parse error (30,28-29): parse error 0(18381) parse error (30,41-42): bad arguments ERROR: bad config file (4 errors) What is wrong with the syntax or am I missing any modules ? Thanks in advance -Biju ____________________________________________________________________________________ Get your own web address. Have a HUGE year through Yahoo! Small Business. http://smallbusiness.yahoo.com/domains/?p=BESTDEAL

2 1

[Users] NAT Types
by raviprakash sunkara 16 Feb '07

16 Feb '07

Hello Users, Good Morning, What are Type of the NAT is used for OpenSER for production , OpenSER is inside the NAT (router/firewall ) 1) I tested with 2 SJ phones and 2 cisco ATA phones, UAC's are in Behind the NAT, UACs are in the Same N/w with one public_address, SJ phone show the NAT type is Symmetric NAT this case is Working Fine, For example : [Behind the NAT1 ] [Behind the NAT2] [Behind the NAT1 ] SJphone -------------------------------> OpenSER---------------------> ATA phone 2) When SJ phones or HardPhones are in Different network behind the NAT of the OpenSER server, here SJ phone shows the Port Restricted core NAT Here media is not is signaling at all , in this Case.... For example : [Behind the NAT1 ] [Behind the NAT2] [Behind the NAT3 ] SJphone -------------------------------> OpenSER---------------------> ATA phone SIP NAT Traversal support only Symmetric NAT , ? Then How to solve this issue ? Please Help ? -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

1 0

[Users] NAT Types Isssues
by raviprakash sunkara 16 Feb '07

16 Feb '07

Hello Users, Good Morning, What are Type of the NAT is used for OpenSER for production , OpenSER is inside the NAT (router/firewall ) 1) I tested with 2 SJ phones and 2 cisco ATA phones, UAC's are in Behind the NAT, UACs are in the Same N/w with one public_address, SJ phone show the NAT type is Symmetric NAT in Display... this case is Working Fine, 2) When SJ phones or HardPhones are in Different network behind the NAT of the OpenSER server, here SJ phone shows the Port Restricted core NAT Here media is not is signaling at all , in this Case.... SIP NAT Traversal support only Symmetric NAT , ? Then How to solve this issue ? Please Help ? -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 Client and Parent company :- www.august-networks.com

1 0

Re: [Users] help-me in 800
by Daniel-Constantin Mierla 16 Feb '07

16 Feb '07

Hello, the PIN is sent usually via IVR codes. OpenSER is just a signaling proxy, it doen't interpret media stream. To get auth via PIN, you need a media server in the midle, like Asterisk. Cheers, Daniel On 02/14/07 01:30, Jobson Andrade wrote: > > Hi All, > > > > I,m use openser 1.0.1 > > > > We imagined the following setting: > > > > 1- It Receive the connections of a pstn that sends all of the > requisition for my one openser > > > > 2- it authenticate the coming calls of that IP > > > > 3- It Receive the PIN, authenticate and direct the call for the routes > > > > The question is like do that verification of the PIN´s and send the > calls after that verification > > > > It plan is: > > > > It receives itself 0800 123 123 of an ONLY IP, through the > src_ip==xxx.xxx.xxx.xxx sends for a route > > > > I obtained to remove with the strip I number him 0800 123 123 and with > the prefix (""); send with the rewritehostport for a route > > > > I need that be attended, authenticated the PIN and receive of the side > "A" I number of fate and sent for the route of smaller cost > > > > How to made this???? > > > > Please help-me!! > > > > Thanks in advanced!! > > > > > > > > Jobson Andrade > > > > Projetos & Desenvolvimento > Obelisk - The Asterisk & VoIP Experts > > > > phone/fax: (11) 2164-4808 ext. 115 > cell Phone: (11) 8175-9916 / 8271-0480 > email: jandrade(a)obelisknet.com.br <mailto:jandrade@obelisknet.com.br> > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

2 2

[Users] Re: Message dispatching to local scripts - problem in modifying the RURI
by Dan-Cristian Bogos 15 Feb '07

15 Feb '07

Bogdan, thxs a loot. It is more clear now when I know how the message is handled. Cheers, Dan

1 0

[Users] error editing openser.cfg
by Javier Allende Astigarraga 15 Feb '07

15 Feb '07

i follow this instrucctions and when i execute open ser appear one failure, this is the openser.cfg # # $Id: openser.cfg,v 1.6.2.1 2006/07/17 15:51:03 klaus_darilion Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); #exit; # break; # }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; 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(); }; exit; } thanks for all -- ===================================================== Legolas_Bilbao[ID2006][GKR] Dios creo un equipo Perfecto a los demas los lleno de extranjeros http://www.forosindicedonkey.com http://usuarios.lycos.es/ligaforo/ =====================================================

2 2

Re: [Users] Send Call using authentication
by Bogdan-Andrei Iancu 15 Feb '07

15 Feb '07

Hi Jobson, does your ITSP require digest authentication? if so, you better take a look at uac module. See: http://www.openser.org/docs/modules/1.2.x/uac.html regards, bogdan Jobson Andrade wrote: > > Hi All, > > I need of send the calls for my ITSP using verification. > > Ex.: > > User = user_itsp > > Passwd = pass_itsp > > SIP Server = itsp.com.br > > I my configuration I made of this way, more not work. > > ……… > > ##Obelisk > > if (uri=~"^sip:00[1-9][1-9][0-9]*@") { > > route(3); > > exit; > > } > > ……… > > route[3]{ > > #sending route to DDI > > strip(2); > > rewritehostport("itsp.com.br:5060"); > > rewriteuser("user_itsp"); > > rewriteuserpass("pass_itsp"); > > route(1); > > exit; > > } > > ……… > > How to made this in openser? > > I use the openser version 1.0.1 > > Thanks in advanced, > > Jobson Andrade > > Projetos & Desenvolvimento > Obelisk - The Asterisk & VoIP Experts > > phone/fax: (11) 2164-4808 ext. 115 > cell Phone: (11) 8175-9916 / 8271-0480 > email: jandrade(a)obelisknet.com.br <mailto:jandrade@obelisknet.com.br> > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

2 1

Re: [Users] Message dispatching to local scripts - problem in modifying the RURI
by Bogdan-Andrei Iancu 15 Feb '07

15 Feb '07

Dan, the env. vars. are also available in 1.1.x : http://www.openser.org/docs/modules/1.1.x/exec.html#AEN33 the buffer that is sent to the script by exec_msg() is the original received buffer, so it does not contain any of your script changes. regards, bogdan Dan-Cristian Bogos wrote: > Hi Bogdan, > > Till 1.2.x will be stable, I have to use 1.1.0 in my production, > therefore I cannot use the new features from 1.2.x (even if I am keen > of doing this). > > About the external script, I am passing the full message to the > external script, and I have considered that this is after I have done > the ruri modification through avps. I have tried also by using strip() > and prefix() functions from the core, but got the same result: my > changes were not applied when echoing the message to the external > script. > > As I have wrote also to users mailing list, here is my routing flow: > > > if (avp_check("$avp(s:message_dst)","re/sip:00.*/g")) { > avp_subst("$avp(s:message_dst)/g", "/sip:00(.*)/sip:+\1/"); > xlog("I have changed the destination from sip:00 into sip:+ and avp > is $avp(s:message_dst)"); > }; > > if (!avp_pushto("$ru","$avp(s:message_dst)")) { > sl_send_reply("403", "Cannot set destination from AVP"); > exit; > }; > > exec_msg("/scripts/echo_message"); > t_newtran(); > t_reply("200", "OK"); > t_release(); > exit; > > Thxs in advance, > Dan > > On 2/15/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: >> Hi Dan, >> >> how are you accessing the ruri from the external script? maybe you >> should use the env vars set by openser - see: >> http://www.openser.org/docs/modules/1.2.x/exec.html#AEN33 >> >> guess SIP_RURI is what you need. >> >> regards, >> bogdan >> >> Dan-Cristian Bogos wrote: >> > Guys, >> > >> > I've got a problem with MESSAGE processing. >> > I am trying to push an avp to replace $ru before dispatching the >> > message to a local script. I can log the modification of the ru >> > successful with xlog, but, as soon as I will send the message to the >> > local script, it looks like the changes are dropped and to the script >> > will be presented the original message. Any ideea why? Are the changes >> > not applied for messages not forwarded out, and dispached locally? >> > >> > In my routing I have: >> > >> > if (avp_check("$avp(s:message_dst)","re/sip:00.*/g")) { >> > avp_subst("$avp(s:message_dst)/g", "/sip:00(.*)/sip:+\1/"); >> > xlog("I have changed the destination from sip:00 into sip:+ and >> avp >> > is $avp(s:message_dst)"); >> > }; >> > >> > if (!avp_pushto("$ru","$avp(s:message_dst)")) { >> > sl_send_reply("403", "Cannot set destination from AVP"); >> > exit; >> > }; >> > >> > exec_msg("echo_message"); >> > t_newtran(); >> > t_reply("200", "OK"); >> > t_release(); >> > exit; >> > >> > I appreciate any kind of advice. >> > >> > Thxs, >> > Dan >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >> >

1 0

[Users] Aliases and attempts from UA's to register their aliases
by Andreas Sikkema 15 Feb '07

15 Feb '07

Hi, I'm working on extending my OpenSER configuration to allow a user account to have one or more aliases. User A is reachable by calling A, but also using alias B or C. I have this working for calls to the UA, but I'm having problems with registration attempts from A and calls from A to other UA's or PSTN where it wants to send B or C as outgoing CLI. For some reason, all UAs I've seen so far need to register their aliases to be able to send an alias as outgoing CLI. My OpenSER config uses the following to check authenticate the user (from an old SER example: if (!www_authorize("domain", "subscriber")) { www_challenge("domain", "0"); exit; } if (!check_to()) { log("LOG: To Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use To=id in REGISTERs"); exit; }; When a UA tries to register an alias the www_authorize() succeeds, because the UA sends the correct authentication data for A, but uses the alias B or C in the To and From headers so check_to() fails with the error that B or C is being spoofed. I (OpenSER) know that B and C are aliases of A, but how do I make this registration logic accept that and send a 200 OK message back? (I also don't want to save the registered alias in the location table, but that part I know how to do, I think). The calls using outgoing CLI of the alias have I think the same problem where the From header has an alias where check_from() expects the data for A. I think the solution for this would be something comparable to solving the registration problem, correct? Thanks! -- Andreas Sikkema

3 4

[Users] Message dispatching to local scripts - problem in modifying the RURI
by Dan-Cristian Bogos 15 Feb '07

15 Feb '07

Guys, I've got a problem with MESSAGE processing. I am trying to push an avp to replace $ru before dispatching the message to a local script. I can log the modification of the ru successful with xlog, but, as soon as I will send the message to the local script, it looks like the changes are dropped and to the script will be presented the original message. Any ideea why? Are the changes not applied for messages not forwarded out, and dispached locally? In my routing I have: if (avp_check("$avp(s:message_dst)","re/sip:00.*/g")) { avp_subst("$avp(s:message_dst)/g", "/sip:00(.*)/sip:+\1/"); xlog("I have changed the destination from sip:00 into sip:+ and avp is $avp(s:message_dst)"); }; if (!avp_pushto("$ru","$avp(s:message_dst)")) { sl_send_reply("403", "Cannot set destination from AVP"); exit; }; exec_msg("echo_message"); t_newtran(); t_reply("200", "OK"); t_release(); exit; I appreciate any kind of advice. Thxs, Dan

2 1

[Users] Transactional Stateful Proxy
by Michel Bensoussan 15 Feb '07

15 Feb '07

Hello In the OpenSER feature list it says that it support transactional stateful SIP Proxy processing. Does that mean Call Stateful? I mean if a session is established and a UA sends a re-INVITATION (for example to add video to voice) will the OpenSER Proxy will be informed of this new transaction ? And eventually can it disallow this new transaction (if there is no enough resources like bandwith)? Regards, Thanks. -- Michel Bensoussan Extricom Ltd. Tel-Aviv University, The Wolfson Computer and Software Engineering Building 30 Haim Levanon Street, P.O.B. 39040 Tel-Aviv, 61390 Israel Tel: +972 3 745 1000 ext 1065 Fax: +972 3 745 1001

2 1

[Users] radius authentication
by Stefano Capitanio 15 Feb '07

15 Feb '07

Ciao! I'm trying to authenticate using Radius. I've installed radiusclient-ng 0.5.2 i've configured radiusclient.conf with the IP address of the radius server. I've configured SER with: ... modparam("auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") ... if (method=="REGISTER") { if (!radius_www_authorize("example.it")) { www_challenge("example.it", "0"); exit; } save("location"); } but it doesn't work: there seem to be a problem in the client because there is no IP Packet that goes out directed to the radius server... in the log of SER i have no errors... thank you for any help, Stefano

2 1

[Users] Method of tracking dead accounts?
by Nick De Cristofaro 15 Feb '07

15 Feb '07

We are looking to find a way to audit the accounts on our SER that haven't been used in a long time so we can decide if they are dead and we can remove them, hence re-use their PSTN numbers. I'm not sure if there's any module built in that could do this? Any ideas? There's 4000 accounts approximetely. Thanks Nick

2 1

[Serusers] user aliases in ottendorf??
by tzieleniewski 15 Feb '07

15 Feb '07

Hi !! Is there still some mechanism for user aliases in ottendorf?? I ask because when I add a user through serweb it adds two records to the uri table one for a canonical name and the other for numeric address. What has to be done (what function called in ser.cfg) so that whenever user will be called on that numerical name there will be a possibility to forward the call to him, like it was previously done in ser 0.9 with lookup("aliases")? Bests -Tomasz

2 2

Re: [Users] codec translation
by vallimamod abdullah 15 Feb '07

15 Feb '07

Hello, On Thu, 15 Feb 2007 11:19:37 +0100, Atle Samuelsen <clona(a)cyberhouse.no> wrote: > Hi Guys, > > Just qurious.. How intrested are you guys in a very simple transcoder, > wich is based on rtpproxy ? > I would be really interested in a g729 to g711 transcoder as I have a pstn gateway that doesn't support g729. I am currently using mediaproxy but I would consider to either move to rtpproxy with transcoder functionality or contribute to implement this feature in mediaproxy. -- Regards, -vma .

3 2

[Users] not install mysql.so
by Javier Allende Astigarraga 15 Feb '07

15 Feb '07

i try to install openser from tar.gz and i do it make all include_modules="mysql" make prefix=/usr/local install include_modules="mysql" But dont appears mysql.so what the matter?? thanks por all -- ===================================================== Legolas_Bilbao[ID2006][GKR] Dios creo un equipo Perfecto a los demas los lleno de extranjeros http://www.forosindicedonkey.com http://usuarios.lycos.es/ligaforo/ =====================================================

3 2

[Users] codec translation
by tusker keg 15 Feb '07

15 Feb '07

Howdy I have a situation I hope you guys will help me out with I am receiving call from a VOIP peer (SIP Call) and the peer can only send them as G711. I need to redirect to call to another voip peer over the wan and due to bandwidth considerations I need to translate the codec to g729. Any ideas on how to do this, Sample config file will be help Regards ./Tusker

6 11

[Serusers] lookup_contacts() error after sd_lookup() from the speeddial module- latest cvs ottendorf
by tzieleniewski 15 Feb '07

15 Feb '07

Hi! I have encountered the following situation after I make the translation from the speeddial number into the proper uri the user cannot be found by the lookup_contacts() function. Calling a user directly works without any problem: My logic is the following: dump_attrs(); if (uri =~ "sip:[0-9]{2}@.*") { sd_lookup("speed_dial"); } dump_attrs(); if (lookup_user("$t.uid", "@ruri")) { dump_attrs(); load_attrs("$tu", "@ruri"); load_attrs("$tr", "@ruri"); dump_attrs(); if (lookup_contacts("location")) { append_hf("P-hint: usrloc applied\r\n"); route(FORWARD); drop; } else { route(CALL_FORWARD_VOICEMAIL); drop; } } Here is the debug output: Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: <INVITE> <sip:01@voip.touk.pl> Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:540: class=GLOBAL Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:560: track=TO class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:570: track=FROM class=USER Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["email"]="tzl(a)touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: AVP["tyk"]="tyk" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:580: track=TO class=USER Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:590: track=FROM class=URI Feb 15 14:15:23 rd ser[14469]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:600: track=TO class=URI Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: (uri =~ sip:[0-9]{2}@.*) <INVITE> <sip:01@voip.touk.pl> Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: sd_lookup(speed_dial) Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:540: class=GLOBAL Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:560: track=TO class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:570: track=FROM class=USER Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["email"]="tzl(a)touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: AVP["tyk"]="tyk" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:580: track=TO class=USER Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:590: track=FROM class=URI Feb 15 14:15:23 rd ser[14469]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:600: track=TO class=URI Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: (lookup_user($t.uid, @ruri)) <INVITE> <sip:hellboy@voip.touk.pl> Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:540: class=GLOBAL Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:560: track=TO class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:570: track=FROM class=USER Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["email"]="tzl(a)touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: AVP["tyk"]="tyk" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:580: track=TO class=USER Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:590: track=FROM class=URI Feb 15 14:15:23 rd ser[14469]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:600: track=TO class=URI Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: load_attrs($tu,@ruri) <INVITE> <sip:hellboy@voip.touk.pl> Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: load_attrs($tr,@ruri) <INVITE> <sip:hellboy@voip.touk.pl> Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:540: class=GLOBAL Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:560: track=TO class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:570: track=FROM class=USER Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["email"]="tzl(a)touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: AVP["tyk"]="tyk" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:580: track=TO class=USER Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:590: track=FROM class=URI Feb 15 14:15:23 rd ser[14469]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:600: track=TO class=URI Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["forward_blindd"]="sip:evn@voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: (lookup_contacts(location)) Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: route(CALL_FORWARD_VOICEMAIL) <INVITE> <sip:hellboy@voip.touk.pl> Bests -tomasz

2 1

Re: [Serusers] Inoltra: Problems with Rtpproxy traffic balancing in SER 0.10.x
by Ladislav Andel 15 Feb '07

15 Feb '07

what is written in your modparam rtpproxy line? how do you start your rtpproxy? (what parameters do you use) There might be some changes in the code as well since the document was issued few months ago and the development of SER is going on. I'm trying to reproduce it with the latest CVS version.. Lada Angelo Taddeo wrote: > I've done correctly anything you wrote in your mail. > > I've found my mistake with $node: in database i put it with flag 32 > instead of 33. > > Anyway one problem remains: it isn't enough inserting rtpproxy IP > address in modparam (in ser.cfg). > If I dont't give this IP to function/ force_rtp_proxy/ in ser.cfg it > doesn't work, that is it doesn't write correctly rtpproxy IP in > connection information (SDP) during SDP rewriting . > I've looked at code. SER expects rtpproxy IP address in reply from > rtpproxy. But rtpproxy in its message to SER doesn't write its IP!!!!! > > So in my ser.cfg I have to define a variable conteining rtpproxy IP > address and I have to give it to function force_rtp_proxy. Only in > this way it works. > > I wait for your replay. > > I'm using SER 10.0.x CVS version! > > Thanks in advance > > Angelo > > > > > */Ladislav Andel <ladaan(a)iptel.org>/* ha scritto: > > It is saved in database table "user_attrs". Look at the page 24 in > optimized_nat_traversal.pdf > document how it > should look like. > UID in that table means username. > The value of variable "node" means the index of RTPproxy starting > from 0. > For example: > modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock > udp:5.6.7.8:22222") > > unix:/var/run/rtpproxy.sock is O > udp:5.6.7.8:22222 is 1 > > Within your ser.cfg you load these AVP variables through > load_attrs(); > and use them in nat mangle route. > You can check the value of "node" by dump_attrs(); before calling > force_rtp_proxy and see if there is correct value. > > Lada > > > > Angelo Taddeo wrote: > > > > > > > > Hi again > > I'm using ser 0.10.x ( CVS version). > > > > About rtpproxy traffic balancing: > > > > Where I have to put <$node> value???????? > > > > Thanks > > > > > ------------------------------------------------------------------------ > > L'email della prossima generazione? Puoi averla con la nuova Yahoo! > > Mail > > > > ------------------------------------------------------------------------ > L'email della prossima generazione? Puoi averla con la nuova Yahoo! > Mail > <%20http://us.rd.yahoo.com/mail/it/taglines/hotmail/nowyoucan/nextgen/*%20http://it.docs.yahoo.com/nowyoucan.html>

1 0

Re: [Users] radius authentication
by Stefano Capitanio 15 Feb '07

15 Feb '07

Hi, i've configured also "servers" files. i've also set: modparam("auth_radius", "service_type", 6) but i cannot set: modparam("auth_radius","Sip-Session",15), it says "parameter <Sip-Session> not found in module <auth_radius>" my dictionary file is: #### Attributes ### ATTRIBUTE User-Name 1 string # RFC2865, acc, auth_radius, avp_radius, group_radius, uri_radius ATTRIBUTE Service-Type 6 integer # RFC2865, acc, auth_radius, avp_radius, group_radius, uri_radius #ATTRIBUTE Called-Station-Id 30 string # RFC2865, acc #ATTRIBUTE Calling-Station-Id 31 string # RFC2865, acc #ATTRIBUTE Acct-Status-Type 40 integer # RFC2865, acc #ATTRIBUTE Acct-Session-Id 44 string # RFC2865, acc ATTRIBUTE Sip-Method 101 integer # Schulzrinne, acc ATTRIBUTE Sip-Response-Code 102 integer # Schulzrinne, acc ATTRIBUTE Sip-Cseq 103 string # Schulzrinne, acc ATTRIBUTE Sip-To-Tag 104 string # Schulzrinne, acc ATTRIBUTE Sip-From-Tag 105 string # Schulzrinne, acc ATTRIBUTE Sip-Translated-Request-URI 107 string # Proprietary, acc ATTRIBUTE Digest-Response 206 string # Sterman, auth_radius ATTRIBUTE Sip-Uri-User 208 string # Proprietary, auth_radius ATTRIBUTE Sip-Group 211 string # Proprietary, group_radius ATTRIBUTE Sip-Rpid 213 string # Proprietary, auth_radius ATTRIBUTE SIP-AVP 225 string # Proprietary, avp_radius ATTRIBUTE Digest-Realm 1063 string # Sterman, auth_radius ATTRIBUTE Digest-Nonce 1064 string # Sterman, auth_radius ATTRIBUTE Digest-Method 1065 string # Sterman, auth_radius ATTRIBUTE Digest-URI 1066 string # Sterman, auth_radius ATTRIBUTE Digest-QOP 1067 string # Sterman, auth_radius ATTRIBUTE Digest-Algorithm 1068 string # Sterman, auth_radius ATTRIBUTE Digest-Body-Digest 1069 string # Sterman, auth_radius ATTRIBUTE Digest-CNonce 1070 string # Sterman, auth_radius ATTRIBUTE Digest-Nonce-Count 1071 string # Sterman, auth_radius ATTRIBUTE Digest-User-Name 1072 string # Sterman, auth_radius ### CISCO Vendor Specific Attributes ### #VENDOR Cisco 9 #ATTRIBUTE Cisco-AVPair 1 string Cisco # VSA, auth_radius ### Acct-Status-Type Values ### #VALUE Acct-Status-Type Start 1 # RFC2866, acc #VALUE Acct-Status-Type Stop 2 # RFC2866, acc VALUE Acct-Status-Type Failed 15 # RFC2866, acc ### Service-Type Values ### VALUE Service-Type Call-Check 10 # RFC2865, uri_radius VALUE Service-Type Group-Check 12 # Proprietary, group_radius VALUE Service-Type Sip-Session 15 # Schulzrinne, acc, auth_radius VALUE Service-Type SIP-Caller-AVPs 30 # Proprietary, avp_radius VALUE Service-Type SIP-Callee-AVPs 31 # Proprietary, avp_radius any idea? raviprakash sunkara ha scritto: > Can U Check the servers in radiusclient dir. > That host and key should match the Radiu server NASlist file or nas > tables and client.conf file > > modparam("auth_radius","Sip-Session",15); > > On 2/15/07, *Stefano Capitanio* <s.capitanio(a)caspur.it > <mailto:s.capitanio@caspur.it>> wrote: > > Ciao! > > I'm trying to authenticate using Radius. > I've installed radiusclient-ng 0.5.2 > i've configured radiusclient.conf with the IP address of the > radius server. > I've configured SER with: > ... > modparam("auth_radius", "radius_config", > "/etc/radiusclient-ng/radiusclient.conf") > ... > if (method=="REGISTER") { > if (!radius_www_authorize(" example.it <http://example.it>")) { > www_challenge("example.it > <http://example.it>", "0"); > exit; > } > save("location"); > } > > but it doesn't work: there seem to be a problem in the client because > there is no IP Packet that goes out directed to the radius server... > in the log of SER i have no errors... > > thank you for any help, > Stefano > > _______________________________________________ > Users mailing list > Users(a)openser.org <mailto:Users@openser.org> > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > M:+91 9985077535 > www.hyperion-tech.com <http://www.hyperion-tech.com> > Client and Parent company :- www.august-networks.com > <http://www.august-networks.com>

1 0

[Serusers] mediaproxy exceptions on session delete
by Klaus Darilion 15 Feb '07

15 Feb '07

Hi! I have a problem with mediaproxy 1.8. Whenever a session is deleted there is an exception: Feb 14 14:18:27 foobar mediaproxy[27053]: delete OGI3Nzk1N2ExODY4NTIzYzIxMjc1NmYxN2MwMDBiZjU. info= Feb 14 14:18:27 foobar mediaproxy[27053]: session OGI3Nzk1N2ExODY4NTIzYzIxMjc1NmYxN2MwMDBiZjU.: 10558/10557/21113 packets, 778927/774750/1553531 bytes (caller/called/relayed) Feb 14 14:18:27 foobar mediaproxy[27053]: session OGI3Nzk1N2ExODY4NTIzYzIxMjc1NmYxN2MwMDBiZjU.: ended. Feb 14 14:18:27 foobar mediaproxy[27053]: error: uncaptured python exception, closing channel <rtphandler.CommandHandler connected 5.6.7.8:38879 at 0xb7d255ec> (exceptions.NameError:global name 'StopRecordSerializer' is not defined [/usr/lib/python2.3/asyncore.py|readwrite|86] [/usr/lib/python2.3/asyncore.py|handle_read_event|390] [/opt/mediaproxy/modules/rtphandler.py|handle_read|441] [/opt/mediaproxy/modules/rtphandler.py|process|568]) Feb 14 14:19:53 foobar mediaproxy[27053]: status Feb 14 14:19:53 foobar mediaproxy[27053]: execution time: 0.11 ms I could not find the problem - accounting.py with StopRecordSerializer is there, and it is imported in rtphandler. Do others also have problems and maybe a workaround? regards klaus -- Klaus Darilion nic.at

1 0

[Serusers] TCP main process and TCP receiver not starting
by Andrew T Gin 15 Feb '07

15 Feb '07

Hi I'm using the Hello World.cgf as my configuration file. My problem is that SER does not appear to be listening for TCP connections; it returns an RST when it receives a SYN. at http://mit.edu/sip/sip.edu/ser.shtml it says a correctly running SER listening on 10.1.2.3 and the loop back configured to use 4 processes looks like this: # serctl ps 0 25254 attendant 1 25258 receiver child=0 sock=0 @ 10.1.2.3:5060 2 25259 receiver child=1 sock=0 @ 10.1.2.3:5060 3 25261 receiver child=2 sock=0 @ 10.1.2.3:5060 4 25263 receiver child=3 sock=0 @ 10.1.2.3:5060 5 25265 receiver child=0 sock=1 @ 127.0.0.1:5060 6 25267 receiver child=1 sock=1 @ 127.0.0.1:5060 7 25274 receiver child=2 sock=1 @ 127.0.0.1:5060 8 25280 receiver child=3 sock=1 @ 127.0.0.1:5060 9 25282 fifo server 10 25312 timer 11 25314 tcp receiver 12 25316 tcp receiver 13 25318 tcp receiver 14 25320 tcp receiver 15 25323 tcp main process However when I run serctl ps I get the following: $ serctl ps 0 31360 stand-alone receiver @ 132.181.14.210:5060 1 31361 timer 2 31362 fifo server 3 0 4 0 5 0 6 0 7 0 This SER is listing on 132.181.14.210, but there is only a stand-alone receiver, not receiving childs. I have specified in the cfg file children=4, but it doesnt appear to do so. What am I missing, and why don't the tcp receivers, receiver childs and tcp main process start? Thanks Andrew

2 1

[Serusers] problem with status in serial forking (maybe related to bug 48)
by Zappasodi Daniele 15 Feb '07

15 Feb '07

Hello, I have a problem with serial forking and it seems the same behaviour of the bug 48: my ser.cfg script stops the first call due to time-out (fr_inv_timer), and the status in the failure_route is 408, and then sends a call to a second destination. If the caller hang up the call, the status in failure-route should be 487, but I have 408 again. The bug is declared as closed since the 0.9.5 (http://tracker.iptel.org/browse/SER-48), where can I take the patch? I have downloaded the version 0.9.6 from http://ftp.iptel.org/pub/ser/0.9.6/src/, but it doesn't seem to contain the patch for the bug. thanks ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

1 0

[Serusers] Inoltra: Problems with Rtpproxy traffic balancing in SER 0.10.x
by Angelo Taddeo 15 Feb '07

15 Feb '07

Hi again I'm using ser 0.10.x ( CVS version). About rtpproxy traffic balancing: Where I have to put <$node> value???????? Thanks --------------------------------- L'email della prossima generazione? Puoi averla con la nuova Yahoo! Mail

2 1

[Serusers] Routing based on FROM
by Edson 15 Feb '07

15 Feb '07

Hi, Guys... I'm challenged to implement a routing that consider the FROM header. That is, if call goes to X and comes from user U, then route to GW G1, else route to GW2. In version 0.9.6, how to find the "from user" value? Edson

1 0

[Serusers] Re: Serusers Digest, Vol 43, Issue 12
by wangdq＠gongye.com.cn 15 Feb '07

15 Feb '07

test On Sun, 2006-11-12 at 12:00 +0000, serusers-request(a)lists.iptel.org wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. SER not sending 486 Busy Here (Ravi *****) > 2. Re: SER pre-release - PRESENCE ISSUES (SER LIST) > 3. Suggestion (Eddie Johnson Jr) > 4. SER pre-release - Presence and NOTIFY (SER LIST) > 5. SER pre-release ser_ctl (SER LIST) > 6. Re: SER pre-release ser_ctl (Alexandr Dubovikov) > 7. Re: SER pre-release - PRESENCE ISSUES (Michal Matyska) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 11 Nov 2006 14:15:03 +0200 > From: "Ravi *****" <mravikreddy(a)gmail.com> > Subject: [Serusers] SER not sending 486 Busy Here > To: serusers(a)lists.iptel.org > Message-ID: > <cfbb389b0611110415l33701327vc3fd09fd9d571e10(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi Users , > > I am Using SER-0.96 , When I make a call from > SER to Pstn it works fine I mean is all "INV"BYE"CAN". > > But when I recieved a call from Pstn I cannot cancel the call because > X-lite is sending to pstn gateway 480 temp. service not available instead > of 486 Busy Here. > > I am using default config. but I dont understand where > it went wrong . > > Any Idea or help greatly appreciated > > Thank You. > Regards, > Ravi. >

1 0

[Serusers] Re: Serusers Digest, Vol 43, Issue 12
by wangdq＠gongye.com.cn 15 Feb '07

15 Feb '07

On Sun, 2006-11-12 at 12:00 +0000, serusers-request(a)lists.iptel.org wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. SER not sending 486 Busy Here (Ravi *****) > 2. Re: SER pre-release - PRESENCE ISSUES (SER LIST) > 3. Suggestion (Eddie Johnson Jr) > 4. SER pre-release - Presence and NOTIFY (SER LIST) > 5. SER pre-release ser_ctl (SER LIST) > 6. Re: SER pre-release ser_ctl (Alexandr Dubovikov) > 7. Re: SER pre-release - PRESENCE ISSUES (Michal Matyska) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 11 Nov 2006 14:15:03 +0200 > From: "Ravi *****" <mravikreddy(a)gmail.com> > Subject: [Serusers] SER not sending 486 Busy Here > To: serusers(a)lists.iptel.org > Message-ID: > <cfbb389b0611110415l33701327vc3fd09fd9d571e10(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi Users , > > I am Using SER-0.96 , When I make a call from > SER to Pstn it works fine I mean is all "INV"BYE"CAN". > > But when I recieved a call from Pstn I cannot cancel the call because > X-lite is sending to pstn gateway 480 temp. service not available instead > of 486 Busy Here. > > I am using default config. but I dont understand where > it went wrong . > > Any Idea or help greatly appreciated > > Thank You. > Regards, > Ravi. >

1 0

[Serusers] Forcing Account
by Gustavo Passos Tourinho 14 Feb '07

14 Feb '07

Hello Guys Is there a way to force proxy accounting on radius without/before use t_relay()? My problem is: I have a SIP URI that is received by the proxy but should change that URI before forward it. For example: Proxy receive an INVITE: sip:123@teste.com Proxy should send leg to radius Proxy change uri to: sip:098@example.com Proxy shoud foward INVITE (t_relay) I saw acc_rad_request ("forced") on ser.cfg, but Im not sure about what it does. Thanks for any help. Gustavo

1 0

[Users] nathelper and flag 2
by Andreas Granig 14 Feb '07

14 Feb '07

Hi, Today I found a UAC which is *not* located behind NAT (public IP 1.2.3.4) and sends this Via-Header, which seems perfectly valid according to RFC3261: SIP/2.0/UDP VINNASUP06C:5060;maddr=1.2.3.4;branch=z9hG4bK-2198d2 I used to check for nated clients using nat_uac_test("3"), which detects NAT in this case, because the host-part doesn't match the received-address. So is the test-flag "2" useless, since the host-part can be "hostname / IPv4address / IPv6reference", or should this particular test be extended to also check for the maddr-parameter? In the meanwhile, I've changed my nat-test to "17" for only testing Contact and Via-Port instead of Contact and Via-Address, but it's still not optimal. Any opinions on this? Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation.

3 8

[Users] imc
by henri keski-sikkila 14 Feb '07

14 Feb '07

Hello, When I invite new member to instance message conference I get the message that doesn't contain header: Content-Type: text/plain This couses me little bit problems...Should this header include to the reply? MESSAGE sip:xx SIP/2.0. Via: SIP/2.0/UDP xx;branch=z9hG4bK74fb.067ba117.0. To: sip:xx. From:sip:x;tag=0c3c711902a8bbb20a1e767e10f39a78-96ab. CSeq: 10 MESSAGE. Call-ID: 7bdbafcb-32586@x. Content-Length: 38. User-Agent: OpenSer (1.2.0-dev6-notls (i386/linux)). . x@x joined room.. left Best Regards, Henri Keski-Sikkila

3 6

[Users] Makefile error on arm-linux OS
by Michel Bensoussan 14 Feb '07

14 Feb '07

Hello I'm trying to compile OpenSER for an arm platform (Linux). The compiler is arm-linux-gcc I tried the following command: *$ make CC=arm-linux-gcc* And there is the following errors: *cc1: error: invalid option `inline-all-stringops' cc1: error: bad value (athlon) for -mcpu= switch * The arm-linux-gcc -v gives: *Reading specs from /usr/local/lib/gcc-lib/arm-linux/3.3.3/specs Configured with: ../configure --target=arm-linux --disable-shared --disable-threads --with-gnu-as --with-gnu-ld --enable-multilib --enable-languages=c Thread model: single gcc version 3.3.3 * I also have to modified the libs. Is there a simple way to do this porting or do I have to check all compilation and link options and modified or re-write the Makefiles? Regards, Michel

1 0

[Serusers] [PRESENCE]-Subscription does not exist
by SER LIST 14 Feb '07

14 Feb '07

Hello- I am running CVS version of ser w/ presence. All seems to be working well, except that I see a lots of packets being returned by SER indicating "Subscription does not exist". SIP/2.0 481 Subscription does not exist. From: <sip:111@xx.com>;tag=3ffe8a226de90fb471c542cb1065d20a. To: <sip:111@xx.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-9321. Call-Id: ssc-3d959a71f819794a430784374ee6f664@4bfc7569f5fad1a1a28a85dcb86badbb. Cseq: 11 SUBSCRIBE. Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK01ee.cbc45cb3.0. Via: SIP/2.0/UDP 192.168.1.3:5065;received=67.X.X.X;branch=z9hG4bK-ecfc3a30c3920b9e;rport=5065. Server: Sip EXpress router (0.10.99-dev57-tm_fixes (i386/linux)). Content-Length: 0. These are in response to a SUBSCRIBE request with Event: presence.winfo - The packet above was sent by SER in response to: SUBSCRIBE sip:X.X.X.X:5060 SIP/2.0. Record-Route: <sip:X.X.X.X;ftag=3ffe8a226de90fb471c542cb1065d20a;lr>. From: <sip:111@xx.com>;tag=3ffe8a226de90fb471c542cb1065d20a. To: <sip:111@xx.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-9321. Call-Id: ssc-3d959a71f819794a430784374ee6f664@4bfc7569f5fad1a1a28a85dcb86badbb. Cseq: 11 SUBSCRIBE. Contact: sip:111@sip.tpad.com. Event: presence.winfo. Expires: 600. Date: Tue, 13 Feb 2007 21:18:18 GMT. Max-Forwards: 16. User-Agent: SomUserAgent Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, INFO, SUBSCRIBE, NOTIFY, MESSAGE. Supported: replaces. Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK01ee.cbc45cb3.0. Via: SIP/2.0/UDP 192.168.1.3:5065;received=67.X.X.X;branch=z9hG4bK-ecfc3a30c3920b9e;rport=5065. Content-Length: 0. P-hint: Loose Routed. Any suggestions? Thanks in advance? _________________________________________________________________ Buy what you want when you want it on Sympatico / MSN Shopping http://shopping.sympatico.msn.ca/content/shp/?ctId=2,ptnrid=176,ptnrdata=08…

2 1

[Users] help-me in 800
by Jobson Andrade 13 Feb '07

13 Feb '07

Hi All, I,m use openser 1.0.1 We imagined the following setting: 1- It Receive the connections of a pstn that sends all of the requisition for my one openser 2- it authenticate the coming calls of that IP 3- It Receive the PIN, authenticate and direct the call for the routes The question is like do that verification of the PIN´s and send the calls after that verification It plan is: It receives itself 0800 123 123 of an ONLY IP, through the src_ip==xxx.xxx.xxx.xxx sends for a route I obtained to remove with the strip I number him 0800 123 123 and with the prefix (""); send with the rewritehostport for a route I need that be attended, authenticated the PIN and receive of the side "A" I number of fate and sent for the route of smaller cost How to made this???? Please help-me!! Thanks in advanced!! Jobson Andrade Projetos & Desenvolvimento Obelisk - The Asterisk & VoIP Experts phone/fax: (11) 2164-4808 ext. 115 cell Phone: (11) 8175-9916 / 8271-0480 email: <mailto:jandrade@obelisknet.com.br> jandrade(a)obelisknet.com.br

1 0

[Users] script to create dbtext tables
by Cesc 13 Feb '07

13 Feb '07

Hi, I see there are scripts to automatically create the tables for mysql and postgres ... Does an equivalent tool exist for dbtext? Cesc

2 2

Re: Re: Re: Re: Re: Re: Re: [Serusers] call forwarding issues - load_attrs and lookup_contacts - ottendorf
by tzieleniewski 13 Feb '07

13 Feb '07

I did some tests. I sent them on the list because I wat not sure how to interpret their correctness beefore putting them on the tracker I did the following general procedure: dump_attrs() lookup_user(..) dump_attrs() load_attrs(..) dump_attrs() lookup_contacts(..) Case one: dump_attrs() lookup_user("$t.uid","@ruri") dump_attrs() load_attrs("$tu","$t.uid") dump_attrs() lookup_contacts("location") my observations: after lookup_user("$t.uid","@ruri") uid avp appered in the uri class and to track, lookup_contacts("location") found the user debug log: Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:540: class=GLOBAL Feb 13 13:48:25 rd ser[9875]: AVP["lang"]="en" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 13:48:25 rd ser[9875]: AVP["did"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: AVP["digest_realm"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 13:48:25 rd ser[9875]: AVP["did"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: AVP["digest_realm"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:570: track=FROM class=USER Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:580: track=TO class=USER Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:590: track=FROM class=URI Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:600: track=TO class=URI Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: route[INBOUND]: (lookup_user($t.uid, @ruri)) Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:540: class=GLOBAL Feb 13 13:48:25 rd ser[9875]: AVP["lang"]="en" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 13:48:25 rd ser[9875]: AVP["did"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: AVP["digest_realm"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 13:48:25 rd ser[9875]: AVP["did"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: AVP["digest_realm"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:570: track=FROM class=USER Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:580: track=TO class=USER Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:590: track=FROM class=URI Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:600: track=TO class=URI Feb 13 13:48:25 rd ser[9875]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: route[INBOUND]: load_attrs($tu,$t.uid) Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:540: class=GLOBAL Feb 13 13:48:25 rd ser[9875]: AVP["lang"]="en" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 13:48:25 rd ser[9875]: AVP["did"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: AVP["digest_realm"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 13:48:25 rd ser[9875]: AVP["did"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: AVP["digest_realm"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:570: track=FROM class=USER Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:580: track=TO class=USER Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:26 rd ser[9875]: INFO: avp.c:590: track=FROM class=URI Feb 13 13:48:26 rd ser[9875]: INFO: No AVP present Feb 13 13:48:26 rd ser[9875]: INFO: avp.c:600: track=TO class=URI Feb 13 13:48:26 rd ser[9875]: AVP["uid"]="hellboy(a)voip.touk.pl" Case two: dump_attrs() lookup_user("Request-uri") dump_attrs() load_attrs("$tu","$tu.uid") dump_attrs() lookup_contacts("location") my observations: after lookup_user("Request-uri") uid avp appered in the user class and to track, lookup_contacts("location") didn't find the user Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:540: class=GLOBAL Feb 13 14:22:57 rd ser[10047]: AVP["lang"]="en" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 14:22:57 rd ser[10047]: AVP["did"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 14:22:57 rd ser[10047]: AVP["did"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:570: track=FROM class=USER Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:580: track=TO class=USER Feb 13 14:22:57 rd ser[10047]: INFO: No AVP present Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:590: track=FROM class=URI Feb 13 14:22:57 rd ser[10047]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:600: track=TO class=URI Feb 13 14:22:57 rd ser[10047]: INFO: No AVP present Feb 13 14:22:57 rd ser[10047]: route[INBOUND]: (lookup_user(Request-uri) Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:540: class=GLOBAL Feb 13 14:22:57 rd ser[10047]: AVP["lang"]="en" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 14:22:57 rd ser[10047]: AVP["did"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 14:22:57 rd ser[10047]: AVP["did"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:570: track=FROM class=USER Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:580: track=TO class=USER Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["ruri_canonical"]="1" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:590: track=FROM class=URI Feb 13 14:22:57 rd ser[10047]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:600: track=TO class=URI Feb 13 14:22:57 rd ser[10047]: INFO: No AVP present Feb 13 14:22:57 rd ser[10047]: route[INBOUND]: load_attrs($tu,$t.uid) Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:540: class=GLOBAL Feb 13 14:22:57 rd ser[10047]: AVP["lang"]="en" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 14:22:57 rd ser[10047]: AVP["did"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 14:22:57 rd ser[10047]: AVP["did"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:570: track=FROM class=USER Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:580: track=TO class=USER Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["ruri_canonical"]="1" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:590: track=FROM class=URI Feb 13 14:22:57 rd ser[10047]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:600: track=TO class=URI Feb 13 14:22:57 rd ser[10047]: INFO: No AVP present Case three: dump_attrs() lookup_user("$tu.uid","@ruri") dump_attrs() load_attrs("$tu","$tu.uid") dump_attrs() lookup_contacts("location") my observations: after lookup_user("$tu.uid","@ruri") uid avp appered in the user class and to track, lookup_contacts("location") didn't find the user I also tried to invoke lookup_contacts("location","$tu.uid") but with the same result user wasn't found. Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:540: class=GLOBAL Feb 13 14:36:42 rd ser[10100]: AVP["lang"]="en" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 14:36:42 rd ser[10100]: AVP["did"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 14:36:42 rd ser[10100]: AVP["did"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:570: track=FROM class=USER Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:580: track=TO class=USER Feb 13 14:36:42 rd ser[10100]: INFO: No AVP present Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:590: track=FROM class=URI Feb 13 14:36:42 rd ser[10100]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:600: track=TO class=URI Feb 13 14:36:42 rd ser[10100]: INFO: No AVP present Feb 13 14:36:42 rd ser[10100]: route[INBOUND]: (lookup_user($tu.uid, @ruri)) Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:540: class=GLOBAL Feb 13 14:36:42 rd ser[10100]: AVP["lang"]="en" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 14:36:42 rd ser[10100]: AVP["did"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 14:36:42 rd ser[10100]: AVP["did"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:570: track=FROM class=USER Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:580: track=TO class=USER Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:590: track=FROM class=URI Feb 13 14:36:42 rd ser[10100]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:600: track=TO class=URI Feb 13 14:36:42 rd ser[10100]: INFO: No AVP present Feb 13 14:36:42 rd ser[10100]: route[INBOUND]: load_attrs($tu,$tu.uid) Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:540: class=GLOBAL Feb 13 14:36:42 rd ser[10100]: AVP["lang"]="en" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 14:36:42 rd ser[10100]: AVP["did"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 14:36:42 rd ser[10100]: AVP["did"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:570: track=FROM class=USER Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:580: track=TO class=USER Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:590: track=FROM class=URI Feb 13 14:36:42 rd ser[10100]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:600: track=TO class=URI Feb 13 14:36:42 rd ser[10100]: INFO: No AVP present So generally lookup_contacts works only if it can find the uid in the uri class even if there are uid set in other classes it doesn' find the user. Another thing is that when invoked lookup_user("$t.uid","@ruri") the uid avp is put in the uri track. Maybe instead of lookup_contacts one should use registerd() function?? But what does it do? does it only check the location table or as well make an uri set?? Bests Tomasz > On Mon, 2007-02-12 at 22:13 +0100, tzieleniewski wrote: > > Thank You:) It help! > > But actually I had it done with the usage of lookup_user("$t.uid","@ruri"). > > does it make any difference from lookup_user("$tu.uid","@ruri") when I use $t.uid instead of $tu.uid? this is putting this avp to general class with the To track isn't it?? > > when I changed to lookup_user("Request-uri") it worked so generally works:) but not for lookup_user("$t.uid","@ruri"). > > Obviously there must be some difference... if it has helped. :-) You can > check what it has done using dump_attrs() call. It will output all AVPs > set to the syslog (must have debug >= 3 iirc). > > It should not go into general class, if not specified the user should be > the correct one... check the dump_attrs output and fill bug report at > tracker.iptel.org. (I guess it set the uri class and lookup_contacts > checks user class only) > > Michal > > > is there some order and priority in which the lookup_contacts() searches for uid?? (for instance first in user class and then I global??) > > Basically uri, user, domain, global is the order if you specify track > only AVP e.g. $t.uid so you can set widely accepted default value > (domain/global) and override it for some users and uris if you want. > > Some functions (like lookup_contacts) don't allow to set the avp name, > have to check the code what they expect. > > If you specify class too (e.g. $tu.uid) just the only one track/class is > searched. > > Michal > > > > > tomasz > > > > > The difference is that there is the forwarded request visible... > > > # > > > U 2007/02/12 20:35:16.241648 192.168.1.2:5060 -> 192.168.1.2:5060 > > > INVITE sip:hellboy@tezet.no-ip.org SIP/2.0. > > > Record-Route: <sip:192.168.1.2;ftag=1173592111;lr=on>. > > > Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKb946.ee8b4793.0. > > > Via: SIP/2.0/UDP > > > 192.168.1.2:7061;rport=7061;branch=z9hG4bK7AF94D00BA07C6380C5EC3434F4EB592. > > > From: tomix <sip:tomix@tezet.no-ip.org:7061>;tag=1173592111. > > > To: <sip:tomix@tezet.no-ip.org>. > > > Contact: <sip:tomix@192.168.1.2:7061>. > > > Call-ID: 0E6FCCFF-D661-0186-3392-977A4BBCDE86(a)192.168.1.2. > > > CSeq: 23808 INVITE. > > > Proxy-Authorization: Digest > > > username="tomix",realm="tezet.no-ip.org",nonce="45d0c2a0f928f22d790a5dfa17228b193d454c7e",response="0df094e8ea2808fa71d2fa28ccdfa8a4",uri="sip:tomix@tezet.no-ip.org",qop=auth,cnonce="67344073698582511BAA60061EBDA625",nc=00000001. > > > Max-Forwards: 16. > > > Content-Type: application/sdp. > > > User-Agent: X-Lite release 1105d. > > > Content-Length: 308. > > > > > > > > > If I understand it correctly you get this request being forwarded to the > > > contacs of tomix(a)tezet.no-ip.org and not hellboy@.... using usrloc > > > lookup (P-hint in the next INVITE). > > > > > > Then you must have set the TO / USER avp name "uid" based on the To > > > header instead of request-uri. Check the lookup_user function calls, it > > > should be > > > for REGISTER > > > lookup_user("To") or lookup_user("$tu.uid","@to.uri") > > > > > > and for other requests > > > lookup_user("Request-uri") or lookup_user("$tu.uid","@ruri") > > > > > > > > > If you want to do some originating services, you can use > > > lookup_user("From") or lookup_user("$fu.uid", "@from.uri") > > > > > > Michal > > > > > > > > > > > > On Mon, 2007-02-12 at 20:41 +0100, tzieleniewski wrote: > > > > Ok I did it. > > > > There is no difference. > > > > Just to make sure I attached the file. > > > > > > > > Because I left work I repeated the situation at home. I call to "myself" tomix(a)tezet.no-ip.org and I try to forward to hellboy(a)tezet.no-ip.org which is not registered. > > > > > > > > I can also send my ser.cfg if it might help. > > > > > > > > tomasz > > > > > > > > > I see. Please capture the network on linux cooked interface "any", so > > > > > even the request sent over loopback will be visible. > > > > > > > > > > Michal > > > > > > > > > > On Mon, 2007-02-12 at 16:53 +0100, tzieleniewski wrote: > > > > > > Yes but this ruri sip:hellboy@192.168.0.116:5060 is set after the invocation of lookup_contacts(). the function is invoked on the message which contains ruri changed by the forward_blind parameter. > > > > > > There is first "round" when processing of the first INVITE reaches the checking of the forward_blind parameter after which I invoke the attr2uri and just after this make the t_relay. then there is second "round" with the changed ruri. before lookup_contacts I see the ruri as mm(a)voip.touk.pl and after sip:hellboy@192.168.0.116:5060 which is in fact the location corresponding to sip:hellboy@voip.touk.pl. > > > > > > there is no message sending through the network but the message with the new ruri is processed which is visible in the log file I see it logged with the changed ruri?? > > > > > > > > > > > > So where is the problem?? > > > > > > Is it the problem of attr2uri? > > > > > > I tried it by using rewriteuri() and it gave me the same result. changed ruri but lookup_contacts() returns value corresponding to the first ruri. Maybe > > > > > > lookup_contacts() checks not the ruri?? > > > > > > > > > > > > > > > > > > > Yes, your request-uri is sip:hellboy@192.168.0.116:5060 which I think > > > > > > > will not be looked up in location (lookup_contact should fail), so it > > > > > > > does not rewrite the request-uri at all. > > > > > > > > > > > > > > Michal > > > > > > > > > > > > > > On Mon, 2007-02-12 at 16:05 +0100, tzieleniewski wrote: > > > > > > > > hmm > > > > > > > > it is strange because I can see that after attr2uri and t_relay() message again enters the main route block and goes through the whole processing but I can't see the message being send through the network. the recourd_route and via headers are being attached which is visible in the message send to unwanted (contained in the first invite - in this case myself because I try this by calling my self and setting forward_blind to another sip uri) destination: > > > > > > > > > > > > > > > > U 2007/02/12 16:05:12.199566 192.168.0.74:506 >

1 1

[Users] Send Call using authentication
by Jobson Andrade 13 Feb '07

13 Feb '07

Hi All, I need of send the calls for my ITSP using verification. Ex.: User = user_itsp Passwd = pass_itsp SIP Server = itsp.com.br I my configuration I made of this way, more not work. ... ##Obelisk if (uri=~"^sip:00[1-9][1-9][0-9]*@") { route(3); exit; } ... route[3]{ #sending route to DDI strip(2); rewritehostport("itsp.com.br:5060"); rewriteuser("user_itsp"); rewriteuserpass("pass_itsp"); route(1); exit; } ... How to made this in openser? I use the openser version 1.0.1 Thanks in advanced, Jobson Andrade Projetos & Desenvolvimento Obelisk - The Asterisk & VoIP Experts phone/fax: (11) 2164-4808 ext. 115 cell Phone: (11) 8175-9916 / 8271-0480 email: <mailto:jandrade@obelisknet.com.br> jandrade(a)obelisknet.com.br

1 0

[Users] Doubts about installation
by Javier Allende Astigarraga 13 Feb '07

13 Feb '07

i'm begining with openser , i've installed on ubuntu dapper but reading installation notes , always speak about openser-mysql-module, persistent, but i cant do apt.get install .... and if i seek i find files for http://packages.ubuntulinux.org/feisty/net/ only download and put con mysql/lib?? thanks for all any information about openserv config?? bye bye -- ===================================================== Legolas_Bilbao[ID2006][GKR] Dios creo un equipo Perfecto a los demas los lleno de extranjeros http://www.forosindicedonkey.com http://usuarios.lycos.es/ligaforo/ =====================================================

2 1

[Serusers] call forwarding issue - attr_destination() unwanted behavior - ottendorf
by tzieleniewski 13 Feb '07

13 Feb '07

Hi!! I am trying to set up the blind call forwarding with the usage of attr_destination() function. I do the following I load the user attributes from user_attrs and uri_attrs table and then according to the loaded parameters I do: if ($tu.call_forward == "blind" && $tr.forward_blind) { xlog("L_INFO", " route[CALL_FORWARD]: fwd \n"); attr_destination("$tr.forward_blind"); xlog("L_INFO", " route[CALL_FORWARD]: route(FORWARD) \n"); route(FORWARD); xlog("L_INFO", " route[CALL_FORWARD]: drop \n"); drop; } after the ser.cfg logic goes through this part I don't see any change in the request uri?? Please point me what do I miss? here is my database contents: mysql> select * from user_attrs where uid like 'hellboy%' and name like 'call_%'; +----------------------+--------------+-------+------+-------+ | uid | name | value | type | flags | +----------------------+--------------+-------+------+-------+ | hellboy(a)voip.touk.pl | call_forward | blind | 2 | 1 | +----------------------+--------------+-------+------+-------+ 1 row in set (0.00 sec) mysql> select * from uri_attrs where username like 'hellboy%'; +----------+--------------+---------------+----------------------+------+-------+--------+ | username | did | name | value | type | flags | scheme | +----------+--------------+---------------+----------------------+------+-------+--------+ | hellboy | voip.touk.pl | forward_blind | sip:tzl@voip.touk.pl | 2 | 1 | sip | +----------+--------------+---------------+----------------------+------+-------+--------+ 1 row in set (0.00 sec) Cheers -tomasz

2 12

[Users] Presence server -Confused about the Notify
by H Quintana 13 Feb '07

13 Feb '07

Hello, I'm a newbie with the SIMPLE extensions. This week I started using the presence server module (with the configuration suggested in the wiki and 2 X-lite's) The softphones are sending the SUBSCRIBE and the PUBLISH messages. The NOTIFYs are received for the Event presence.winfo with some XML content: <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="0" state="full"> <watcher-list resource="sip:10092011@a.b.c.d" package="presence"/> </watcherinfo> But, when I add each softphone as a Contact for the other there is no XML content for the Event presence, and the contact's status never changes even if I do multiple changes. I noticed that in the table xcap_xml there is no data, Should I put some information manually? or the information should be inserted when receiving the PUBLISH? Am I missing something? Any ideas? Thanks for your help, Humberto Quintana ____________________________________________________________________________________ Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121

2 1

[Users] several questions/concerns about branches
by Klaus Darilion 13 Feb '07

13 Feb '07

Hi! I hav the following questions about branch handling in openser 1.1.1: In route[1] I use the dispatcher to forward to the gateway 11.22.33.46. Thus, the DURI will be set. Further I use port 6060 to send calls to the GW. So far everything is ok. ... setting fr_timer to 2 seconds XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX route[1]: request's uri: $ru=sip:123456789@foobar.net route[1]: request's destination uri: $du=sip:11.22.33.46:5060 route[1]: request's force_send_sock: $fs=udp:11.22.33.44:6060 route[1]: request's first branch: $br=<null> route[1]: request's all branches: $bR= route[1]: request's destination set: $ds=Contact: sip:123456789@foobar.net ... t_relay Then, if the GW does not reply within 2 seconds the failure route triggers: First question: Why is the DURI set to NULL now? ... entering failure route ERROR: no response from gateway or clients a1.net ... branches before ds_next_dst: $ru=sip:123456789@foobar.net, $du=, $bR= XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX failure_route[1]: request's uri: $ru=sip:123456789@foobar.net failure_route[1]: request's destination uri: $du= failure_route[1]: request's force_send_sock: $fs=udp:11.22.33.44:6060 failure_route[1]: request's first branch: $br=<null> failure_route[1]: request's all branches: $bR= failure_route[1]: request's destination set: $ds=Contact: sip:123456789@foobar.net ... ds_next_dst branches after ds_next_dst: $ru=sip:123456789@foobar.net, $du=, $bR=sip:123456789@foobar.net XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX failure_route[1]: request's uri: $ru=sip:123456789@foobar.net failure_route[1]: request's destination uri: $du= failure_route[1]: request's force_send_sock: $fs=udp:11.22.33.44:6060 failure_route[1]: request's first branch: $br=sip:123456789@foobar.net failure_route[1]: request's all branches: $bR=sip:123456789@foobar.net failure_route[1]: request's destination set: $ds=Contact: sip:123456789@foobar.net, sip:979004369911 ... activating branch_route ... t_relay Second question: In the branch route the send_socket is reported as NULL. AFAIK it will be copied from branches[0] to the new branch during ds_next_dst. Also the request is sent from port 6060 - thus it looks like the $fs is wrong in branch_route. ==================================================== branch_route[1]: request's uri: $ru=sip:123456789@foobar.net branch_route[1]: request's destination uri: $du=sip:11.22.33.45:5060 branch_route[1]: request's force_send_sock: $fs=<null> branch_route[1]: request's first branch: $br=sip:123456789@foobar.net branch_route[1]: request's all branches: $bR=sip:123456789@foobar.net branch_route[1]: request's destination set: $ds=Contact: sip:123456789@foobar.net, sip:9790043699111 thanks klaus

5 8

[Serusers] branch parameter
by Kapil Dhawan 12 Feb '07

12 Feb '07

Whats the importance of branch parameter. I am using RTC/1.2 library and its not generating branch parameter. _________________________________________________________________ Tried the new MSN Messenger? Its cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in

5 7

Re: [Users] Corrupted 'To' after uac_restore_from
by Mahesh Paolini-Subramanya 12 Feb '07

12 Feb '07

Bogdan I'm sorry - I commited the cardinal cut/paste sin (copying stuff incorrectly :-( ) Anyhow, I've *correctly* cut/paste the INVITE and BYE headers below, and it looks to me like the route headers *are* being rewritten correctly. So, it doesn't look like that was the issue - unless you are referring to something else? cheers --- Original Message ----- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> To: mahesh(a)aptela.com Cc: users(a)openser.org Sent: Thursday, February 8, 2007 3:29:26 AM GMT-0600 Subject: Re: [Users] Corrupted 'To' after uac_restore_from Hi Mahesh, I suspect you have a client that does not preserve the Route/Record-Route parameters. According to RFC326, the parameters must not be altered at al, but I saw UAC doing lowercase on the param's value. Most probably this is your case also - the uac module use B64 encoding of the old uri in RR param, so this is case sensitive. Get the full trace of the call and check the original RR param (inserted by openser in INVITE) against the one in put in the BYE. regards, bogdan ---- CORRECTED MESSAGE----- I'm seeing this *very* weird error where uac_restore_from is ending up with a corrupted 'From' header. The weird thing is that it happens with only one end point (a Televantage SIP Trunk). With virtually *everything* else - and I am talking about tens of thousands of end points, spanning all sorts of sip devices, we are having *no* problems. The call is set up as follows -> the endpoint sends the INVITE, which is forwarded to a PSTN gateway The ACKs and the 200 OK all have their From/To headers appropriately rewritten The problem arises when the PSTN gateway initiates the BYE As you can see below, the 'To' header gets corrupted The *only* thing I can think of is that it has something to do with the structure of the Televantage tags (the '+' signs?). Any ideas? FYI, Openser 1.1.0 (latest CVS checkout), and modparam("uac","from_restore_mode","auto") modparam("uac","rr_store_param","aptelavsf") --INVITE-- UAC->proxy INVITE sip:17038677006@69.25.47.136 SIP/2.0. Via: SIP/2.0/UDP 10.10.35.100;rport;branch=z9hG4bK+ec6a2a20436621337016546fa4dc9e87+10.10.35.100+1. Allow-Events: message-summary. Allow-Events: presence. Allow-Events: refer. Max-Forwards: 70. Call-ID: 56796C1E(a)10.10.35.100. From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>. CSeq: 132610708 INVITE. Expires: 180. Supported: replaces. Supported: 100rel. Contact: <sip:6462174165@10.10.35.100>. Content-Type: application/sdp. Content-Length: 242. . User-Agent: TeleVantage-UA/7.50.4817. . v=0. o=TeleVantage 10671 0 IN IP4 10.10.35.100. s=phone-call. c=IN IP4 10.10.35.100. t=0 0. m=audio 6036 RTP/AVP 0 18 101. a=rtpmap:0 pcmu/8000/1. a=rtpmap:18 g729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000/1. a=ptime:20. ---INVITE--- proxy->gateway INVITE sip:17038677006@65.91.91.16:5060;transport=udp SIP/2.0. Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>. Via: SIP/2.0/UDP 69.25.47.136;branch=z9hG4bKf67d.2d31dbb.0. Via: SIP/2.0/UDP 10.10.35.100;rport=5060;branch=z9hG4bK+ec6a2a20436621337016546fa4dc9e87+10.10.35.100+1. Allow-Events: message-summary. Allow-Events: presence. Allow-Events: refer. Max-Forwards: 69. Remote-Party-ID: "Membersevice Queue" <sip:6462174165@flareon.aptela.com>;party=calling;id-type=subscriber;screen=yes;privacy=off. Supported: replaces. Call-ID: 56796C1E(a)10.10.35.100. From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>. CSeq: 132610708 INVITE. Expires: 180. Contact: <sip:6462174165@10.10.35.100:5060>. Content-Type: application/sdp. Content-Length: 262. User-Agent: TeleVantage-UA/7.50.4817. . v=0. o=TeleVantage 10671 0 IN IP4 10.10.35.100. s=phone-call. c=IN IP4 66.150.122.23. t=0 0. m=audio 59520 RTP/AVP 0 18 101. a=rtpmap:0 pcmu/8000/1. a=rtpmap:18 g729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000/1. a=ptime:20. a=nortpproxy:yes . ---BYE---gateway->proxy BYE sip:6462174165@10.10.35.100:5060 SIP/2.0. Via: SIP/2.0/UDP 66.52.236.25:5060;branch=z9hG4bKd8orpb20eg2hais2c2g1sd0000g00.1. To: "Membersevice Queue"<sip:6462174165@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. Call-ID: 56796C1E(a)10.10.35.100. CSeq: 1 BYE. Content-Length: 0. Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>. Max-Forwards: 70. ---BYE--- proxy->UAC BYE sip:6462174165@10.10.35.100:5060 SIP/2.0. Record-Route: <sip:69.25.47.136;lr;ftag=1386496>. Via: SIP/2.0/UDP 69.25.47.136;branch=z9hG4bK7791.48e0e2f5.0. Via: SIP/2.0/UDP 66.52.236.25:5060;branch=z9hG4bKd8orpb20eg2hais2c2g1sd0000g00.1. To: "Membersevice Queue"<sip:user100.cru>6'(!.l asr}XV.R\[>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. Call-ID: 56796C1E(a)10.10.35.100. CSeq: 1 BYE. Content-Length: 0. Max-Forwards: 69. -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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[Users] MAJOR FIXUP: avp access in reply route
by Bogdan-Andrei Iancu 12 Feb '07

12 Feb '07

Hi everybody, The ability of accessing/processing AVPs in reply route was a long debated and wanted functionality. Finally, OpenSER 1.2.x fixes this. The new code allows two types of behaviours: - use per message AVPs in reply route - no special synchronization is needed, but there is no persistence outside reply route. - use per transaction AVPS in reply route (as in failure route). the above behaviours apply to the reply routes triggered by TM. You can control this via the new TM parameter "onreply_avp_mode": http://www.openser.org/docs/modules/1.2.x/tm.html#AEN303 in default reply route only per message AVPs will be available. regards, bogdan

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[Users] invitation: SER User and Developer Meeting, collocated with the IETF, in Prague, March
by Jiri Kuthan 12 Feb '07

12 Feb '07

Ladies and gentlemen, boys and girls, be wholeheartly welcome to the SER user/developer meeting in Prague, March, colocated with the IETF meeting. For more information see http://www.iptel.org/ietf_meeting If interested in participation, just register yourself. Whereas this event is coming from the SER crowd, the content is going to be primarily technical and openser all in all draws greatly from SER. I believe this can be most useful for users(a)openser.org as well. -jiri -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] ser log file
by zahra bahar 12 Feb '07

12 Feb '07

could we see the processes that ser has when receives each massage? is there any log file for ser in about this? --------------------------------- Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit.

1 0

[Users] DB tables for presence module
by Reinhold Buchinger 12 Feb '07

12 Feb '07

Hello, Can you explain me the difference between the information stored in the "watchers" table and "active-watchers" table? When is a watcher insert in which table? Thanks a lot! Best regards, Reinhold --

2 1

[Users] MEDIAPROXY - Cannot detect inactive rtp sessions, any other BYE fix method?
by Dan-Cristian Bogos 12 Feb '07

12 Feb '07

Hi Guys, I know that this is not the support channel of mediaproxy, but just hoping that somebody from ag-projects is monitoring this list as well. I found out that, in my case, the SIP-BYE issue is still persistent even by using mediaproxy . If I will stop sending rtp through mediaproxy (eg by killing my client from task manager, or loosing the internet), mediaproxy will still find the session active, like nothing happened and therefore the bill will be inaquarate. I am using the version 1.7.2, and as read in the change log, 1.8.0 didn't change anything on this chapter. Is there any other solution or workarround available? Thxs, Dan

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[Serusers] Patch for MySQL 5.0.19 and above in SER 0.9.x
by sip 12 Feb '07

12 Feb '07

I've attached a patch file for people to use (it really only adds 3 lines) to patch the my_con.c file in the mysql module for SER 0.9.x I've tested it for a week with inactivity, restarts of mysql, etc, and it holds up rather well, but of course I would be remiss if I didn't add a 'use at your own risk' disclaimer. ;) The basic problem is that, after MySQL 5.0.3, they changed the default behaviour of the mysql connection socket to NOT reconnect in the case of a timeout or a restart of the server. If you had SER up and running for a while, or if there were a glitch with your mysql, the connection would time out/disconnect, and SER would no longer be able to hit the database (requiring a restart of SER). The MySQL coders added an option to enable the reconnect ability of the mysql connection, but it didn't work until 5.0.19. SO... if you're using mysql 5.0.19 or above with SER 0.9.x (there may only be 3 of us doing this ;) ), here's a patch file for you to apply to my_con.c in your modules/mysql directory. I'd recommend backing up your my_con.c file before patching. N.

1 0

[Users] ?opsr? - setup
by Michel Bensoussan 12 Feb '07

12 Feb '07

Hello I have some setup problems. I compiled and install as following: /prompt$ make mode=debug all prompt$ su prompt# make install /To run openSer I tried: /prompt# /usr/local/sbin/openser / and /prompt# /usr/local/sbin/openser D E / in both case there is the following output: / 127.0.0.1 [127.0.0.1]:5060 192.168.1.86 [192.168.1.86]:5060 192.169.4.86 [192.169.4.86]:5060 192.168.13.86 [192.168.13.86]:5060 127.0.0.1 [127.0.0.1]:5060 192.168.1.86 [192.168.1.86]:5060 192.169.4.86 [192.169.4.86]:5060 192.168.13.86 [192.168.13.86]:5060 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.86 [192.168.1.86]:5060 udp: 192.169.4.86 [192.169.4.86]:5060 udp: 192.168.13.86 [192.168.13.86]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.86 [192.168.1.86]:5060 tcp: 192.169.4.86 [192.169.4.86]:5060 tcp: 192.168.13.86 [192.168.13.86]:5060 Aliases: tcp: michel:5060 tcp: michel.local:5060 tcp: localhost:5060 udp: michel:5060 udp: michel.local:5060 udp: localhost:5060 //prompt# /Attached my config file ( /etc/openser/openser.cfg). On this config file, I set listen=192.168.13.86 so I expected the openSER to listen only 192.168.13.86. Actually I don't understand from where the application takes the openser.cfg file? I removed /etc/openser/openser.cfg but the application is still running. / /Regards, Michel. / / # # $Id: openser.cfg,v 1.6.2.1 2006/07/17 15:51:03 klaus_darilion Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode */ fork=no log_stderror=yes /**/ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.13.86 port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/openser/modules/auth.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; 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(); }; exit; }

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Re: [Users] A better way to count the length of a variable
by Tavis P 12 Feb '07

12 Feb '07

Ah efficient length counting, you're only a version away... Thanks Brandon Brandon Armstead wrote: > Tavis P wrote: >> I'm trying to find a fast way to count the length of a variable (and >> detect when the length exceeds a certain value) >> All i've come up with so far is a regex, does anyone know of a >> faster/better way to do this? >> >> >> Heres my current approach: >> !avp_check("$ci", "re/^.{1,128}$/g") >> >> >> tavis >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > Hello Tavis, > > Currently that is the only method available, however keep your eye > out for the release of OpenSER version 1.2.0, it was recently froze in > CVS and hopefully will be ready pretty soon, however here are some of > the methods that will be available: > http://openser.org/dokuwiki/doku.php/transformations:devel. > > Regards, > Brandon > >

1 0

[Users] A better way to count the length of a variable
by Tavis P 12 Feb '07

12 Feb '07

I'm trying to find a fast way to count the length of a variable (and detect when the length exceeds a certain value) All i've come up with so far is a regex, does anyone know of a faster/better way to do this? Heres my current approach: !avp_check("$ci", "re/^.{1,128}$/g") tavis

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[Serusers] Problems with Rtpproxy traffic balancing in SER 0.10.x
by Angelo Taddeo 12 Feb '07

12 Feb '07

I'm using ser 0.10.x ( CVS version). About rtpproxy traffic balancing, I don't know where I have to put rtpproxy IP address and where I have to set $node variable value. How does SER obtain rtpproxy IP address ? What the relation between rtpproxy IP address and N0, N1.....etc ? What parameters have I to put in force_rtp_proxy() function? Only rtpproxy IP address? In my framework happens : when SER receives INVITE Request and 200 OK it rewrites the body of SIP message (SDP). The problem is that it doesn't put in connection information (c) rtpproxy IP address; it puts its IP address. So clients send their RTP packet to SER instead to rtpproxy. How I can force SER to rewrite correctly SDP so that clients receive rtpproxy IP address in INVITE and 200 OK SIP message? Thanks in advance Alekos --------------------------------- L'email della prossima generazione? Puoi averla con la nuova Yahoo! Mail

1 0

[Serusers] register with ser
by zahra bahar 12 Feb '07

12 Feb '07

Hi. I installed SER.when I want to register with xlite in ser.xlite phone sends INVITE to ser.but ser doesn't send ok or any reply. what is the wrong. wen I run ser I have: Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.10.10.138 [10.10.10.138]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.10.10.138 [10.10.10.138]:5060 Aliases: tcp: Shahab2:5060 tcp: Shahab2.Sarv:5060 tcp: localhost:5060 udp: Shahab2:5060 udp: Shahab2.Sarv:5060 udp: localhost:5060 WARNING: no fork mode and more than one listen address found(will use only thethe first one) stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 112640 0(0) INFO: udp_init: SO_RCVBUF is finally 225280 0(0)WARNING: using only the first listen address (no fork) 2(6248) INFO: fifo process starting: 6248 2(6248) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(6248) WARNING: no fifo_db_url given - fifo DB commands disabled! --------------------------------- Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games.

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RE: [Serusers] register with ser
by Su Kerry 10 Feb '07

10 Feb '07

1 0

[Users] NAT problem for PRACK replies
by Jeff Williams 09 Feb '07

09 Feb '07

Hi, I seem to have a problem replies to PRACKs from NATed UAs. The UA is behind the NAT on a private IP listening on port 5060. This port gets mapped to 1350 on the external IP by the NAT device. The INVITE goes out fine and gets the contact updated for NAT by: if (client_nat_test("3")) { setflag(7); force_rport(); fix_contact(); }; This then receives a 183 reply to which it sends a PRACK. However, the 200 response to the PRACK gets sent to port 5060 on the external IP => which of course doesn't get to the UA. Is there something I'm missing in my config? I have attached a trace of the call setup. Jeff No. Time Source Destination Protocol Info 19424 6.838985 203.24.54.17 202.74.164.44 SIP/SDP Request: INVITE sip:0893680949@proxy.sipone.com;user=phone, with session description Frame 19424 (941 bytes on wire, 941 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:0893680949@proxy.sipone.com;user=phone SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK26f185af4ccc0d42 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone> Call-ID: 3478456911(a)192.168.10.30 CSeq: 1 INVITE Max-Forwards: 70 Contact: Jeff <sip:610089001@192.168.10.30:5060;user=phone;transport=udp> User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 300 Content-Length: 288 Content-Type: application/sdp Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 610089001 147590 147590 IN IP4 192.168.10.30 Session Name (s): Cisco 7912 SIP Call Connection Information (c): IN IP4 192.168.10.30 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 16384 RTP/AVP 0 18 8 101 Media Attribute (a): rtpmap:0 PCMU/8000/1 Media Attribute (a): rtpmap:18 G729/8000/1 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:8 PCMA/8000/1 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 No. Time Source Destination Protocol Info 19428 6.842541 202.74.164.44 203.24.54.17 SIP Status: 100 trying -- your call is important to us Frame 19428 (655 bytes on wire, 655 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: 1350 (1350) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK26f185af4ccc0d42;rport=1350;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone> Call-ID: 3478456911(a)192.168.10.30 CSeq: 1 INVITE Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11082 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:0893680949@proxy.sipone.com;user=phone out_uri=sip:61893680949@202.74.164.72:5060;user=phone via_cnt==1" No. Time Source Destination Protocol Info 19429 6.842579 202.74.164.44 202.74.164.72 SIP/SDP Request: INVITE sip:61893680949@202.74.164.72:5060;user=phone, with session description Frame 19429 (1148 bytes on wire, 1148 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 202.74.164.72 (202.74.164.72) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:61893680949@202.74.164.72:5060;user=phone SIP/2.0 Message Header Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU-> Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK7a7e.3cf74a67.0 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=1350;received=203.24.54.17;branch=z9hG4bK26f185af4ccc0d42 From: Jeff <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone> Call-ID: 3478456911(a)192.168.10.30 CSeq: 1 INVITE Max-Forwards: 69 Contact: Jeff <sip:610089001@203.24.54.17:1350;user=phone;transport=udp> User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 300 Content-Length: 288 Content-Type: application/sdp Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 610089001 147590 147590 IN IP4 192.168.10.30 Session Name (s): Cisco 7912 SIP Call Connection Information (c): IN IP4 202.74.164.44 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 63054 RTP/AVP 0 18 8 101 Media Attribute (a): rtpmap:0 PCMU/8000/1 Media Attribute (a): rtpmap:18 G729/8000/1 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:8 PCMA/8000/1 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 No. Time Source Destination Protocol Info 19450 6.860058 202.74.164.72 202.74.164.44 SIP Status: 100 Trying Frame 19450 (445 bytes on wire, 445 bytes captured) Ethernet II, Src: 00:12:44:c0:cc:1b, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 202.74.164.72 (202.74.164.72), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header From: "Jeff" <sip:0892209080@sipone.com:5060>;tag=1166274291 To: <sip:0893680949@202.74.164.72:5060;user=phone>;tag=669b638bbf9e0903fd8a2a66 CSeq: 1 INVITE Call-ID: 3478456911(a)192.168.10.30 Via: SIP/2.0/UDP 202.74.164.44:5060;branch=z9hG4bK7a7e.3cf74a67.0 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=1350;received=203.24.54.17;branch=z9hG4bK26f185af4ccc0d42 Content-Length: 0 No. Time Source Destination Protocol Info 21001 8.339800 202.74.164.72 202.74.164.44 SIP/SDP Status: 183 Session Progress, with session description Frame 21001 (1064 bytes on wire, 1064 bytes captured) Ethernet II, Src: 00:12:44:c0:cc:1b, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 202.74.164.72 (202.74.164.72), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 183 Session Progress Message Header Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK7a7e.3cf74a67.0 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=1350;received=203.24.54.17;branch=z9hG4bK26f185af4ccc0d42 From: "Jeff" <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:07 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Require: 100rel RSeq: 3067 Contact: <sip:610089001@202.74.164.72:5060;user=phone> Record-Route: <sip:202.74.164.72:5060;lr> Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU-> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 214 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): CiscoSystemsSIP-GW-UserAgent 4912 6655 IN IP4 202.74.164.72 Session Name (s): SIP Call Connection Information (c): IN IP4 202.74.164.72 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 27182 RTP/AVP 0 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 No. Time Source Destination Protocol Info 21003 8.341340 202.74.164.44 203.24.54.17 SIP/SDP Status: 183 Session Progress, with session description Frame 21003 (1016 bytes on wire, 1016 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: 1350 (1350) Session Initiation Protocol Status-Line: SIP/2.0 183 Session Progress Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;rport=1350;received=203.24.54.17;branch=z9hG4bK26f185af4ccc0d42 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:07 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Require: 100rel RSeq: 3067 Contact: <sip:610089001@202.74.164.72:5060;user=phone> Record-Route: <sip:202.74.164.72:5060;lr> Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU-> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 214 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): CiscoSystemsSIP-GW-UserAgent 4912 6655 IN IP4 202.74.164.72 Session Name (s): SIP Call Connection Information (c): IN IP4 202.74.164.44 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 63054 RTP/AVP 0 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 No. Time Source Destination Protocol Info 21044 8.397481 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 21044 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 21045 8.397734 202.74.164.44 202.74.164.72 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 21045 (725 bytes on wire, 725 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 202.74.164.72 (202.74.164.72) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291> Route: <sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK4a7e.215a51c5.0 Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 69 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 21060 8.419413 202.74.164.72 202.74.164.44 SIP Status: 200 OK Frame 21060 (489 bytes on wire, 489 bytes captured) Ethernet II, Src: 00:12:44:c0:cc:1b, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 202.74.164.72 (202.74.164.72), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK4a7e.215a51c5.0 Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: "Jeff" <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:08 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 PRACK Content-Length: 0 No. Time Source Destination Protocol Info 21061 8.419532 202.74.164.44 203.24.54.17 SIP Status: 200 OK Frame 21061 (441 bytes on wire, 441 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:08 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 PRACK Content-Length: 0 No. Time Source Destination Protocol Info 21070 8.429213 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 21070 (469 bytes on wire, 469 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 21611 8.897497 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 21611 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 21612 8.897734 202.74.164.44 203.24.54.17 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) Frame 21612 (649 bytes on wire, 649 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11082 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:610089001@202.74.164.72 out_uri=sip:610089001@202.74.164.72 via_cnt==1" No. Time Source Destination Protocol Info 21625 8.909606 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 21625 (590 bytes on wire, 590 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 22692 9.898092 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 22692 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 22693 9.898343 202.74.164.44 203.24.54.17 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) Frame 22693 (649 bytes on wire, 649 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11078 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:610089001@202.74.164.72 out_uri=sip:610089001@202.74.164.72 via_cnt==1" No. Time Source Destination Protocol Info 22702 9.912093 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 22702 (590 bytes on wire, 590 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 24851 11.899557 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 24851 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 24853 11.899808 202.74.164.44 203.24.54.17 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) Frame 24853 (649 bytes on wire, 649 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11081 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:610089001@202.74.164.72 out_uri=sip:610089001@202.74.164.72 via_cnt==1" No. Time Source Destination Protocol Info 24866 11.911720 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 24866 (590 bytes on wire, 590 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 27365 13.900358 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 27365 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 27366 13.900630 202.74.164.44 202.74.164.72 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 27366 (725 bytes on wire, 725 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 202.74.164.72 (202.74.164.72) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291> Route: <sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK4a7e.315a51c5.0 Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 69 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 27426 13.938284 202.74.164.72 202.74.164.44 SIP Status: 200 OK Frame 27426 (489 bytes on wire, 489 bytes captured) Ethernet II, Src: 00:12:44:c0:cc:1b, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 202.74.164.72 (202.74.164.72), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK4a7e.315a51c5.0 Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: "Jeff" <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:14 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 PRACK Content-Length: 0 No. Time Source Destination Protocol Info 27427 13.938422 202.74.164.44 203.24.54.17 SIP Status: 200 OK Frame 27427 (441 bytes on wire, 441 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:14 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 PRACK Content-Length: 0 No. Time Source Destination Protocol Info 27438 13.947350 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 27438 (469 bytes on wire, 469 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 30349 15.901399 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 30349 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 30350 15.901638 202.74.164.44 203.24.54.17 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) Frame 30350 (649 bytes on wire, 649 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11082 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:610089001@202.74.164.72 out_uri=sip:610089001@202.74.164.72 via_cnt==1" No. Time Source Destination Protocol Info 30359 15.913057 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 30359 (590 bytes on wire, 590 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol

4 4

[Users] avpops on reply_route
by Klaus Darilion 09 Feb '07

09 Feb '07

Hi! Once again I want to discuss the problem of avpops in reply_routes (e.g. http://www.openser.org/pipermail/users/2005-October/001281.html) I can bypass the syntax parser by putting the AVP-commands into a normal route block which is executed from the reply_route. As far as I understand this may cause problems as the AVP-ops are done without locking. But what may happen in worst case? Bad AVPs or a crash? thanks klaus btw: is there any other way to handle data from a reply_route ?

2 1

Re: [Serusers] failed to load mysql.so module
by Josef.Bajada＠go.com.mt 09 Feb '07

09 Feb '07

3 5

[Users] switch - pseudo vars
by Cseke Tamas 09 Feb '07

09 Feb '07

Hello, I'm using openser 1.1.0 I'd like to make a dinamic routing, the plan is radius set an avp, which contains the routeid, but my config is bad, and i don't know what can be the problem route($avp(i:20)); don't work so i tryied this way: switch($avp(i:20)) { case 0: route(0); break; case 1: route(1); break; case 2: route(2); break; } but this is bad too, can anybody give me an advice? the problem is in the line "switch($avp(i:20)) {" i guess, but switch can accept pseudo vars, so i don't kow what the problem caould be? Thanks any help, Tamas

3 5

[Users] using multiple DNS servers
by Will Quan 09 Feb '07

09 Feb '07

I have a primary and secondary dns server setup in resolv.conf, with a timeout option of 1 second.The idea is when the primary fails, my secondary server picks up the workload.When openser is running, and primary dns is down, I am seeing several re-transmissions of 200 OK (invite) to my originator *after* openser has finished processing ACK. I was thinking the final responses would stop reTX after the ACK was t_relayed. Have you seen such behavior? And better yet- Is there a solution for this?Thanks, Will

4 3

[Serusers] Radius accounting - User-Name parameter in the Start Acct-Status-Type
by tzieleniewski 09 Feb '07

09 Feb '07

Hi! Is it possible to configure acc_radius in such a way that it will send a User-Name parameter in the Accounting Packet because now it doesn't: rad_recv: Accounting-Request packet from host 127.0.0.1 port 32844, id=246, length=474 SER-Attr = "" Acct-Session-Id = "6012E956-D8D3-2367-979B-753579FE59FF(a)192.168.1.2" Sip-To-Tag = "572241288" SER-From = "tomix <sip:tomix@tezet.no-ip.org:7060>;tag=2020612254" SER-Flags = 8 SER-Original-Request-ID = "sip:hellboy@tezet.no-ip.org" Sip-Method = "INVITE" Sip-Cseq = 20886 Sip-Translated-Request-ID = "sip:hellboy@192.168.1.10:5060" Sip-Source-IP-Address = 192.168.1.2 Sip-From-Tag = "2020612254" SER-To = "<sip:hellboy@tezet.no-ip.org>;tag=572241288" SER-Digest-Username = "tomix" SER-Request-Timestamp = 1171023334 Calling-Station-Id = "sip:tomix@tezet.no-ip.org:7060" Sip-Source-Port = 7060 SER-Digest-Realm = "tezet.no-ip.org" Sip-Response-Code = 200 Called-Station-Id = "sip:hellboy@tezet.no-ip.org" SER-Response-Timestamp = 1171023341 Acct-Status-Type = Start Service-Type = IAPP-Register NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Cheers Tomasz

1 0

[Serusers] SER 0.10.99 (Carpo) - NAT call - long ringing after hang up
by Theodor Meier 09 Feb '07

09 Feb '07

hello, using my fritzbox fon with an account from Carpo, ser 0.10.99, to call a cisco 7960 behind the same fritzbox fon, that uses as well an Carpo account, an hang up, the called cisco 7960 continues ringing for a minute. (fritzbox - ser - (same) fritzbox - 7960) Vice versa the same. (7960 - fritzbox - ser - fritzbox) Not so when I use the 7960 with a sipgate account (7960 - sipgate - telefonica gateway - Carpo ser - fritzbox) to call the fritzbox. Is this a ser problem? Perhaps a problem, when the call-phone and the called-phone use the same IP? Or is this a configuration problem of ser? What can I debug to find where the problem is? ___________________________________________________________ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de

5 6

[Users] RTP fixes for mediaproxy
by Jeff Williams 09 Feb '07

09 Feb '07

Hi, I was having a few issues with mediaproxy when it came to re-invites and call setup where the SDP does not get sent with the first INVITE. I noticed this was due a couple couple of assumptions in mediaproxy which don't always hold: 1) That SDP is always set up first by the caller (i.e. in the INVITE) then updated by the called party (i.e. in the 200). This is not the case. The first SDP can be sent in the 200 response by the called party and the caller send their SDP in the ACK. 2) That RTP streams don't change once set up. This is wrong for reINVITEs. This would cause problems when a user agent sent a reINVITE to redirect RTP to on hold music for example - the RTP change would be ignored. I think this had an impact on some fax setups as well, but I haven't tested it. I found a patch to fix this by storing the streams in a hash by media type, but that is a limiting assumption also (maybe I want 2 audi streams?). The attached patch fixes this by changing the following: 1) This pathc allows a mediaproxy RTPSession to be set up by either the caller or called party. A 'caller' argument is passed to the RTPSession functions telling it if the caller or called party is doing the setup or update. 2) The official mediaproxy does not even look at SDP details for streams which are already set up. This patch lets every SDP packet update all of the RTP streams for that party (either caller or called). If you have issues with one-way-audio with media proxy after reINVITEs, have calls the first SDP from the called party or are just plain adventurous, please give it a go. It has been used in a production environment for a bit over a month now. The patch was made against mediaproxy 1.7.2, but will work against 1.8.0 as well since none of the rtphander.py code changed. Regards, Jeff

2 2

[Serusers] failed to load mysql.so module
by Josef.Bajada＠go.com.mt 08 Feb '07

08 Feb '07

Hi, I am having this problem with the mysql module on ser. 0(31212) ERROR: load_module: could not open module </opt/ser/modules/mysql/mysql.so>: /opt/ser/modules/mysql/mysql.so: undefined symbol: log 0(31212) parse error (30,13-14): failed to load module ERROR: bad config file (1 errors) I can confirm that mysql.so exists in that directory. I have mysql 5.0.27 installed. I have already tried to clean and make all from scratch but the problem remained. Any ideas how to solve this? Thanks a lot, Josef

3 2

[Users] set RURI with dispatcher module
by Klaus Darilion 08 Feb '07

08 Feb '07

Hi! I want to use the dispatcher module to use load-balancing/failover to gateways. Thus, I also want to change the domainpart of the RURI to be the IP address of the gateway - but dispatcher only sets the DURI. Is there an easy way to set the RURI/domain too or do I have to use: avp_write("$rU","$avp(i:666)"); ds_select_dst(...); avp_pushto("$ru","$du"); avp_pushto("$ru/username","$avp(i:666)"); Is there a more easy way? btw - I think above workaround would not work in failure_route as ds_next_dst adds a new branch but avp_pushto works against branches[0] - correct? thanks klaus

3 4

[Users] SIP Proxy - CAC
by Michel Bensoussan 08 Feb '07

08 Feb '07

Hello, I need to write a CAC (Call Admission Control) module for an 802.11 AP (Access Point). The idea is to use a SIP Proxy to monitoring bandwith utilization, for example according to codec, and allow or disallow new sessions, depending on resources. I'm thinking using OpenSER for this purpose. On a first stage I want to install a SIP Proxy on the AP (embedded with OS Linux) and on a second stage adding the CAC feature. There is many modules in OpenSER and I'm not sure which one I should use for a SIP Proxy. I'm learning about SIP but my knowledge is limited (till last week I never heard about it). Thank you for advising. Regards, Michel.

2 3

[Serusers] 0.9.x mysql.so: undefined symbol: log error
by Steve Blair 08 Feb '07

08 Feb '07

Hi: I keep getting the following error on a 0.9.x system. The latest 0.9.x version of SER was just download and built on this server but I continue to get this same error. I've check the mysql Makefile and it seems to be correct. What else could cause this error? ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: log Thanks,Steve

4 8

[Serusers] sequential hunting
by Steve Blair 08 Feb '07

08 Feb '07

Hello: I would like to implement sequential hunting. Hunting is when a call arrives for a given subscriber, no one answers the original number so the proxy populates the username portion of the r-uri with another number (from the database) and tries again. This process continues until all numbers in the database are exhausted. I have the "failover" working but the r-uri field is not updated on any interation though the failure_route. Does anyone have any ideas how to make the following work? Thanks,Steve # hunting (serial forking) if (avp_db_load("$ruri/username", "s:allow_hunt")) { if (avp_check("s:allow_hunt", "eq/y/i")) { avp_delete("s:allow_hunt"); xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Call Hunting Enabled for <%ru>\n"); avp_db_load("$ruri/username", "s:huntlist/sam"); xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Call Hunting Started to <%ru>\n"); t_on_failure("6"); t_relay(); break; }; }; # Iterate through extension list for subscribers with call hunting enabled failure_route[6] { xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Failure Block #6: HUNTING for <%ru> from <%fu> at <%is>\n"); if (avp_pushto("$ruri/username", "s:huntlist/g")) { append_branch(); avp_delete("s:huntlist"); t_on_failure("6"); t_relay(); }; }

5 12

[Users] RADIUS acc ok, RADIUS auth not
by Luca Corti 08 Feb '07

08 Feb '07

Hello, I'm having a hard time getting RADIUS auth to work at all. I've reached the point where I am debugging the network traffic with tshark. I can see RADIUS packets generated from the openser box for accounting, but no RADIUS authentication packets are sent. Also, it seems no response comes back from the RADIUS server even for accounting requests. radclient against my radius server returns me an Access-Reject at least and the tshark dump contains auth request and response. OpenSER 1.1.1 tells me: Feb 6 17:27:04 openser openser[22596]: rc_send_server: no reply from RADIUS server I suspect the issues lie in the radiusclient-ng layer. I'm using radiusclient-ng from Debian Etch (0.5.3-2), but I see many branches in CVS, like RELENG_0_5_5 which seems newer. thanks

2 6

[Users] uac_replace_from() not work for Linksys ATA
by Howard Tang 08 Feb '07

08 Feb '07

HI All, I have a problem with uac_replace_from() function not working for Linksys Unit. This is what i have in the route[1], if (search("From:.*<sip:900*")) { uac_replace_from("anonymous","sip:anonymous@x.x.x.x"); } I have 0900xx as internal sip account, then I have to remove the 0900xx callerid before i send the call to PSTN gateway. Because 0900xx is not a valid number, Mobile phone providers block the call because of that, as a result, I need to replace the callerid to anonymous. I have tested with many other ATA and X-Lite without problem. The problem only ocurrs when i use Linksys ata. I have monitored the SIP message, the first request will change to anonymous then back to the original username. Anyone have an idea on how i can fix this ? -- Howard Tang ICQ : 259083 MSN : howard615(a)hotmail.com

2 7

[Serusers] Digest Authentication in Ottendorf server
by raj gopal 08 Feb '07

08 Feb '07

Hi, I can not use www_challenge(realm,qop)function in ser.cfg file even though auth module is loaded. The readme file has documentation for it. When I use the following in config file and start SER, the following error is thrown if (!radius_www_authorize("")) { www_challenge("", "1"); // parse error break; }; parse error (197,49-50): unknown command, missing loadmodule? I am using Ottendorf release. How to do digest authentication with new release??** <http://ftp.iptel.org/pub/ser/daily-snapshots/unstable/> -Venkat

2 1

[Users] Angle bracket in Route header coming from Contact and strict router
by Marcello Lupo 08 Feb '07

08 Feb '07

Hi to all, little question about populating the Route Header field. I'm dealing with 2 CPE of different vendors (Patton and Netsynt) and an Openser in the middle. Patton in their SIP is not using angle bracket on SIP From,To and Contact headers. Netsynt is using strict routing method to build BYE messages. This is causing that when Netsynt is creating the BYE it is putting the proxy IP in the R-URI and the URI of Contact received from previuos packets from patton in the Route header to let it be processed by the proxy. In this way the Route header field constructed is without the <> enclosing the uri. Netsynt keep it from previous Contact without any manipulation on it. This is causing an error on the proxy parsing the name-addr field of the Route header field and the loose_route() fail. Seems that it is mandatory for Openser to get angle bracket in Route header fields. Now in the section 20 of RFC3261 is said that it is not a MUST that From,To and Contact header have to be enclosed in angle bracket. So i'm searching in the RFC where is said that if the Contact is without the angle bracket the UAC have to check if the Contact have the <> and if it don't have the <> it have to insert it before to out it on the Route (i searched the RFC2543 too). can someone give me clarification on this issue? Where is written the Route header field is mandatory to have <> enclosing the URI. I'm searching all of this because i have to give a good reason to each vendor (Netsynt and Patton) to modify their sip stack for this issue. I don't know who of the 2 should fix the problem. If Patton to add <> in each request or Netsynt to fix the Route field population. Thanks, Bye, Marcello

2 1

RE: [Serusers] xlite and ser
by Su Kerry 08 Feb '07

08 Feb '07

1 0

[Serusers] xlite and ser
by zahra bahar 08 Feb '07

08 Feb '07

Hi all. could any one send me all the things that I should do for having only a call between two xlite phones with ser server? I have installed ser and xlite on 10.10.10.138 and another xlite phone on 10.10.10.195. the xlite phone on 10.10.10.195 sends register massage to 10.10.10.138 but xlite phone that is on 10.10.10.138 doesn't send massage to ser. I didn't change any thing in ser only fork=no,log-stderror=yes.children=4. and I don't user sql. thanks. --------------------------------- Access over 1 million songs - Yahoo! Music Unlimited.

1 0

[Users] Corrupted 'To' after uac_restore_from
by Mahesh Paolini-Subramanya 08 Feb '07

08 Feb '07

I'm seeing this *very* weird error where uac_restore_from is ending up with a corrupted 'From' header. The weird thing is that it happens with only one end point (a Televantage SIP Trunk). With virtually *everything* else - and I am talking about tens of thousands of end points, spanning all sorts of sip devices, we are having *no* problems. The call is set up as follows -> the endpoint sends the INVITE, which is forwarded to a PSTN gateway The ACKs and the 200 OK all have their From/To headers appropriately rewritten The problem arises when the PSTN gateway initiates the BYE As you can see below, the 'To' header gets corrupted The *only* thing I can think of is that it has something to do with the structure of the Televantage tags (the '+' signs?). Any ideas? FYI, Openser 1.1.0 (latest CVS checkout), and modparam("uac","from_restore_mode","auto") modparam("uac","rr_store_param","aptelavsf") ---INVITE--- endpoint->proxy From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8d200003+32cc4d0e. To: <sip:17038677006@69.25.47.136>;tag=277e3aa2408f2ad73a234f5bbb99133b.c13b. ---INVITE--- proxy->gateway From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136> Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> --- 183 Session Progress--- gateway->proxy From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496. Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---183 Session Progress--- proxy->endpoint From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496 ---200 OK--- gateway->proxy From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496 Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---200 OK--- proxy->endpoint From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496 Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---BYE--- gateway->proxy To: "Membersevice Queue"<sip:6462174165@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---BYE--- proxy->endpoint To: "Membersevice Queue"<sip:user100.cru>6'(!.l asr}XV.R\[>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

2 1

[Users] qop="auth" with uac_auth
by Andreas Heise 08 Feb '07

08 Feb '07

Hello, since a lot of providers has change there authentication to qop="auth" the uac_auth(); function of the uac module can't use anymore. by google I found a lot of requests for qop with uac_auth and also the official feature request 1345887. Is a target date known for a solution, maybe it's possible with 1.2.0? Feature Request [ 1345887 ] Implement qop functionality in uac module p.s.: I know it's on the road map, but it is needed so often...... thanks. Andreas

4 10

[Users] Reg:NAT Routing
by venkid 08 Feb '07

08 Feb '07

hai, can any body help me to test audio call between two different private IPs. am using X-lite as client .am not able to get voice between two private ips and able to get voice with public IP .i have run media proxy saparatly. please find attached my openser.cfg file too. Thanks and Regards venkatesh.d debug=9 fork=yes log_stderror=yes listen=192.168.0.25 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:openser@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/presence.so" modparam("auth_db|domain|uri_db|usrloc","db_url","mysql://openser:openser@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("presence", "db_url","mysql://openser:openser@localhost/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'a') modparam("presence", "totag_avpid", 987) modparam("presence", "lock_set_size", 8) modparam("presence", "expires_offset", 10) modparam("presence", "force_active", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE log("setting nat as yes *****************"); record_route_preset("192.168.0.25:5060;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 ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); exit; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { log("setting falg 6 here **********"); setflag(6); use_media_proxy(); }; }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); exit; }; if (method=="ACK") { route(1); exit; } else if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; 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"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } 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(); }; }

1 0

Re: [Serusers] Problemswith compiling ser 0.10.99
by Dmitry Isakbayev 07 Feb '07

07 Feb '07

The OSP toolkit project has been moved from sipfoundry to sourceforge - http://sourceforge.net/projects/osp-toolkit Regards, Dmitry > Date: Mon, 08 Jan 2007 10:10:47 -0500 > From: Dmitry Isakbayev <dmitry(a)transnexus.com> > Subject: Re: [Serusers] Problemswith compiling ser 0.10.99 > To: "tzieleniewski" <tzieleniewski(a)o2.pl> > Cc: "'Support@Transnexus. Com'" <support(a)transnexus.com>, > serusers(a)lists.iptel.org > Message-ID: <45A25EF7.7070502(a)transnexus.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi, > > The OSP toolkit project is being moved from sipfoundry to sourceforge. > It should be available shortly. Meanwhile, the latest snapshot of the > toolkit can be downloaded from > > http://www.transnexus.com/OSP%20Toolkit/OSP%20Toolkit.htm > > Regards, > Dmitry > > Date: Mon, 08 Jan 2007 11:25:24 +0100 > > From: "tzieleniewski" <tzieleniewski(a)o2.pl> > > Subject: [Serusers] Problemswith compiling ser 0.10.99 > > To: serusers(a)lists.iptel.org > > Message-ID: <54c54dce.20e2e287.45a21c14.f0624(a)o2.pl> > > Content-Type: text/plain; charset="utf-8" > > > > Hi All!! > > I am trying to compile the lattest cvs version of ser but I get errors > > which I cann't handle myself so i kindly ask for your help:) > > What I want to achive basically is the ser instance with presence > > support. > > I am compiling ser in the following way: > > make group_include="standard standard-dep stable" all > > I read that osp requires osptoolkit i tried to get it from sipfoundry > > but i couldn't find any source of that. > > I also cann't fight problems with tls i didn't includethe experimental > > modules(tls) but still I get some errors.

2 1

[Serusers] SIP Proxy - CAC
by Michel Bensoussan 07 Feb '07

07 Feb '07

Hello, I need to write a CAC (Call Admission Control) module for an 802.11 AP (Access Point). The idea is to use a SIP Proxy to monitoring bandwith utilization, for example according to codec, and allow or disallow new sessions, depending on resources. I'm thinking using SER (or maybe OpenSER) for this purpose. On a first stage I want to install a SIP Proxy on the AP (embedded with OS Linux) and on a second stage adding the CAC feature. There is many modules in SER and I'm not sure which one I should use for a SIP Proxy. I'm learning about SIP but my knowledge is limited (till last week I never heard about it). Thank you for advising. Regards, Michel.

1 0

[Serusers] Newbie ser user, moving from asterisk
by Antonio José dos Santos Brandão 07 Feb '07

07 Feb '07

Hi, I'm starting with ser and would like to know if it's possible no make the following: Today i use asterisk with sip users registered to it and a DeadAgi (a2billing) to manage calls and prepaid/postpaid accounts. Works fine. Since i have too many features and customizations i can't abone it. The problem is that many users are behind nat and have to canreinvite=no, forcing the rtp media to passthru asterisk. I would like to implement media_proxy in this scenario to balance the rtp's bandwith with another hosts. If there were some kind of media_proxy for asterisk would be great, but there isn't. So, i wonder: can I put ser in this, after or before asterisk, making possible to user media_proxy without change so much in asterisk setup? tks, -- Antonio J. S. Brandão

2 1

Re: [Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
by Sahria Hao 07 Feb '07

07 Feb '07

Hi Edson. I want to know that why my Cisco AS 5300 didn't send BYE for SER...? Maybe... I doubt that maybe my 5300 have only dial-peer 6000 voice "POTS" configure for outgoing PSTN call. In case of PSTN incoming call have no problem about sending BYE for SER, Because it is apply dial-peer voice 5000 "VOIP" confiure as follows: [SER] <- [5300 (VOIP dial-peer)] <- [PSTN] So I'll try to re-configure my 5300 dial-peer, or please give me a hint If anyone have some way to solve this problem. Thanks, Sahria 2007/2/6, Edson <4lists(a)gmail.com>: > > I have this same behaviour, but never give it great importance, since we > didn't bill incomming calls… > > But it would be great to know if it's because of a misconfiguration or a > bug… but we notice that many ports become unavaliable (blocked) over time. > To release we programmed a reboot every day on 3AM…J > > Even with 'ngrep' the BYE, when PSTN side disconnects, didn't show up… > > Edson. > > ------------------------------ > > *From:* serusers-bounces(a)lists.iptel.org [mailto: > serusers-bounces(a)lists.iptel.org] *On Behalf Of *Sahria Hao > *Sent:* segunda-feira, 5 de fevereiro de 2007 08:36 > *To:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] Cisco AS 5300 can't send BYE for SER... It's > bug? > > Hi Greger, > > And I'm very sorry for my poor exposition. > > >>Do you get an error on the 5300? > > No, my 5300 works well and there's no error. > > >> Is it sent, but never reaches SER? > > No, when I finished call by PSTN side, 5300 didn't send BYE for SER. > >>Does SER receive, but does not recognize it? > > SER didn't receive a message from 5300 entirely. > > I think that when I finished this call, 5300 must send a BYE message for > SER... but didn't send it. > 2007/2/5, Greger V. Teigre <greger(a)teigre.com>: > > 09. [Cisco] can't send BYE for SER *****why??***** > > What does that mean?! Do you get an error on the 5300? Is it sent, but > never reaches SER? > Does SER receive, but does not recognize it? > g-) > > Sho Aihara wrote: > > Hi all. > > I have a problem for the following scenario. > When I make a call for PSTN and on hook by PSTN side, > Cisco As can't send BYE for SER. > > 01. [UA via Asterisk] dialing "08022223333" -> [SER] > 02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") > -> [Cisco] > 03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from > "033308022223333" to "008022223333" > 04. [Cisco] process an outbound call to "008022223333" -> [e.g. Mobile] > 05. [e.g. Mobile] Catch call > 06. [SER] log CDR start > 07. [Cisco] talking > 08. [e.g. Mobile] On hook and call disconnect > 09. [Cisco] can't send BYE for SER *****why??***** > 10. [UA via Asterisk] On hook > 11. [UA via Asterisk] Send BYE for SER > 12. [SER] log CDR End [Cisco] Call finished > > But another scenario, if make a call from PSTN to Asterisk and > on hook by PSTN side, Cisco As send BYE to SER. > > 01. [e.g. Mobile] dialing "0377771111(Asterisk user number)" > 02. [Cisco] receive "77771111" call number > 03. [Cisco] dial-peer voice 5000 voip, session target ipv4: > my.ser.ip.address -> [SER] > 04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk] > 05. [UA via Asterisk] Catch call > 06. [SER] log CDR start > 07. [UA via Asterisk] talking > 08. [e.g. Mobile] On hook and call disconnect > 09. [Cisco] Send BYE to SER > 10. [SER] log CDR End [Cisco] Call finished > 11. [UA via Asterisk] receive BYE from SER > > And sorry for my diffucult example. > > Why Cisco AS 5300 can't send BYE to SER > When PSTN call is disconnected by PSTN side? > > My ser.cfg as follows: > > # > -------------------------------------------------------------------------- > # global configuration parameters > # > -------------------------------------------------------------------------- > fork=no > log_stderror=yes > check_via=no > dns=no > rev_dns=no > listen=my.ser.ip.address > port=5060 > fifo="/tmp/ser_fifo" > fifo_db_url="mysql://ser:heslo@localhost/ser" > > # > -------------------------------------------------------------------------- > # 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/avpops.so" > loadmodule "/usr/local/lib/ser/modules/permissions.so" > loadmodule "/usr/local/lib/ser/modules/acc.so" > loadmodule "/usr/local/lib/ser/modules/exec.so" > > # > -------------------------------------------------------------------------- > # setting module-specific parameters > # > -------------------------------------------------------------------------- > modparam("usrloc", "db_mode", 2) > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("rr", "enable_full_lr", 1) > modparam("usrloc", "db_url", " mysql://ser:heslo@localhost/ser") > modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("permissions", "db_url", "mysql://ser:heslo@localhost /ser") > modparam("tm", "fr_inv_timer", 27) > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("acc", "db_flag", 2) > modparam("acc", "db_missed_flag", 3) > > # > -------------------------------------------------------------------------- > # route pattern > # > -------------------------------------------------------------------------- > route { > > 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; > }; > > record_route(); > > if (loose_route()) { > if (method=="ACK") { > acc_db_request("01:CallStart\n", "acc"); > }; > if (method=="BYE" || method=="CANCEL") { > acc_db_request("02:CallEnd\n", "acc"); > }; > t_relay(); > break; > }; > > if (uri==myself) { > if (method=="REGISTER") { > if (!www_authorize("", "subscriber")) { > www_challenge("", "0"); > break; > }; > save("location"); > break; > }; > > if (search("^(f|From): .*(a)(my\.cisco\.ip\.address<.*(a)%28my%5C.cisco%5C.ip%5C.address>)")) > { > #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111 > rewritehost("my.asterisk.ip.address "); > }; > > lookup("aliases"); > > if (!lookup("location")) { > if (method=="INVITE" && !search("^(f|From): > .*(a)(my\.cisco\.ip\.address <.*(a)%28my%5C.cisco%5C.ip%5C.address>)")) { > if (!proxy_authorize("", "subscriber")) { > proxy_challenge("", "0"); > break; > }; > if (uri=~"^sip:0[0-9]{10}@") { > # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333 > prefix("0333"); > rewritehostport("my.cisco.ip.address:5060"); > avp_write("i:45", "inv_timeout"); > } else { > sl_send_reply("404", "Not Found"); > break; > }; > consume_credentials(); > }; > }; > > }; > > if (!t_relay()) { > sl_reply_error(); > }; > > } > > And my Cisco AS 5300 config as follows: > > voice call send-alert > voice rtp send-recv > > voice service pots > fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco > > voice service voip > fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco > sip > min-se 60 > > translation-rule 50 > Rule 0 0333 0 > Rule 1 ^7777 037777 > > voice class codec 2 > codec preference 1 g711ulaw > codec preference 2 g711alaw > > dial-peer voice 5000 voip > tone ringback alert-no-PI > description ser-asterisk-cisco-test > huntstop > destination-pattern 77771111$ > translate-outgoing called 50 > voice-class codec 2 > session protocol sipv2 > session target ipv4:my.ser.ip.address > dtmf-relay rtp-nte > max-conn 1 > > dial-peer voice 6000 pots > application session > max-conn 2 > destination-pattern 0333T > progress_ind alert enable 8 > translate-outgoing called 50 > port 0:D > > Thanks, > Sahria > > > ------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > -- > ---------- > shosuke > msn : anseie(a)hotmail.co.jp > email : sahria.hao(a)gmail.com >

4 6

[Users] distinguish between fr_timer and fr_inv_timer timeout
by Klaus Darilion 07 Feb '07

07 Feb '07

Hi! If a failure route is executed - how can I distinguish between fr_timer timeout (a broken gateway) and fr_inv_timer timeout (nobdy took up the phone)? thanks klaus

3 4

[Users] 180 Ringing / 183 Session Progress
by Sam Lee 07 Feb '07

07 Feb '07

Hey Guys, Is it mandatory for SIP servers to generate to UA 180 Ringing to inform them that the called party is now ringing ? The reason i'm asking is because I am seeing strange results using different UAs. Using SIP clients to call out , instead of seeing 180 Ringing , i see 183 session progress even before call was picked up. On some sip client , ring back can be heard. On some others , ring back cannot be heard. Is there any reason to suspect that these UAs needs to receive 180 Ringing before generating any ringback ? Regards, Sam

4 6

[Users] OpenSER v1.0.1 to v1.1.1
by Max Gregorian 06 Feb '07

06 Feb '07

Hi, with regards to the release of v.1.1.1 I have a few questions. Firstly, what is to become of v1.0.1? Are you recommending all users with 1.0.1 also upgrade directly to v.1.1.1? Secondly, does 1.0.1 suffer from most (if not all) the same bugs as 1.1.0? I am particularly interested in any documented issues with handling response ACKs for 4xx class errors as I am currently experiencing a few issues with these, and I noticed in the changelog for 1.1.1 that a bug with ACK matching for non-200 class responses has recently been fixed. I ask this I noticed you are gradually phasing out all documentation to do with 1.0.1 on the openser website. As such, is 1.0.1 to be deprecated?

2 3

[Serusers] Has anyone got Jabber module to work w/SER
by SER LIST 06 Feb '07

06 Feb '07

Subject says it all. I am trying to make this work. I have installed JABBERD2 with a PyMSNT gateway. I can use a jabber cliient (exodus) and connect to Jabberd server and communicate w/ other Jabber users and MSN users. I now want to forward IM messages from Eyebeam UA to Jabber users and MSN users. I followed all instructions. When the message gets forwarded to SER and it executes the "xjab" function, I get an error saying "Invalid Content-Type". Any pointers will be highly appreciated. _________________________________________________________________ Buy what you want when you want it on Sympatico / MSN Shopping http://shopping.sympatico.msn.ca/content/shp/?ctId=2,ptnrid=176,ptnrdata=08…

1 0

[Users] avp_db_load question
by Klaus Darilion 06 Feb '07

06 Feb '07

Hi! What are the return values of avp_db_load? E.g. is it FALSE on problems accessing the DB or FALSE if it did not found any AVP to load? I couldn't find this documented somewhere. regards klaus -- Klaus Darilion nic.at

2 1

[Serusers] Anyone using CDRTool ?
by Lokesh Kumar 06 Feb '07

06 Feb '07

Hello All, Is there anyone using CDRTool in conjugation with ser, asterisk, radius setup. Can anyone give me a hand to setup it properly? Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.411 / Virus Database: 268.17.27/671 - Release Date: 2/5/2007

1 0

[Serusers] Request for Daniel-Constantin Mierla
by Giovanni Chirico 06 Feb '07

06 Feb '07

Hi Daniel, Im trying to use the dispatcher module. Im using the call forwarding ser.cfg file of the SER Getting Started <http://www.iptel.org/ser/doc/gettingstarted> pack. I have inserted the function ds_select_dst("1", "0"); In the main route() and the function forward(uri:host, uri:port); in the PSTN Handler, route(5). How I can made the failure route, so if a gateway is down another gateway is load? In attachment there are my ser.cfg and dispatcher.list files. Please help me :-( Bye, Giò

2 3

[Serusers] automated response
by Myron Groening 06 Feb '07

06 Feb '07

Vacation Alert !! I will be out of the office, and will not be checking my message till FEB. 26 2007 Thanks Myron Groening

1 0

[Serusers] new version of radiusclient-ng
by Jan Janak 06 Feb '07

06 Feb '07

Hello, I just released a new version of radiusclient-ng library. The new version, 0.5.4, fixes problems on 64-bit architectures. You can download the new release from http://ftp.iptel.org/pub/radiusclient-ng/radiusclient-ng-0.5.4.tar.gz Credits go to Peter Nixon. Jan.

2 4

[Users] ERROR: module version mismatch for,/usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls,(i386/freebsd); module: openser 1.1.1-notls (i386/freebsd)
by Hansi Lao 06 Feb '07

06 Feb '07

Hello All, I have installed openser-1.1.1 with tls on my box. However, when I try to run openser. It returns me this error: ERROR: module version mismatch for /usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls (i386/freebsd); module: openser 1.1.1-notls (i386/freebsd). Would someone be kind enough to help me how to compile a mysql.so which supports tls? Thank you.

2 4

[Users] migrating openser v1.1.x to v1.2.x
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Hello, since we got in testing phase for 1.2.0 release, I think is better to start early a document with migration guidelines. Issues encountered during upgrade should be gathered together in the next page, to share and help each others: http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x It is just the skeleton and few stuff there. Hopefully the content will grow quickly. Everybody can contribute. Cheers, Daniel

1 0

[Users] How can i make my OpenSER server with presence module work properly with the db_text database?
by Kc Ayuba 05 Feb '07

05 Feb '07

Hi, I'm trying to use db_text database with the presence module of OpenSER(1.2.x ). User agent is X-Lite(3.0). When I try to send a REGISTER message to OpenSER, OpenSER still works. But if i sent a SUBSCRIBE or a PUBLISH message, then openser crashes. If i use the MySQL database, then everything works just fine. But i prefer to use the db_text database. Here is the core file that was generated when OpenSER crashed. Core generated: Program terminated with signal 11, Segmentation fault. #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); (gdb) bt #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 #1 0x080a3f6c in timer_ticker () at timer.c:141 #2 0x0806b51f in main_loop () at main.c:724 #3 0x0806d43b in main (argc=3, argv=0xaff0a2e4) at main.c:1438 (gdb) frame 0 #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); (gdb) list 605 606 subs.from_domain.s = row_vals[from_domain_col].val.str_val.s; 607 subs.from_domain.len = strlen(row_vals[from_domain_col].val.str_val.s); 608 609 subs.event_id.s = row_vals[event_id_col].val.str_val.s; 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); 611 612 subs.to_tag.s = row_vals[to_tag_col].val.str_val.s; 613 subs.to_tag.len = strlen(row_vals[to_tag_col].val.str_val.s); 614 (gdb) 615 subs.from_tag.s = row_vals[from_tag_col].val.str_val.s; 616 subs.from_tag.len = strlen(row_vals[from_tag_col].val.str_val.s); 617 618 subs.callid.s = row_vals[callid_col].val.str_val.s; 619 subs.callid.len = strlen(row_vals[callid_col].val.str_val.s); 620 621 subs.contact.s = row_vals[contact_col].val.str_val.s; 622 subs.contact.len = strlen(row_vals[contact_col].val.str_val.s); 623 624 subs.cseq = row_vals[cseq_col].val.int_val; I really hope that someone can help me with this problem. Thanks in advance. Kind regards, Kc

4 9

RE: [Users] Asterisk OpenSER and Exchange 2007 Unified Messaging
by Jon Webster 05 Feb '07

05 Feb '07

The significant difference of debug from a good call and openser is next. Below that are the full debug logs of both the openser and goodproxy call with an ngrep of openser's call in between. significant differences ======================== Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:30660 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/openser-082966e8 answered SIP/jonlaptop-0828e888 -- Attempting native bridge of SIP/jonlaptop-0828e888 and SIP/openser-082966e8 == Spawn extension (local, 8886000, 2) exited non-zero on 'SIP/jonlaptop-0828e888' Feb 5 09:38:58 WARNING[28172]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 59f0033f3761cbf949fd42714b5d2b8f@asterisk for seqno 103 (Non-critical Request) pbx*CLI> VERSES --- (11 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:47688 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:goodproxy:5060> set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set destination to goodproxy, port 5060 Transmitting (no NAT) to goodproxy:5060: asterisk debug of openser call =========== --- (9 headers 0 lines) --- pbx*CLI> <-- SIP read from openser:5060: SIP/2.0 180 Ringing FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078 TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f CSEQ: 102 INVITE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 70 VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- -- Called openser/8886000 -- SIP/openser-082966e8 is ringing pbx*CLI> <-- SIP read from openser:5060: SIP/2.0 200 OK FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078 TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f CSEQ: 102 INVITE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 70 VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060 CONTACT: <sip:swordfish:5065;transport=Tcp;maddr=exchangeUM> CONTENT-LENGTH: 197 CONTENT-TYPE: application/sdp SERVER: RTCC/2.0.6017.0 v=0 o=- 0 0 IN IP4 exchangeUM s=Microsoft Exchange Speech Engine c=IN IP4 exchangeUM t=0 0 m=audio 30660 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:30660 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/openser-082966e8 answered SIP/jonlaptop-0828e888 -- Attempting native bridge of SIP/jonlaptop-0828e888 and SIP/openser-082966e8 == Spawn extension (local, 8886000, 2) exited non-zero on 'SIP/jonlaptop-0828e888' Feb 5 09:38:58 WARNING[28172]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 59f0033f3761cbf949fd42714b5d2b8f@asterisk for seqno 103 (Non-critical Request) pbx*CLI> <-- SIP read from openser:5060: BYE sip:3149@asterisk SIP/2.0 Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f> Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f> FROM: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f TO: <sip:3149@asterisk>;tag=as4aa53078 CSEQ: 1 BYE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 69 Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825 VIA: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65 CONTENT-LENGTH: 0 USER-AGENT: RTCC/2.0.6017.0 P-hint: outbound --- (13 headers 0 lines) --- Sending to openser : 5060 (non-NAT) Transmitting (no NAT) to openser:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825;received=openser Via: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65 Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f> Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f> From: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f To: <sip:3149@asterisk>;tag=as4aa53078 Call-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:3149@asterisk> Content-Length: 0 ngrep of above call ====================== # U +0.000616 openser:5060 -> asterisk:5060 SIP/2.0 180 Ringing. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078. TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. CSEQ: 102 INVITE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # U +0.067855 openser:5060 -> asterisk:5060 SIP/2.0 200 OK. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078. TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. CSEQ: 102 INVITE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060. CONTACT: <sip:swordfish:5065;transport=Tcp;maddr=exchangeUM>. CONTENT-LENGTH: 197. CONTENT-TYPE: application/sdp. SERVER: RTCC/2.0.6017.0. . v=0. o=- 0 0 IN IP4 exchangeUM. s=Microsoft Exchange Speech Engine. c=IN IP4 exchangeUM. t=0 0. m=audio 30660 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. # U +32.353338 openser:5060 -> asterisk:5060 BYE sip:3149@asterisk SIP/2.0. Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>. Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>. FROM: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. TO: <sip:3149@asterisk>;tag=as4aa53078. CSEQ: 1 BYE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 69. Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825. VIA: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65. CONTENT-LENGTH: 0. USER-AGENT: RTCC/2.0.6017.0. P-hint: outbound. . # U +0.000734 asterisk:5060 -> openser:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825;received=openser. Via: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65. Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>. Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>. From: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. To: <sip:3149@asterisk>;tag=as4aa53078. Call-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:3149@asterisk>. Content-Length: 0. . good call from working proxy ======== We're at asterisk port 17508 Video is at asterisk port 11130 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to goodproxy:5060: INVITE sip:6000@goodproxy SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK4af21ea7;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy> Contact: <sip:3149@asterisk> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 05 Feb 2007 14:55:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 216 v=0 o=root 381 381 IN IP4 asterisk s=session c=IN IP4 asterisk t=0 0 m=audio 17508 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called exchange12/6000 pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK4af21ea7;rport;received=goodproxy From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 INVITE User-Agent: M-Networks USR/1.0 Allow: INVITE, INFO, ACK, CANCEL, BYE, NOTIFY, BENOTIFY, SUBSCRIBE Content-Length: 0 --- (9 headers 0 lines) --- pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK4af21ea7;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;epid=BD-70-82-06-F9;tag=f51839ab98 CSEQ: 102 INVITE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- -- SIP/exchange12-082b8e18 is ringing pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK4af21ea7;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;epid=BD-70-82-06-F9;tag=f51839ab98 CSEQ: 102 INVITE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTACT: <sip:goodproxy:5060> CONTENT-LENGTH: 197 CONTENT-TYPE: application/sdp SERVER: RTCC/2.0.6017.0 v=0 o=- 0 0 IN IP4 exchangeUM s=Microsoft Exchange Speech Engine c=IN IP4 exchangeUM t=0 0 m=audio 47688 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (11 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:47688 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:goodproxy:5060> set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set destination to goodproxy, port 5060 Transmitting (no NAT) to goodproxy:5060: ACK sip:goodproxy:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK6c3f8b53;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy>;tag=f51839ab98 Contact: <sip:3149@asterisk> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/exchange12-082b8e18 answered SIP/jonlaptop-082b0fb8 -- Attempting native bridge of SIP/jonlaptop-082b0fb8 and SIP/exchange12-082b8e18 Scheduling destruction of call '109c82647d6340be2a5ab19e7eb18a14@asterisk' in 32000 ms set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set de5060 SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK47d7f855;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy>;tag=f51839ab98 Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (local, 7776000, 3) exited non-zero on 'SIP/jonlaptop-082b0fb8' pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK47d7f855;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;tag=f51839ab98;epid=BD-70-82-06-F9 CSEQ: 103 BYE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- Destroying call '109c82647d6340be2a5ab19e7eb18a14@asterisk' pbx*CLI>

1 0

[Users] CVS frozen - testing phase of 1.2.0
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Hello, starting with today the development for 1.2.0 is closed. No new features should be added until the release. The next month will be dedicated to testing, integration of submitted patches and fixing reported bugs. We have a lot of new code to test, your help is very much appreciated. I want to list the major changes you should focus on while testing: - usage of pseudo-variables and transformations is script (http://openser.org/dokuwiki/doku.php/transformations:devel) - TM timers (millisecond accuracy) - IP blacklists - DNS failover - automatic error handling - flags system (script flags, message flags, branch flags) - usrloc enhancements (licking system refurbished, database content optimization via DB fetch support) - new management interface (mi_fifo, mi_xmlrpc) - SIP Servlet Java Application server interoperability (seas module) - PERL AGI (perl module) - SNMP support via snmpstats module - Presence support (modules presence, pua, pua_usrloc, pua_mi) - xmpp IM gateway (xmpp module) - sip session timer support (module sst) - accounting clean up - federation based peering via domainpolicy module If I forgot something, please contribute. In addition, great help would be with creating 1.2.0 variants of documentation and dokuwiki pages (where is the case), testing the tools (openserctl, DB creation scripts) and developing scripts to migrate database from 1.1.x to 1.2.0. To keep track of discovered issues, please register them on the tracker -- it this way you make sure it is not lost in mail threads: http://sourceforge.net/tracker/?group_id=139143&atid=743020 Have nice testing sessions! Your feedback is always helpful! Cheers, Daniel

2 3

[Serusers] DB disconnects with MySQL 5.0.27
by sip 05 Feb '07

05 Feb '07

Something I've never run into before when I was using MySQL 4.x, but now crops up on MySQL 5.0.27... I get these errors: Feb 5 07:30:40 qabalah ser[26765]: submit_query(): MySQL server has gone away Feb 5 07:30:40 qabalah ser[26765]: db_update: Error while submitting query Feb 5 07:30:40 qabalah ser[26765]: db_upd_ucontact(): Error while updating database Feb 5 07:30:40 qabalah ser[26765]: wb_timer(): Error while updating contact in db I assume there's some sort of timeout of the SER connection to mysql? Is there a standard way to rectify this? Is this something new in MySQL 5? N.

1 1

[Users] Socket Errors
by Martin Burns 05 Feb '07

05 Feb '07

We have over 4 proxies running openser-1.1.0-notls. Recently we started to see the following error in the logs of one of the proxies: Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: ERROR: udp_send: sendto(sock,0xb3c35898,594,0,0xb3c257f4,16): Invalid argument(22) Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: msg_send: ERROR: udp_send failed Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: ERROR:tm:t_forward_nonack: sending request failed Now the errors suggests a binding to localhost, however our openser.cfg has the following: listen=eth0:5060 After a brief search we came across a couple of over people having a similar problem. Once of the suggested fixes was to add a mhomed=1 line to the cfg. Well we tried this (even though we are not multi homed) and the error changed to: Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR: get_out_socket: connect failed: Invalid argument Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR: get_out_socket: no socket found Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR:tm:add_uac: can't fwd to af 2, proto 1 (no corresponding listening socket) Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR:tm:t_forward_nonack: failure to add branches Does anyone have any suggestions or ideas as to what is causing this error. Thanks mburns

2 7

[Users] dbtext script
by Ovidiu Sas 05 Feb '07

05 Feb '07

Hi, Since dbtext is part of the standard, my suggestion would be to ship/install the sc.dbtext script along with the openserctl script. We can install it as openser_dbtext_ctl and in openserctl we can make a reference to it if no DBENGINE is provided: see the following patch for openserct (openser-v1.1.1): diff -Naurp sip-server/scripts/openserctl openser/scripts/openserctl --- sip-server/scripts/openserctl 2006-09-15 08:41:42.000000000 -0400 +++ openser/scripts/openserctl 2007-01-17 09:39:09.000000000 -0500 @@ -126,6 +126,8 @@ do $f done echo + echo "Note: Use openser_dbtext_ctl for DBTEXT operations!" + echo } Regards, Ovidiu Sas

2 3

[Users] auth_radius.so - ERROR: load_module: could not open module
by Luca Corti 05 Feb '07

05 Feb '07

Hello, I've built openser 1.1.1 from source with radius support. I installed radiusclient-ng. When I execute openser with to loadmodule auth_radius.so in my configuration I get: # openser 0(20268) ERROR: load_module: could not open module </usr/local/lib/openser/modules/auth_radius.so loadmodule >: /usr/local/lib/openser/modules/auth_radius.so loadmodule : cannot open shared object file: No such file or directory 0(20268) parse error (60,13-14): failed to load module 0(20268) parse error (60,14-15): syntax error 0(20268) parse error (60,14-15): ERROR: bad config file (3 errors) But the module is there. # ls -l /usr/local/lib/openser/modules/auth_radius.so -rwxr-xr-x 1 root staff 61369 Feb 5 10:33 /usr/local/lib/openser/modules/auth_radius.so This is really strange. ciao Luca

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Re: [Users] In NAT, I didn't see the ACK packet after the Invited the Callee
by Bogdan-Andrei Iancu 05 Feb '07

05 Feb '07

have you checked this on the x-lite machine? regards, bogdan raviprakash sunkara wrote: > it didn't see any Ack packet is sending to SIP server, > > > On 2/5/07, *Bogdan-Andrei Iancu* <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > I would say the X-lite does not obey the RR header - please try to get > an trace from the X-lite machine to see where the ACK is sent. > > regards, > bogdan > > raviprakash sunkara wrote: > > > Hi BogDan, > > > > Yes, I see Record-Route header in the 200 ok....... > > Please can u check the ngrepped of the X-lite > > with Hard phones iit is fine ........... > > Below one is ngrep from X-lite users, > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > www.hyperion-tech.com <http://www.hyperion-tech.com>

1 0

[Users] default send-socket
by Klaus Darilion 05 Feb '07

05 Feb '07

Hi! What is the default send socket? Is it the first listen=..... statement in openser.cfg? Or can I manually set the default sendsocket? regards klaus

2 1

[Serusers] Re: load mysql.so failed
by Su Kerry 05 Feb '07

05 Feb '07

2 1

[Users] mediaproxy MySQL server has gone away error
by James Holden 05 Feb '07

05 Feb '07

Hi all, I'm having a problem with the mediaproxy rtp proxy. I'm using to do accounting into a MySQL database. After a period of inactivity, it loses it's connection to the MySQL server and the update queries begin to fail. Openser stays connected, however. The error in the logs is: error: accounting failed. MySQL error: (2006, 'MySQL server has gone away') I'm running: mediaproxy 1.8.0 openser-1.1.0 (using the RHEL3 rpm from the openser website) openser-mysql-1.1.0 (again using the openser RHEL3 rpm) Python-2.4.4 MySQL-python-1.2.1_p2 My Python skills aren't up to much. So far, I've tried added the following line into /usr/local/mediaproxy/modules/mysql.py just before the query call, to try and re-establish the connection: def query(self, query): self._query = query for c in range(0, 3): try: self.conn.db.ping() <<<<<<---- added by me return self.cursor.execute(query) except MySQLdb.OperationalError, why: if why[0] not in (CR.SERVER_GONE_ERROR, CR.SERVER_LOST): raise raise ... but it hasn't helped. Has anyone seen this before? Best regards, James -- James Holden AQ Limited Senior Systems Architect 13-15 Hunslet Road james.holden(a)uk.aql.com Leeds, LS10 1JQ 07989 702968 0800 389 5815 ===================================================== http://aql.com/documents/news/about_aql.pdf "The service has been perfect." BSI Inspectorate ltd "really really really useful. easy to use and reduces manpower." Fortnums "without a shadow of a doubt...excellent" Ruella James Plc ----------------------------------------------------- (aq) Limited 13-15 Hunslet Road Leeds LS10 1JQ 194 Harrogate Road Leeds LS7 4NZ http://www.aql.com Sales: 08707 449227 Fax: 08707 449228 PGP public key http://aql.co.uk/A-Q.txt Subject to our standard terms, found at http://aql.com/site/termsofservice.php ----------------------------------------------------- ****************************** CONFIDENTIALITY This e-mail is intended only for the use of the addressees named above and may be confidential or legally privileged. If you are not an addressee you must not read it and must not use any information contained in nor copy it nor inform any person other than (aq) limited or the addressees of its existence or contents. If you have received this email and are not a named addressee, please delete it and notify the (aq) limited customer service department via admin(a)a-q.co.uk. Many thanks. =========================================================

2 3

[Serusers] Dispatcher Module Help
by Giovanni Chirico 05 Feb '07

05 Feb '07

Hi, I would like to use the dispatcher module. I have configured it on my ser.cfg file for the PSTN calls and it works fine. However when the select gateway is unavailable the module don't switch to another gateway. Can you help me to write the "failure_route" so I can use another gateway when the first is unavailable? 'm using the "call forwarding ser.cfg" file of the SER Getting Started <http://www.iptel.org/ser/doc/gettingstarted> pack (in attachment I'm sending you my ser.cfg and dispatcher.list files) Thanks in advance for your help. Bye, Giovanni Chirico

1 0

Re: [Users] strip_tail() string length?
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Just a typo correction On 02/05/07 13:54, Brandon Armstead wrote: > Daniel-Constantin Mierla wrote: >> Stripping is now possible with the new introduced 'transformations': >> http://openser.org/dokuwiki/doku.php/transformations:devel >> >> So if you want to strip last two characters of username in R-URI >> >> $var(x) = $(rU{s.len})-2; >> $rU = $(ru{s.substr,0,$var(x)}); ^^^ this is: $rU = $(rU{s.substr,0,$var(x)}); Cheers, Daniel >> >> Cheers, >> Daniel >> >> >> On 01/30/07 09:08, Juha Heinanen wrote: >>> Brandon Armstead writes: >>> >>> > i.e. strip_tail(strlen($ru) - 2); ... or something like such. >>> >>> sorry, but my suggestion >>> >>> substr($ru, -2, 2) >>> >>> does not work, since it returns the last two chars of $ru not the >>> remaining chars. strlen is indeed needed for the above. using php >>> substr and strlen >>> >>> strip_tail(strlen($ru) - 2) >>> >>> becomes >>> >>> substr($ru, 0, strlen($ru) - 2) >>> >>> -- juha >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> > Hello Daniel, > > Glad to see this implemented, I think it will be a definite asset > to the future and improvements of OpenSER. Keep up the good work! > > Regards, > Brandon >

1 0

Re: [Users] In NAT, I didn't see the ACK packet after the Invited the Callee
by Bogdan-Andrei Iancu 05 Feb '07

05 Feb '07

Ravi, so you see any Record-Route hdrs in the 200 OK reply? regards, bogdan raviprakash sunkara wrote: > Hi BogDan, > Good AfterNoon, and Thanks For replying to me, > > I used the Following code for Record_route(); > > route { > .... > ..... > if(method=="INVITE" && nat_uac_test("3") ) > { > record_route_preset("sip: > 192.168.2.3:5060 <http://192.168.2.3:5060>;nat=yes"); > }else if ( method ! = " REGISTER ") > { > record_route(); > } > } > where 192.168.2.3:5060 <http://192.168.2.3:5060> is the private IP of > the SIP server behind the NAT. > > > On 2/5/07, * Bogdan-Andrei Iancu* <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi Ravi, > > do you do record_route() for the INVITE? if not, the sequential > requests > of the dialog will bypass your server. > > regards, > bogdan > > raviprakash sunkara wrote: > > > Hello Users, > > > > When I tested the OpenSER server with CISCO ATA 186/188 IP phones > > SIP EyeP media (Softphnoes ) is working fine with RTP , when Both > > Users and SIP Server is in Behind the NAT, > > > > But Problem with Softphones ( X-lite ) . > > > > When I ngrep the Calls of X-lite users i didn't see any Ack > Packet to > > OpenSER After the Invite Packets received . > > > > Please Help me in this Part > > > > > > -- > > Thanks and Regards > > Ravi Prakash Sunkara > > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com> > <mailto:ravi.sunkara@hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com>> > > M:+91 9985077535 > > O:+91 40 23114549 > > F:+91 40 40208727 > > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com> <mailto: > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com>> > > www.hyperion-tech.com <http://www.hyperion-tech.com> > <http://www.hyperion-tech.com> > > > >------------------------------------------------------------------------ > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > www.hyperion-tech.com <http://www.hyperion-tech.com>

2 2

Re: [Users] 200 ok not cought in ONREPLY
by Bogdan-Andrei Iancu 05 Feb '07

05 Feb '07

Dan, that is rather strange if you are confident you are setting correctly a reply route (not default one) for the INVITE transaction. add the following debug message in modules/tm/t_reply.c, after line 1312: DBG("---------onreply route is %u\n",t->on_reply); recompile ans see the output on runtime. regards, bogdan Dan-Cristian Bogos wrote: > Salut Bogdan, > > theoretically I have. Saw now that with default onreply I can catch it > (log). It is really strange since I am relying both BYE and INVITES in > the same t_relay block with t_on_reply("1"); > t_on_failure("1"); but the 200 OK comes to default. > > Cheers, > Dan > > On 2/5/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > >> Salut Dan, >> >> are you sure you have an onreply route sent for this transactions? from >> the output I see no problems. >> >> regards, >> bogdan >> >> Dan-Cristian Bogos wrote: >> >> > Hi Guys, >> > >> > By receiving incoming traffic from some other sip server (configured >> > for different domain than mines) I will not be able to catch in the >> > routing, onreply section, the 200 OK for the INVITE , OK which my >> > client, behind my server, is sending to remote side, therefore not >> > beeing able to modify sdp to use mediaproxy. Bellow is a level8 of >> > debug of the OK. >> > Is there any other way of catching this message in the routing? Any >> > help will be appreciated. >> > >> > Thxs in advance, >> > Dan >> > >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: SIP Reply >> > (status): >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: version: >> > <SIP/2.0> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: status: >> <200> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: reason: >> <OK> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.11b641a2.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the first via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: After >> > parse_msg... >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module nathelper, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.2ffc88b3.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the second via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 234, <received> = <87.139.12.167>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 235, <rport> = <5096>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bK521554023>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_param: tag=8b2d7e45 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:parse_to:end of header reached, state=29 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DBUG:parse_to: display={}, ruri={sip:dan@de.babble.net} >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > get_hdr_field: <To> [38]; uri=[sip:dan@de.babble.net] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: to >> > body [<sip:dan@de.babble.net>] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > get_hdr_field: cseq <CSeq>: <20> <INVITE> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module tm, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=0 T start=0xffffffff >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=22 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=8 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: hash 39598 label 705981201 branch 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:REF_UNSAFE: after is 1 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: reply matched (T=0x4081f430)! >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=1 T end=0x4081f430 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 >> > is_invite=1) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:t_should_relay_response: T_code=180, new_code=200 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: branch=0, save=0, relay=0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: old size: >> > 911, new size: 850 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > build_res_from_sip_res: copied size: orig:77, new: 16, rest: 834 msg= >> > SIP/2.0 200 OK^M Via: SIP/2.0/UDP >> > 87.102.50.17;branch=z9hG4bKeaa9.2ffc88b3.0^M Via: SIP/2.0/UDP >> > >> 10.10.10.132:5096;received=87.139.12.167;rport=5096;branch=z9hG4bK521554023^M >> >> > >> > Record-Route: <sip:87.102.50.13;lr=on;ftag=2240571840>^M Record-Route: >> > <sip:87.102.50.17;lr;ftag=2240571840>^M Contact: >> > <sip:dan@87.139.12.167:7472;rinstance=37d0f83d79f3ede2>^M To: >> > <sip:dan@de.babble.net>;tag=8b2d7e45^M From: >> > <sip:dan@babble.net>;tag=2240571840^M Call-ID: >> > 1174762859(a)10.10.10.132^M CSeq: 20 INVITE^M Allow: INVITE, ACK, >> > CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO^M >> > Content-Type: application/sdp^M User-Agent: X-Lite release 1006e stamp >> > 34025^M Content-Length: 185^M ^M v=0^M o=- 5 2 IN IP4 87.139.12.167^M >> > s=CounterPath X-Lite 3.0^M c=IN IP4 87.139.12.167^M t=0 0^M m=audio >> > 34556 RTP/AVP 0 101^M a=fmtp:101 0-15^M a=rtpmap:101 >> > telephone-event/8000^M a=sendrecv^M >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_to_tail_of_timer[2]: 0x4081f478 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: sent buf=0x814a048: SIP/2.0 2..., >> > shmem=0x40821528: SIP/2.0 2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > cleanup_uac_timers: RETR/FR timers reset >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:UNREF_UNSAFE: after is 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:destroy_avp_list: destroying list (nil) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: receive_msg: >> > cleaning up >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >> >

1 0

[Users] Hijack 'Move Temporarily'
by Sam Lee 05 Feb '07

05 Feb '07

Hey Guys, Normally if the SIP-UA is call-forwarded to a different number , it will send a 'Move Temporarily' to the proxy , which in turn relay this message to the originating gateway. The problem is , if they call forward to a number which the originating gateway doesn't know what to do with it , it will just kill the call. Can i hijack 'Move Temporarily' directly in the proxy and sending this call to a different gateway rather than relaying it to the originating gateway ? Please let me know if you can't understand me. Thanks! Regards, Sam

6 23

[Users] database persistent connection
by tele 05 Feb '07

05 Feb '07

Hi, I've an annoying problem with database persistent connection. I'm using mysql database for location and avp stuff. During the day all it's but after a day i retest some call and seams that the proxy lost connection with the database, and avp and location stuff didn't works anymore. I'm using this version of openser: # openser -V version: openser 1.2.0-dev8-notls (i386/freebsd) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, select, kqueue. @(#) $Id: main.c,v 1.23 2006/10/02 15:51:45 bogdan_iancu Exp $ main.c compiled on 12:38:52 Nov 27 2006 with gcc 3.4.4 i've also configure a small ping_interval for the mysql module modparam("mysql", "ping_interval", 60) modparam("mysql", "auto_reconnect", 1) and the usrloc: modparam("usrloc", "db_mode", 3) modparam("usrloc", "matching_mode", 1) I've also check that the NTP is correct on the servers. I don't know why openser lost connection with the database. any hints? thanks :tele

2 2

[Users] Help. in OpenSER in Post-paid and pre-paid
by raviprakash sunkara 05 Feb '07

05 Feb '07

Hello Users, I'm Using The Following software of My Sip productions, 1) OpenSER is a Sip Server,proxy, in front-end 2) Asterisk is PBX server, in Back-end 3) Radius for AAA to OpenSER, 4) Asterisk-java for IVR for Billing Customer care if I did any mistakes in these mail , forgive me . My Question is 1) If Caller calls to Callee , Before the call established to callee. The Caller has to know his Balance Account. if the Balance is there, then only the call has to establish, 2) I want the OpenSER total in RealTime application Server like Java application . Call Authorization ( Question no 1 ). If Asterisk is Front-end , ---> problem is Solved, How can a b2bua implement with OpenSER, please when -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

2 2

[Users] strip_tail() string length?
by Brandon Armstead 05 Feb '07

05 Feb '07

Hello everyone, I'm wondering what the easiest method would be to grab how long the RURI is and strip based on that information... or would I have to actually use regex, or is there an easier STRLEN() or LENGTH() operation? Any ideas, or is this something that could be implemented? Thanks. i.e. strip_tail(strlen($ru) - 2); ... or something like such. -- Brandon Armstead

3 5

[Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
by Sho Aihara 05 Feb '07

05 Feb '07

Hi all. I have a problem for the following scenario. When I make a call for PSTN and on hook by PSTN side, Cisco As can't send BYE for SER. 01. [UA via Asterisk] dialing "08022223333" -> [SER] 02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") -> [Cisco] 03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from "033308022223333" to "008022223333" 04. [Cisco] process an outbound call to "008022223333" -> [e.g. Mobile] 05. [e.g. Mobile] Catch call 06. [SER] log CDR start 07. [Cisco] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] can't send BYE for SER *****why??***** 10. [UA via Asterisk] On hook 11. [UA via Asterisk] Send BYE for SER 12. [SER] log CDR End [Cisco] Call finished But another scenario, if make a call from PSTN to Asterisk and on hook by PSTN side, Cisco As send BYE to SER. 01. [e.g. Mobile] dialing "0377771111(Asterisk user number)" 02. [Cisco] receive "77771111" call number 03. [Cisco] dial-peer voice 5000 voip, session target ipv4: my.ser.ip.address -> [SER] 04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk] 05. [UA via Asterisk] Catch call 06. [SER] log CDR start 07. [UA via Asterisk] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] Send BYE to SER 10. [SER] log CDR End [Cisco] Call finished 11. [UA via Asterisk] receive BYE from SER And sorry for my diffucult example. Why Cisco AS 5300 can't send BYE to SER When PSTN call is disconnected by PSTN side? My ser.cfg as follows: # -------------------------------------------------------------------------- # global configuration parameters # -------------------------------------------------------------------------- fork=no log_stderror=yes check_via=no dns=no rev_dns=no listen=my.ser.ip.address port=5060 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # -------------------------------------------------------------------------- # 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/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" # -------------------------------------------------------------------------- # setting module-specific parameters # -------------------------------------------------------------------------- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser") modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # -------------------------------------------------------------------------- # route pattern # -------------------------------------------------------------------------- route { 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; }; record_route(); if (loose_route()) { if (method=="ACK") { acc_db_request("01:CallStart\n", "acc"); }; if (method=="BYE" || method=="CANCEL") { acc_db_request("02:CallEnd\n", "acc"); }; t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; if (search("^(f|From): .*(a)(my\.cisco\.ip\.address)")) { #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111 rewritehost("my.asterisk.ip.address"); }; lookup("aliases"); if (!lookup("location")) { if (method=="INVITE" && !search("^(f|From): .*(a)(my\.cisco\.ip\.address)")) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; if (uri=~"^sip:0[0-9]{10}@") { # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333 prefix("0333"); rewritehostport("my.cisco.ip.address:5060"); avp_write("i:45", "inv_timeout"); } else { sl_send_reply("404", "Not Found"); break; }; consume_credentials(); }; }; }; if (!t_relay()) { sl_reply_error(); }; } And my Cisco AS 5300 config as follows: voice call send-alert voice rtp send-recv voice service pots fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco voice service voip fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco sip min-se 60 translation-rule 50 Rule 0 0333 0 Rule 1 ^7777 037777 voice class codec 2 codec preference 1 g711ulaw codec preference 2 g711alaw dial-peer voice 5000 voip tone ringback alert-no-PI description ser-asterisk-cisco-test huntstop destination-pattern 77771111$ translate-outgoing called 50 voice-class codec 2 session protocol sipv2 session target ipv4:my.ser.ip.address dtmf-relay rtp-nte max-conn 1 dial-peer voice 6000 pots application session max-conn 2 destination-pattern 0333T progress_ind alert enable 8 translate-outgoing called 50 port 0:D Thanks, Sahria

3 2

Re: [Users] 200 ok not cought in ONREPLY
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Hello, On 02/05/07 12:30, Dan-Cristian Bogos wrote: > Dear Daniel, > > many thanks for the reply. I can perfectly catch the OK now. ok > > The only thing which I must find out is why for a normal INVITE the OK > was not coming in the same onreply block as the BYE, since both INVITE > and BYE are relied in the same way out and processed theoretically in > the same route block. BYE is usually following the loose routing -- but I do not know if it is the case of your comfig. Cheers, Daniel > > Anyway, thank you again, it was really great help. > > Cheers, > Dan > > > On 2/5/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> Hello, >> >> to get a reply in an onreply_route other than the default on, you have >> to arm it via t_on_reply() when you relay the request. >> ... >> t_on_reply("1"); >> ... >> >> t_on_reply[1] { >> } >> >> Otherwise, all replies are processed in the default onreply_route (the >> one which has no index) >> >> onreply_route { >> .... >> } >> >> Cheers, >> Daniel >> >> On 02/01/07 15:48, Dan-Cristian Bogos wrote: >> > Hi Guys, >> > >> > By receiving incoming traffic from some other sip server (configured >> > for different domain than mines) I will not be able to catch in the >> > routing, onreply section, the 200 OK for the INVITE , OK which my >> > client, behind my server, is sending to remote side, therefore not >> > beeing able to modify sdp to use mediaproxy. Bellow is a level8 of >> > debug of the OK. >> > Is there any other way of catching this message in the routing? Any >> > help will be appreciated. >> > >> > Thxs in advance, >> > Dan >> > >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: SIP Reply >> > (status): >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: version: >> > <SIP/2.0> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: status: >> <200> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: reason: >> <OK> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.11b641a2.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the first via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: After >> > parse_msg... >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module nathelper, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.2ffc88b3.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the second via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 234, <received> = <87.139.12.167>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 235, <rport> = <5096>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bK521554023>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_param: tag=8b2d7e45 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:parse_to:end of header reached, state=29 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DBUG:parse_to: display={}, ruri={sip:dan@de.babble.net} >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > get_hdr_field: <To> [38]; uri=[sip:dan@de.babble.net] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: to >> > body [<sip:dan@de.babble.net>] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > get_hdr_field: cseq <CSeq>: <20> <INVITE> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module tm, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=0 T start=0xffffffff >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=22 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=8 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: hash 39598 label 705981201 branch 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:REF_UNSAFE: after is 1 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: reply matched (T=0x4081f430)! >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=1 T end=0x4081f430 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 >> > is_invite=1) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:t_should_relay_response: T_code=180, new_code=200 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: branch=0, save=0, relay=0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: old size: >> > 911, new size: 850 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > build_res_from_sip_res: copied size: orig:77, new: 16, rest: 834 msg= >> > SIP/2.0 200 OK^M Via: SIP/2.0/UDP >> > 87.102.50.17;branch=z9hG4bKeaa9.2ffc88b3.0^M Via: SIP/2.0/UDP >> > >> 10.10.10.132:5096;received=87.139.12.167;rport=5096;branch=z9hG4bK521554023^M >> >> > >> > Record-Route: <sip:87.102.50.13;lr=on;ftag=2240571840>^M Record-Route: >> > <sip:87.102.50.17;lr;ftag=2240571840>^M Contact: >> > <sip:dan@87.139.12.167:7472;rinstance=37d0f83d79f3ede2>^M To: >> > <sip:dan@de.babble.net>;tag=8b2d7e45^M From: >> > <sip:dan@babble.net>;tag=2240571840^M Call-ID: >> > 1174762859(a)10.10.10.132^M CSeq: 20 INVITE^M Allow: INVITE, ACK, >> > CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO^M >> > Content-Type: application/sdp^M User-Agent: X-Lite release 1006e stamp >> > 34025^M Content-Length: 185^M ^M v=0^M o=- 5 2 IN IP4 87.139.12.167^M >> > s=CounterPath X-Lite 3.0^M c=IN IP4 87.139.12.167^M t=0 0^M m=audio >> > 34556 RTP/AVP 0 101^M a=fmtp:101 0-15^M a=rtpmap:101 >> > telephone-event/8000^M a=sendrecv^M >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_to_tail_of_timer[2]: 0x4081f478 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: sent buf=0x814a048: SIP/2.0 2..., >> > shmem=0x40821528: SIP/2.0 2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > cleanup_uac_timers: RETR/FR timers reset >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:UNREF_UNSAFE: after is 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:destroy_avp_list: destroying list (nil) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: receive_msg: >> > cleaning up >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >

1 0

[Users] OpenSER 1.2.0 + Presence
by Bernard Chan 05 Feb '07

05 Feb '07

Hello, I'm currently experimenting with OpenSER 1.2.0 with Presence module. Earlier I tried OpenSER 1.0/1.1 with PA. I'm using a third-party softphone which for some reasons, I would not like to disclose the name of it. After switching to 1.2.0+Presence, the softphone segfaults on startup. After tracing the source code of the phone and OpenSER and performing hundreds of packet sniffing over the past several days, I have grown to suspect that a certain behaviour with the new presence module may be the cause, but I would like to verify whether that is the case. For some reasons it is infeasible for me to modify the source code of the softphone used (actually I may be able to but that is what I would like to avoid). But it seems to choke on receiving a NOTIFY before the OK for the corresponding SUBSCRIBE is received by the softphone. So, any chance that I can test it by just modifying the config of OpenSER alone without changing the sources, both of OpenSER and the softphone? Thanks in advance for any invaluable insights. Regards, Bernard Chan.

2 1

[Users] In NAT, I didn't see the ACK packet after the Invited the Callee
by raviprakash sunkara 05 Feb '07

05 Feb '07

Hello Users, When I tested the OpenSER server with CISCO ATA 186/188 IP phones SIP EyeP media (Softphnoes ) is working fine with RTP , when Both Users and SIP Server is in Behind the NAT, But Problem with Softphones ( X-lite ) . When I ngrep the Calls of X-lite users i didn't see any Ack Packet to OpenSER After the Invite Packets received . Please Help me in this Part -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

2 1

[Users] 200 ok not cought in ONREPLY
by Dan-Cristian Bogos 05 Feb '07

05 Feb '07

Hi Guys, By receiving incoming traffic from some other sip server (configured for different domain than mines) I will not be able to catch in the routing, onreply section, the 200 OK for the INVITE , OK which my client, behind my server, is sending to remote side, therefore not beeing able to modify sdp to use mediaproxy. Bellow is a level8 of debug of the OK. Is there any other way of catching this message in the routing? Any help will be appreciated. Thxs in advance, Dan Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: SIP Reply (status): Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: version: <SIP/2.0> Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: status: <200> Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: reason: <OK> Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: flags=2 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 232, <branch> = <z9hG4bKeaa9.11b641a2.0>; state=16 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of header reached, state=5 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: Via found, flags=2 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: this is the first via Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: After parse_msg... Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: forward_reply: found module nathelper, passing reply to it Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: flags=20 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 232, <branch> = <z9hG4bKeaa9.2ffc88b3.0>; state=16 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of header reached, state=5 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: Via found, flags=20 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: this is the second via Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 234, <received> = <87.139.12.167>; state=6 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 235, <rport> = <5096>; state=6 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 232, <branch> = <z9hG4bK521554023>; state=16 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of header reached, state=5 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: Via found, flags=20 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: add_param: tag=8b2d7e45 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:parse_to:end of header reached, state=29 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DBUG:parse_to: display={}, ruri={sip:dan@de.babble.net} Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: get_hdr_field: <To> [38]; uri=[sip:dan@de.babble.net] Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: to body [<sip:dan@de.babble.net>] Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: get_hdr_field: cseq <CSeq>: <20> <INVITE> Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: forward_reply: found module tm, passing reply to it Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: t_check: msg id=1 global id=0 T start=0xffffffff Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: flags=22 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: flags=8 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: t_reply_matching: hash 39598 label 705981201 branch 0 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:REF_UNSAFE: after is 1 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: t_reply_matching: reply matched (T=0x4081f430)! Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: t_check: msg id=1 global id=1 T end=0x4081f430 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 is_invite=1) Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:t_should_relay_response: T_code=180, new_code=200 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:relay_reply: branch=0, save=0, relay=0 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: old size: 911, new size: 850 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: build_res_from_sip_res: copied size: orig:77, new: 16, rest: 834 msg= SIP/2.0 200 OK^M Via: SIP/2.0/UDP 87.102.50.17;branch=z9hG4bKeaa9.2ffc88b3.0^M Via: SIP/2.0/UDP 10.10.10.132:5096;received=87.139.12.167;rport=5096;branch=z9hG4bK521554023^M Record-Route: <sip:87.102.50.13;lr=on;ftag=2240571840>^M Record-Route: <sip:87.102.50.17;lr;ftag=2240571840>^M Contact: <sip:dan@87.139.12.167:7472;rinstance=37d0f83d79f3ede2>^M To: <sip:dan@de.babble.net>;tag=8b2d7e45^M From: <sip:dan@babble.net>;tag=2240571840^M Call-ID: 1174762859(a)10.10.10.132^M CSeq: 20 INVITE^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO^M Content-Type: application/sdp^M User-Agent: X-Lite release 1006e stamp 34025^M Content-Length: 185^M ^M v=0^M o=- 5 2 IN IP4 87.139.12.167^M s=CounterPath X-Lite 3.0^M c=IN IP4 87.139.12.167^M t=0 0^M m=audio 34556 RTP/AVP 0 101^M a=fmtp:101 0-15^M a=rtpmap:101 telephone-event/8000^M a=sendrecv^M Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: add_to_tail_of_timer[2]: 0x4081f478 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:relay_reply: sent buf=0x814a048: SIP/2.0 2..., shmem=0x40821528: SIP/2.0 2 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:UNREF_UNSAFE: after is 0 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:destroy_avp_list: destroying list (nil) Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: receive_msg: cleaning up

3 2

Res: [Users] 407 and algorithm=MD5
by Bruno Machado 05 Feb '07

05 Feb '07

The section 3.2.1 (The WWW-Authenticate Response Header) from RFC 2617 says: "algorithm A string indicating a pair of algorithms used to produce the digest and a checksum. If this is not present it is assumed to be "MD5". " I send a INVITE to Openser but it send back a 407 (Proxy Authentication). The parameter response was OK. But, without the parameter 'algorithm', the proxy didn't allow the authentication. It neither consult at Radius. Looking at the RFC above, the SIP proxy must to use MD5 by default, but it don't accept the message without the parameter. Who needs to assume the algorithm? UAC or SIP Proxy? Can it be a bug of OpenSER? Or my interpretation is wrong? Best regards. Bruno Machado ----- Mensagem original ---- De: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Para: Bruno Machado <bsmufu(a)yahoo.com.br> Cc: users(a)openser.org Enviadas: Segunda-feira, 29 de Janeiro de 2007 14:21:31 Assunto: Re: [Users] 407 and algorithm=MD5 Hi Bruno, you cannot configure to have the algorithm param added. If I'm not wrong, the digest RFC says this is an optional param and the default value is MD5. regards, bogdan Bruno Machado wrote: > Hi. > > When I use the function proxy_challenge(), the 407 don't have the > parameter 'algorithm=MD5'. Look below: > > ... > Proxy-Authenticate: Digest realm="proxy.com.br", > nonce="45be00f941f137ff037436c77f80a9531a02155f". > ... > > How can I put this parameter in digest informations? > > ... > Proxy-Authenticate: Digest realm="proxy.com.br", > nonce="45be00f941f137ff037436c77f80a9531a02155f", > algorithm="MD5". > ... > > > Thanks in advance. > > Bruno > > __________________________________________________ > Fale com seus amigos de graça com o novo Yahoo! Messenger > http://br.messenger.yahoo.com/ > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

3 3

[Users] permission.allow example
by Ion Minzu 05 Feb '07

05 Feb '07

Hello , Can someone describe me a real example how to write rules in permission.allow, permission.deny. I use the example from file permission.* and it doesn't work, permission is set ALL to ALL. Or maybe exist other method to make call restrictions? #openser.cfg .... loadmodule "/usr/lib/openser/modules/permissions.so" .... modparam("permissions", "default_deny_file", "/etc/openser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/openser/permissions.allow") .... -- Best regards, Ion Minzu, mailto:ion.minzu@cts.md

2 1

[Serusers] Cisco AS 5300 can't send BYE for SER
by Sahria Hao 05 Feb '07

05 Feb '07

Hi all. I have a problem for the following scenario. When I make a call for PSTN and on hook by PSTN side, Cisco As can't send BYE for SER. 01. [UA via Asterisk] dialing "08022223333" -> [SER] 02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") -> [Cisco] 03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from "033308022223333" to "008022223333" 04. [Cisco] process an outbound call to "008022223333" -> [e.g. Mobile] 05. [e.g. Mobile] Catch call 06. [SER] log CDR start 07. [Cisco] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] can't send BYE for SER *****why??***** 10. [UA via Asterisk] On hook 11. [UA via Asterisk] Send BYE for SER 12. [SER] log CDR End [Cisco] Call finished But another scenario, if make a call from PSTN to Asterisk and on hook by PSTN side, Cisco As send BYE to SER. 01. [e.g. Mobile] dialing "0377771111(Asterisk user number)" 02. [Cisco] receive "77771111" call number 03. [Cisco] dial-peer voice 5000 voip, session target ipv4: my.ser.ip.address -> [SER] 04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk] 05. [UA via Asterisk] Catch call 06. [SER] log CDR start 07. [UA via Asterisk] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] Send BYE to SER 10. [SER] log CDR End [Cisco] Call finished 11. [UA via Asterisk] receive BYE from SER And sorry for my diffucult example. Why Cisco AS 5300 can't send BYE to SER When PSTN call is disconnected by PSTN side? My ser.cfg as follows: # -------------------------------------------------------------------------- # global configuration parameters # -------------------------------------------------------------------------- fork=no log_stderror=yes check_via=no dns=no rev_dns=no listen=my.ser.ip.address port=5060 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # -------------------------------------------------------------------------- # 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/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" # -------------------------------------------------------------------------- # setting module-specific parameters # -------------------------------------------------------------------------- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("usrloc", "db_url", " mysql://ser:heslo@localhost/ser") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_url", " mysql://ser:heslo@localhost /ser") modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # -------------------------------------------------------------------------- # route pattern # -------------------------------------------------------------------------- route { 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; }; record_route(); if (loose_route()) { if (method=="ACK") { acc_db_request("01:CallStart\n", "acc"); }; if (method=="BYE" || method=="CANCEL") { acc_db_request("02:CallEnd\n", "acc"); }; t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; if (search("^(f|From): .*(a)(my\.cisco\.ip\.address<.*(a)(my%5C.cisco%5C.ip%5C.address>)")) { #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111 rewritehost("my.asterisk.ip.address "); }; lookup("aliases"); if (!lookup("location")) { if (method=="INVITE" && !search("^(f|From): .*(a)(my\.cisco\.ip\.address<.*(a)(my%5C.cisco%5C.ip%5C.address>)")) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; if (uri=~"^sip:0[0-9]{10}@") { # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333 prefix("0333"); rewritehostport("my.cisco.ip.address:5060"); avp_write("i:45", "inv_timeout"); } else { sl_send_reply("404", "Not Found"); break; }; consume_credentials(); }; }; }; if (!t_relay()) { sl_reply_error(); }; } And my Cisco AS 5300 config as follows: voice call send-alert voice rtp send-recv voice service pots fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco voice service voip fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco sip min-se 60 translation-rule 50 Rule 0 0333 0 Rule 1 ^7777 037777 voice class codec 2 codec preference 1 g711ulaw codec preference 2 g711alaw dial-peer voice 5000 voip tone ringback alert-no-PI description ser-asterisk-cisco-test huntstop destination-pattern 77771111$ translate-outgoing called 50 voice-class codec 2 session protocol sipv2 session target ipv4:my.ser.ip.address dtmf-relay rtp-nte max-conn 1 dial-peer voice 6000 pots application session max-conn 2 destination-pattern 0333T progress_ind alert enable 8 translate-outgoing called 50 port 0:D Thanks, Sahria

1 0

[Users] OpenSER devel & SNOM 360
by mika.saari＠wipsl.com 05 Feb '07

05 Feb '07

Hi, I do have OpenSER 1.1.0 with TLS support on Debian Sarge working nicely, and my SNOM 360 registers against it nicely. When using OpenSER 1.2 (2 days old CVS build) with TLS on Debian Etch with SNOM 360 and own signed certificates (with tls/tools & own machine names) I get debug info (below) from OpenSER. Any ideas with that "peer did not send sertificate" ? Thank you so many times, -Mika --- clip --- Feb 2 13:15:10 localhost openser[20855]: tcp_receive_timeout: 0xb59e1900 expired (296, 297) lt=0 Feb 2 13:15:10 localhost openser[20855]: DBG: io_watch_del (0x8122980, 18, -1, 0x10) fd_no=2 called Feb 2 13:15:10 localhost openser[20855]: releasing con 0xb59e1900, state 0, fd=18, id=2 Feb 2 13:15:10 localhost openser[20855]: extra_data (nil) Feb 2 13:15:10 localhost openser[20859]: handle_tcp_child: reader response= b59e1900, 0 from 0 Feb 2 13:15:10 localhost openser[20859]: DBG: io_watch_add(0x8122820, 23, 2, 0xb59e1900), fd_no=15 Feb 2 13:15:10 localhost openser[20859]: handle_tcp_child: CONN_RELEASE 0xb59e1900 refcnt= 0 Feb 2 13:15:10 localhost openser[20859]: tcpconn_new: new tcp connection to: 193.65.183.233 Feb 2 13:15:10 localhost openser[20859]: tcpconn_new: on port 2063, type 3 Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Entered: Creating a whole new ssl connection Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Looking up socket based TLS server domain [193.65.183.13:5061] Feb 2 13:15:10 localhost openser[20859]: tls_find_server_domain: virtual TLS server domain not found, Using default TLS server domain settings Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Found socket based TLS server domain [0.0.0.0:0] Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Setting in ACCEPT mode (server) Feb 2 13:15:10 localhost openser[20859]: tcpconn_add: hashes: 462, 9 Feb 2 13:15:10 localhost openser[20859]: handle_new_connect: new connection: 0xb5a06c88 24 flags: 0002 Feb 2 13:15:10 localhost openser[20859]: send2child: to tcp child 0 6(20855), 0xb5a06c88 Feb 2 13:15:10 localhost openser[20855]: received n=4 con=0xb5a06c88, fd=18 Feb 2 13:15:10 localhost openser[20855]: DBG: io_watch_add(0x8122980, 18, 2, 0xb5a06c88), fd_no=1 Feb 2 13:15:11 localhost openser[20855]: tls_update_fd: New fd is 18 Feb 2 13:15:11 localhost openser[20855]: tls_update_fd: New fd is 18 Feb 2 13:15:11 localhost openser[20855]: tls_accept: Error in SSL: Feb 2 13:15:11 localhost openser[20855]: tls_error: error:140890C7:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate Feb 2 13:15:11 localhost openser[20855]: DBG: io_watch_del (0x8122980, 18, -1, 0x10) fd_no=2 called Feb 2 13:15:11 localhost openser[20855]: releasing con 0xb5a06c88, state -2, fd=18, id=9 Feb 2 13:15:11 localhost openser[20855]: extra_data 0xb59f1a18 Feb 2 13:15:11 localhost openser[20859]: handle_tcp_child: reader response= b5a06c88, -2 from 0 Feb 2 13:15:11 localhost openser[20859]: tcpconn_destroy: destroying connection 0xb5a06c88, flags 0002 Feb 2 13:15:11 localhost openser[20859]: tls_close: Closing SSL connection Feb 2 13:15:11 localhost openser[20859]: tls_update_fd: New fd is 24 Feb 2 13:15:11 localhost openser[20859]: tls_shutdown: Shutdown successful Feb 2 13:15:11 localhost openser[20859]: tls_tcpconn_clean: Entered -- clip ---

2 2

[Users] Asterisk OpenSER and Exchange 2007 Unified Messaging
by Jon Webster 05 Feb '07

05 Feb '07

Greetings list, I'm in the process of configuring Asterisk <-> OpenSER <-> Microsoft Exchange 2007 Unified Messaging. I have a windows based proxy that works great, and want to accomplish the same feat using OpenSER. Below are network captures from the working proxy and OpenSER. From comparing the working call and the OpenSER call it looks like OpenSER needs to modify the CONTACT header in its last "OK" response. I'm not sure that's right, but more importantly I'm not sure how to do this in the config. Any help or direction is greatly appreciated. Thanks, -jon p.s. Please cc me on any replies, I'm not currently subscribed to the list. openSER unsuccessfull call negotiation =========== # U +9.215930 asterisk:5060 -> openser:5060 INVITE sip:8885555@openser SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as43593b57. To: <sip:8885555@openser>. Contact: <sip:3149@asterisk>. Call-ID: 46109877480231241e3b8e9d57036e69@asterisk. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Sat, 03 Feb 2007 18:01:48 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 216. . v=0. o=root 381 381 IN IP4 asterisk. s=session. c=IN IP4 asterisk. t=0 0. m=audio 13614 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U +0.000305 openser:5060 -> asterisk:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport=5060. From: "Jon Webster" <sip:3149@asterisk>;tag=as43593b57. To: <sip:8885555@openser>. Call-ID: 46109877480231241e3b8e9d57036e69@asterisk. CSeq: 102 INVITE. Server: OpenSer (1.1.1-tls (i386/linux)). Content-Length: 0. Warning: 392 openser:5060 "Noisy feedback tells: pid=6890 req_src_ip=asterisk req_src_port=5060 in_uri=sip:8885555@openser out_uri=sip:5555@openser via_cnt==1". . # U +0.025313 openser:5060 -> asterisk:5060 SIP/2.0 180 Ringing. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as43593b57. TO: <sip:8885555@openser>;epid=BD-70-82-06-F9;tag=e38b6cdb9b. CSEQ: 102 INVITE. CALL-ID: 46109877480231241e3b8e9d57036e69@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport=5060. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # U +0.016314 openser:5060 -> asterisk:5060 SIP/2.0 200 OK. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as43593b57. TO: <sip:8885555@openser>;epid=BD-70-82-06-F9;tag=e38b6cdb9b. CSEQ: 102 INVITE. CALL-ID: 46109877480231241e3b8e9d57036e69@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport=5060. CONTACT: <sip:exchangeUM:5065;transport=Tcp;maddr=exchangeUM>. CONTENT-LENGTH: 197. CONTENT-TYPE: application/sdp. SERVER: RTCC/2.0.6017.0. . v=0. o=- 0 0 IN IP4 exchangeUM. s=Microsoft Exchange Speech Engine. c=IN IP4 exchangeUM. t=0 0. m=audio 59544 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. working proxy =================== # U +8.036801 asterisk:5060 -> goodproxy:5060 INVITE sip:5555@goodproxy SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK42383fff;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>. Contact: <sip:3149@asterisk>. Call-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Sat, 03 Feb 2007 18:02:40 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 216. . v=0. o=root 381 381 IN IP4 asterisk. s=session. c=IN IP4 asterisk. t=0 0. m=audio 17160 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U +0.003872 goodproxy:5060 -> asterisk:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK42383fff;rport;received=goodproxy. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>. Call-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. CSeq: 102 INVITE. User-Agent: M-Networks USR/1.0. Allow: INVITE, INFO, ACK, CANCEL, BYE, NOTIFY, BENOTIFY, SUBSCRIBE. Content-Length: 0. . # U +0.139238 goodproxy:5060 -> asterisk:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK42383fff;rport. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as07e96b4e. TO: <sip:5555@exchangeUM>;epid=BD-70-82-06-F9;tag=eb4f3cf30. CSEQ: 102 INVITE. CALL-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. MAX-FORWARDS: 70. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # U +0.010224 goodproxy:5060 -> asterisk:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK42383fff;rport. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as07e96b4e. TO: <sip:5555@exchangeUM>;epid=BD-70-82-06-F9;tag=eb4f3cf30. CSEQ: 102 INVITE. CALL-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. MAX-FORWARDS: 70. CONTACT: <sip:goodproxy:5060>. CONTENT-LENGTH: 197. CONTENT-TYPE: application/sdp. SERVER: RTCC/2.0.6017.0. . v=0. o=- 0 0 IN IP4 exchangeUM. s=Microsoft Exchange Speech Engine. c=IN IP4 exchangeUM. t=0 0. m=audio 32458 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. . # U +0.000894 asterisk:5060 -> goodproxy:5060 ACK sip:goodproxy:5060 SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK440651a7;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>;tag=eb4f3cf30. Contact: <sip:3149@asterisk>. Call-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . # U +5.475898 asterisk:5060 -> goodproxy:5060 BYE sip:goodproxy:5060 SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK5fabb75e;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>;tag=eb4f3cf30. Call-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. CSeq: 103 BYE. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . # U +0.028752 goodproxy:5060 -> asterisk:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK5fabb75e;rport. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as07e96b4e. TO: <sip:5555@exchangeUM>;tag=eb4f3cf30;epid=BD-70-82-06-F9. CSEQ: 103 BYE. CALL-ID: 7d0450601533753042ce2d835a3f98d2@asterisk. MAX-FORWARDS: 70. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=openser port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc /openser/tls/user/user-cert.pem" #tls_private_key = "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc /openser/tls/user/user-privkey.pem" #tls_ca_list = "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc /openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/openser/modules/auth.so" #loadmodule "/usr/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (method=="INVITE" && uri=~"^sip:888[0-9]*@") { log("\n\nSIP invite received from exchange12\n\n"); strip(3); t_relay("tcp:exchangeUM:5065"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri=="sip:openser:5060;transport=TCP") { sl_send_reply("200", "OK"); exit; } if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; 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(); }; exit; } route[2] { t_on_reply("1"); }

2 1

[Users] Change a Status Code From 182 to 180
by Marc Schaefer 05 Feb '07

05 Feb '07

Hello all, I've a small problem that causes a lot of problems and is in some way specific: I've a Openser 1.1.1 Server running on a public rootserver. User are registered with FritzBox ( DSL-Router with Voip-Stack) or a Soft/Hardphone being Nated. For the PSTN connectivity i've access to a PSTN Gateway from the BT. Everything works very nice with just one problem remaining. When one User gets a call from PSTN and picks it up, no problems. But the problem comes when a second call is made to the same User. His phone tells him that somebody is knocking and he can pickup the phone without any problem. The problem is on the callers side because he doesn't get any feedback that he is ringing/knocking, so most of the time he will just cancel the call. I did some research about it and here are my conclusions: The INVITE from the second call isn't answered with a "180 Ringing.." but with a "182 Queued.." which doesn't trigger a Ringing for the PSTN Call. When I talked to the BT support they did not really know about the status code 182 and could not give me an answer how their mediagateway will handle it ( They promised me to get the informtion :-) ). On the other side I can't change the Firmware so there's no way of persuading the DSL-Router to use 180 instead of 182. So my questions are: Can i change the status code at the proxy? If so, in which part of the routing logic do I have to change it? I tried a lot of different subst() and replace() commands but none had any effect on the sip messages it always kept being 182. Below you can see my configuration which is based on the template from ixcall ( http://www.ixcall.net/?q=node/32 ) with a lot of changes and support from the "getting started" guide from onsip.org, of course adapted to the openser 1.1.1 version I have attached my config. Keep in mind that this config will get through a m4 parser before being ready to be used ( mechanism can be seen at ixcall ). I hope my english is good enough to have my problem explained in a understandable manner Have fun Marc /* ==================================================================== Schnuffle Openser Configuration V6.3 Based on IxCall template V2.4 (http://www.ixcall.net/?q=node/32) And OnSip.org "Getting Started" Guid (http://siprouter.onsip.org/doc/gettingstarted/) ==================================================================== list of flags used 1 - Accounting 2 - Missed calls Accouting 3 - NAT SIP ping flag 4 - NAT invite detected 5 - registrar module nat_flag 6 - Enable NAT support 7 - Enable Trunk (pstn, asterisk or yate) 13 - Enable debugging -------------------------------------------------------------------- Routes 1 - Relay with NAT helper and no proxy authentication 2 - Method Register handler 3 - Method Invite handler (call routing) 4 - Trunk and PSTN gateway routing 5 - TLS routing ( not really used but still in config to keep the given message flow intact ) 6 - Relay with no NAT support and no proxy authentication 7 - User verify 10 - Relay with NAT helper and proxy authentication Reply 1 - NAT helper Fail 1 - Voicemail and message store ( stripped from all the voicemail and message storage stuff, so in fact just empty ) -------------------------------------------------------------------- */ #Global settings debug=3 log_facility=LOG_LOCAL7 fork=yes log_stderror=no mhomed=no # Multihomed - set this to no if you set a "listen" addres listen=OPENSER_IP # only want to bind to this ip port=OPENSER_PORT children=4 tcp_children=4 tcp_accept_aliases=yes alias=OPENSER_REALM # your realm alias=OPENSER_HOST # alias domains of your SRV records alias=OPENSER_IP # and for the die hard ip users. # uncomment to override config values for debugging /* debug=7 # debug level (cmd line: -ddd) fork=no port=OPENSER_PORT log_stderror=yes # (cmd line: -E) fifo="OPENSER_FIFO" */ dns=no rev_dns=no fifo="OPENSER_FIFO" fifo_mode=OPENSER_FIFO_MODE #TLS specific settings disable_tls = DISABLE_TLS #listen = tls:OPENSER_TLS_IP:OPENSER_TLS_PORT #tls_verify = 1 #tls_require_certificate = 0 #tls_method = SSLv23 #tls_certificate = "DEFAULT_TLS_CERTIFICATE" #tls_private_key = "DEFAULT_TLS_PRIVATE_KEY" #tls_ca_list = "DEFAULT_TLS_CA_LIST" mpath="OPENSER_MODULE_PATH" #path to openser modules loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "acc.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "group.so" loadmodule "xlog.so" loadmodule "enum.so" loadmodule "lcr.so" fifo_db_url="mysql://OPENSER_DB_USER:OPENSER_DB_PW@OPENSER_DB_HOST/OPENSER_DB" modparam("avpops|auth_db|alias_db|permissions|uri_db|usrloc|acc|msilo|group|domain|lcr|pdt","db_url", "mysql://OPENSER_DB_USER:OPENSER_DB_PW@OPENSER_DB_HOST/OPENSER_DB") modparam("auth_db", "calculate_ha1", CALCULATE_HA1) modparam("auth_db", "password_column", "PASSWORD_COLUMN") modparam("auth_db", "password_column_2", "ha1b") modparam("auth", "secret", "OPENSER_DIGEST_PW") #nathelper, mediaproxy or both can be used #we will only be using mediaproxy modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "MEDIAPROXY_SOCKET") #modparam("mediaproxy","sip_asymmetrics","MEDIAPROXY_SIP_ASYMMETRICS") #modparam("mediaproxy","rtp_asymmetrics","MEDIAPROXY_RTP_ASYMMETRICS") # -- lcr params -- modparam("lcr","gw_table","gw") modparam("lcr","gw_name_column","gw_name") modparam("lcr","ip_addr_column","ip_addr") modparam("lcr","port_column","port") modparam("lcr","uri_scheme_column","uri_scheme") modparam("lcr","transport_column","transport") modparam("lcr","grp_id_column","grp_id") modparam("lcr","db_mode",0) # -- group params -- modparam("group", "table", "grp") modparam("group", "use_domain", 0) # might be usefull for future expansions modparam("group", "user_column", "username") modparam("group", "domain_column", "domain") modparam("group", "group_column", "grp") # -- usrloc params -- modparam("usrloc", "db_mode", 1) modparam("usrloc", "desc_time_order", 1) # -- registrar params -- modparam("registrar", "sip_natping_flag", 3) modparam("registrar", "nat_flag", 5) modparam("registrar", "desc_time_order", 1) # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- tm params -- #modparam("tm", "fr_timer", 20 ) modparam("tm", "fr_inv_timer", 90 ) #modparam("tm", "wt_timer", 20 ) modparam("tm", "via1_matching", 1) modparam("tm", "fr_inv_timer_avp", "s:inv_timeout") # -- permission params -- modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") # -- acc params -- modparam("acc", "db_flag", 1 ) modparam("acc", "db_missed_flag", 2) modparam("acc", "report_cancels", 1) modparam("acc", "log_flag", 1) # -- alias_db params -- modparam("alias_db", "user_column", "username") modparam("alias_db", "domain_column", "domain") modparam("alias_db", "alias_user_column", "alias_username") modparam("alias_db", "alias_domain_column", "alias_domain") #--enum params -- #modparam("enum", "domain_suffix", "e164.arpa.") # -- domain params -- modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") # -- avpops params -- # avpops is used to store the local prefix from each customer in subscriber table # and append it as soon as a local call is done (meaning a call matching [^0][0-9]*) modparam("avpops","db_scheme","prefix-scheme:username_col=username;domain_col=domain;value_col=prefix;value_type=string;table=subscriber") modparam("avpops","avp_table","subscriber") modparam("avpops","use_domain",0) modparam("avpops","username_column","username") modparam("avpops","domain_column","domain") modparam("avpops","avp_aliases","prefix=s:800") # ------------------------- request routing logic ------------------- # main routing logic route { # ----------------------------------------------------------------- # Configuration flags # ----------------------------------------------------------------- NAT_SUPPORT_FLAG_6 TRUNK_ENABLED_FLAG_7 DEBUG_FLAG_13 # changing return code 182 to 180 so that the calling phone will ring # does not work at all #if (status=="182") { # xlog("Changing Status Code from 182 to 180"); # subst('/182 Queued/180 Ringing/ig'); #} #xlog("Replacing Status 1820 with 180"); #replace_all("182 Queued","180 Ringing"); if (isflagset(13) && !is_method("ACK") && !is_method("REGISTER")) { #debug route route(13); }; # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too Many Hops"); }; return; }; if (msg:len > max_len) { if (method!="ACK") { sl_send_reply("513", "Message Overflow"); }; return; }; if (status=="482") { #loop detection xlog("L_ERR", "OPENSER_NAME: $mi $rm $fu -> $ru status 482 Loop Detected\n"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE") { record_route(); if (client_nat_test("3") && isflagset(6)) { add_rr_param(";nat=yes"); }; }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (isflagset(6)) { if (has_totag() && (method=="INVITE" || method=="ACK" || method=="BYE")) { setflag(1); # not really clear which of the setflags triggered the "BYE" and "CANCEL" to be accounted setflag(2); if (client_nat_test("3")||search("^Route:.*;nat=yes")){ xlog("L_NOTICE","Loose Route: Using MediaProxy"); use_media_proxy(); }; }; }; route(6); return; }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (isflagset(6)) { if (method=="BYE" || method=="CANCEL") { setflag(1); setflag(2); end_media_session(); }; }; # ----------------------------------------------------------------- # Unauthorized relay # ----------------------------------------------------------------- if (!is_uri_host_local()) { xlog("L_NOTICE","Prevent unauthorized Access is_uri_host_local=false, URI: $u, From: $fu -> Request: $ru\n"); if (is_from_local() || allow_trusted()) { # We prevent unautherised relays xlog("is_from_local or allow_trusted is true\n"); append_hf("P-hint: outbound\r\n"); route(10); # need auth } else { xlog("L_NOTICE","Uauthorized Relay: Reply a Forbidden\n"); sl_send_reply("403", "Forbidden"); }; return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri=~"sip:daemon@" ) { sl_send_reply("410", "daemon has gone for a beer"); return; }; # Calls matching means it is a local call without prefix # So prefix needs to be added expect for special numbers if (uri=~"sip:[1-9][0-9]*@" ) { # Check if uri contains special numbers ( 110,112 and some other 11[0-9]+ numbers ) # that should not be changed if (!uri=~"sip:11.*") { # add prefix to number # load user prefix and store it in $avp(s:800) # the prefix is stored in the subscriber table in an supplement attribute prefix if(!avp_db_load("$fu","$avp(s:800)/$prefix-scheme")) { xlog("Failed to load User Prefix"); } else {; # prefix R-URI xlog("Adding Prefix: $avp(s:800) to $ru"); subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:800)\1@\2;orig_uri=\1-\2/i'); xlog("R-URI: $ru"); } } } if (method=="INVITE") { route(3); return; } else if (method=="ACK") { route(1); return; } else if (method=="BYE" || method=="CANCEL") { setflag(1); setflag(2); route(1); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); #alias_db_lookup("dbaliases"); if (uri!=myself) { xlog("L_NOTICE", "Lookup Aliases failed"); route(10); return; }; if (!lookup("location")) { xlog("L_NOTICE", "Lookup Location failed"); route(8); return; }; route(1); # Do not proxy authenticate - needed by system to send messages return; } route[1] { # ----------------------------------------------------------------- # Default Route relay Handler with NAT ENABLED # ----------------------------------------------------------------- if (isflagset(6)) { if (client_nat_test("3")) { setflag(4); force_rport(); fix_contact(); }; if (isflagset(4) || isflagset(5)) { xlog("L_NOTICE","Route(1) using Mediaproxy"); use_media_proxy(); }; t_on_reply("1"); }; route(6); # Relay the call return; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- # This cool register script came from http://openser.org/docs/scripting.html sl_send_reply("100", "Trying"); if (!route(7)) { # verify the user return(0); }; if (!search("^Contact:[ ]*\*") && client_nat_test("7") && isflagset(6)) { setflag(3); setflag(5); force_rport(); fix_contact(); }; if (!is_user_in("From", "noauth")) { #no authentication required if (!www_authorize("", "subscriber")) { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu Authorization for $fu ($si) requested\n"); }; www_challenge("", "0"); return(0); } else { if (isflagset(13)){ xlog("L_NOTICE","OPENSER_NAME: $mi route[$rm][2] \ $fu Authorization OK\n"); }; }; # Prohibit attempts to grab someone else's To address using valid credentials if (!check_to()) { if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu -> $ru REPLY 401 - Unauthorized check_to() failed\n"); }; sl_send_reply("401", "Unauthorized - account and username must match"); return(0); }; consume_credentials(); }; if (!save("location")) { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu save(location) Failed\n"); }; sl_reply_error(); return(0); }else { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu save(location)OK\n"); }; }; return; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- setflag(1); #turn on accounting setflag(2); #also missed calls lookup("aliases"); #in the event that an alias resolves to another domain. #alias_db_lookup("dbaliases"); if (uri!=myself) { xlog("L_NOTICE","No Local Call Check Auth I: r-uri: <$ru> from-uri:<$fu>\n"); route(10); #check authentication return; }; if (!lookup("location")) { xlog("L_NOTICE","Lookup Location Failed: r-uri: <$ru> from-uri:<$fu>\n"); if (src_ip==OPENSER_PBX_IP || !isflagset(7)) { # loopback prevention route(8); return; } else { # if it's not here, go hassle asterisk. xlog("L_NOTICE","Trunk Routing: r-uri: <$ru> from-uri:<$fu>\n"); route(4); return; }; }; t_on_failure("1"); #call waiting,busy,DND,voicemail route(1); # local normal routing return; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- avp_write("$avp(i:120)", "$avp(s:inv_timeout)"); xlog("L_NOTICE","Loading Gateways form Database"); if (!load_gws()){ xlog("L_ERR","****************************** LoadGateways Failed **********************"); sl_send_reply("500", "Internal server error, unable to load phone gateways"); return; }; if (!next_gw()) { xlog("L_NOTICE","******************** Didn'T found Any Gateways *************************"); sl_send_reply("503", "Service not available - No gateways"); return; }; # check if we route through BT # if so, change From Header domain to IP ( @callway-voip.de -> @212.65.6.4 ) # BT Gateway is in GatewayGroup 1 if (to_gw("2")) { xlog("L_NOTICE","Relaying to External Gateway, so rewriting From-HEADER:$fu\n"); #subst('/182 Queued/180 Ringing/ig'); #replace_all("182 Queued","180 Ringing"); subst('/From:([^@]*)@callway-voip.de/From:\1@212.65.6.4/ig'); }; #if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][4] \ time [$Tf] Next Gateway $ruri_user_avp method <$rm> r-uri <$ru> 2nd via <$hdr(via[1])>\n"); #}; route(1); return; } route[5] { # ----------------------------------------------------------------- # TLS Handling # ----------------------------------------------------------------- # if (!is_uri_host_local()) { # if you have some interdomain connections via TLS # if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; # } else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; # }; # }; if (!is_uri_host_local()) { remove_hf("Alert-Info"); if (method=="INVITE" && proto==tls) { append_hf("Alert-info: 1\r\n"); #cisco 7960 append_hf("Alert-info: Bellcore-dr4\r\n"); #cisco ATA # append_hf("Alert-info: http://foo.bar/x.wav\r\n"); #snom }; }; return; } route[6] { # ----------------------------------------------------------------- # Route relay Handler # ----------------------------------------------------------------- #subst('/182 Queued/180 Ringing/ig'); #replace_all("182 Queued","180 Ringing"); if (!t_relay()) { if (isflagset(6)) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; }; xlog("L_NOTICE","Route6: T_relay failed"); sl_reply_error(); }; return; } route[7] { # ----------------------------------------------------------------- # User verify # ----------------------------------------------------------------- # Check for known domain(s) if (!is_from_local()) { if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu Unknown domain\n"); sl_send_reply("403", "Unknown domain"); }; return(0); }; # Check to see that the user has not been disabled if (is_user_in("From", "disabled")) { if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu Your account has been disabled\n"); }; sl_send_reply("403", "Your account has been disabled"); return(0); }; if (is_user_in("To", "disabled")) { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $tu That account has been disabled\n"); }; sl_send_reply("403", "That account has been disabled"); return(0); }; return(1); } route[8] { # ----------------------------------------------------------------- # Handling of unavailable user # ----------------------------------------------------------------- # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) { sl_send_reply("404", "Not Found"); return; }; t_newtran(); t_reply("404", "Not Found"); # we account missed incoming calls; previous statteful processing # guarantees that retransmissions are not accounted acc_db_request("404 missed call", "missed_calls"); return; } route[10] { # ----------------------------------------------------------------- # Default Message Handler with Proxy Authentication # ----------------------------------------------------------------- if(method=="ACK") { #these you never proxy authenticate route(1); return; }; if(method=="BYE" || method=="CANCEL") { #these you never proxy authenticate setflag(1); route(1); return; }; xlog("L_INFO", "Proxy auth $fd $dP destination:$du $dd $ds"); if (!route(7)) { #verify the user return(0); }; if (!is_user_in("From", "noauth")) { #no authentication required if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; # if (is_user_in("Credentials", "local")) { # Uncomment to use the group options route(1); # }else{ # sl_send_reply("403", "Denied, you are not allowed this route"); # }; return; } onreply_route[1] { # ----------------------------------------------------------------- # Return Route # ----------------------------------------------------------------- #replace_all("182 Queued","180 Ringing"); if ((isflagset(4) || isflagset(5)) && isflagset(6) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { xlog("L_NOTICE","OnReplyRoute(1) Using Mediaproxy"); use_media_proxy(); }; }; if (client_nat_test("1") && isflagset(6)) { fix_contact(); }; return; } failure_route[1] { # ----------------------------------------------------------------- # No answer, cancel, busy handling # ----------------------------------------------------------------- return; } route[13] { # ----------------------------------------------------------------- # Debuging info # ----------------------------------------------------------------- if (status=="100") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 100 Trying\n"); } else if (status=="180") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 180 Ringing\n"); } else if (status=="181") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 181 Call Is Being Forwarded\n"); } else if (status=="182") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 182 Queued\n"); } else if (status=="183") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 183 Session Progress\n"); } else if (status=="200") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 200 OK\n"); } else if (status=="202") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 202 Accepted\n"); } else if (status=="300") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 300 Multiple Choices $ct\n"); } else if (status=="301") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 301 Moved Permanently $ct\n"); } else if (status=="302") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 302 Moved Temporarily $ct\n"); } else if (status=="305") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 305 Use Proxy\n"); } else if (status=="380") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 380 Alternative Service\n"); } else if (status=="400") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 400 Bad Request\n"); } else if (status=="401") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 401 Unauthorized\n"); } else if (status=="402") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 402 Payment Required\n"); } else if (status=="403") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 403 Forbidden\n"); } else if (status=="404") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 404 Not Found\n"); } else if (status=="405") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 405 Method Not Allowed\n"); } else if (status=="406") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 406 Not Acceptable\n"); } else if (status=="407") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 407 Proxy Authentication Required\n"); } else if (status=="408") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 408 Request Timeout\n"); } else if (status=="409") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 409 Conflict\n"); } else if (status=="410") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 410 Gone\n"); } else if (status=="411") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 411 Length Required\n"); } else if (status=="412") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 412 Conditional Request Failed\n"); } else if (status=="413") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 413 Request Entity Too Large\n"); } else if (status=="414") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 414 Request-URI Too Long\n"); } else if (status=="415") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 415 Unsupported Media Type\n"); } else if (status=="416") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 416 Unsupported URI Scheme\n"); } else if (status=="420") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 420 Bad Extension\n"); } else if (status=="421") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 421 Extension Required\n"); } else if (status=="422") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 422 Session Interval Too Small\n"); } else if (status=="423") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 423 Interval Too Brief\n"); } else if (status=="429") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 429 Provide Referrer Identity\n"); } else if (status=="480") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 480 Temporarily Unavailable\n"); } else if (status=="481") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 481 Call Leg/Transaction Does Not Exist\n"); } else if (status=="482") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 482 Loop Detected\n"); } else if (status=="483") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 483 Too Many Hops\n"); } else if (status=="484") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 484 Address Incomplete\n"); } else if (status=="485") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 485 Ambiguous\n"); } else if (status=="486") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 486 Busy Here\n"); } else if (status=="487") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 487 Busy Here; Request Terminated\n"); } else if (status=="488") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 488 Not Acceptable Here\n"); } else if (status=="489") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 489 Bad Event\n"); } else if (status=="491") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 491 Request Pending\n"); } else if (status=="493") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 493 Undecipherable\n"); } else if (status=="494") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 494 Security Agreement Required\n"); } else if (status=="500") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 500 Server Internal Error\n"); } else if (status=="501") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 501 Not Implemented\n"); } else if (status=="502") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 502 Bad Gateway\n"); } else if (status=="503") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 503 Service Unavailable\n"); } else if (status=="504") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 504 Server Timeout\n"); } else if (status=="505") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 505 Version Not Supported\n"); } else if (status=="513") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 513 Message Too Large\n"); } else if (status=="580") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 580 Precondition Failure\n"); } else if (status=="600") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 600 Busy Everywhere\n"); } else if (status=="603") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 603 Decline\n"); } else if (status=="604") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 604 Does Not Exist Anywhere\n"); } else if (status=="606") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 606 Not Acceptable\n"); } else { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status not listed\n"); }; xlog("L_NOTICE", "OPENSER_NAME: $mi $mb\n"); return; }

2 1

[Serusers] jabber2foreign IM
by Allan Chao ( 策略性新技術開發處申請 ) 05 Feb '07

05 Feb '07

Dear All: I am a ser user, i want to use sip client that can commuicate with foreign IM client(msn), my flow is sip client -> ser(jabber module) (0.10 version) -> jabber server(has msn gateway) -> MSN network. but i did not see any function that can register to MSN server(just like jabber client browse transport in the jabber server and then register to MSN network) and there is no function or program using msn id and password. i doubt jabber module does not implement this part of program.Maybe there is another mathod can solve it. Does anyone can help me ? thanks. allan DISCLAIMER: Sample Disclaimer added in a VBScript.

2 1

[Users] Exec - environment variables - not found?
by Kjell Arild Frigstad 04 Feb '07

04 Feb '07

Hello! I have just upgraded from 1.0.0 til 1.1.1, and I having problem with exec_msg (the exec_msg working, but environment variabels is "not found".) It was OK in 1.0.0 version. The coding: exec_msg("echo SIP_SRCIP $SIP_SRCIP >>/var/log/messages"); At startup (with debuggging) : 0(0) fixing exec_msg, line 160 0(0) xl_parse_format: parsing [echo SIP_SRCIP $SIP_SRCIP >>/var/log/messages] 0(0) xl_lookup_spec_name: not found [SIP_SRCIP] 0(0) xl_parse_format: format parsed OK: [2] items Error message when running the command (with debugging) : 0(28888) xl_get_spec_value: error - null sp->itf 0(28888) exec:w_exec_msg: executing [echo SIP_SRCIP >>/var/log/messages] Error message when running in "normal" mode Feb 4 21:51:24 test /usr/local/sbin/openser[28947]: xl_get_spec_value: error - null sp->itf SIP_SRCIP As you see the text SIP_SRCIP is beeing displayd, but not the variabel. Any ide what can bee wrong? Best regards K.A.Frigstad

2 1

[Users] auth_radius extra attributes
by Cseke Tamas 03 Feb '07

03 Feb '07

Hello, can anybody tell me how can i send extra attributes to radius in authpacket, in acctpacket i can with radius_extra parameter, is there a way to use it in auth too? Thanks any help, Tamas

1 0

[Serusers] Problem with mysql.so on new server
by sip 03 Feb '07

03 Feb '07

So... I've had ser 0.9.6 running on RH EL3, and FC3 and 4. I'm building out a server on RH EL4 for SER, but I've run into a snag. I'm sure it's something simple, but it has me stumped at the moment. I've recompiled mysql.so on the new server... compiles fine (as do the rest of the modules). However, when I try to start up SER with mysql.so loaded, it gives me this: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: log Undefined symbol: log Never seen this before. Anyone know what it means or, more importantly, how to fix it (and no... I'm not going to scrap it all and go with Ottendorf yet, so that's not a solution ;) )? N.

4 12

[Serusers] Which locking mechanism is useful for usrloc in production environment?
by Ramu Yadav 03 Feb '07

03 Feb '07

Hello, At present we are using "sys v semaphores"(USE_SYSV_SEM) for locking. But we see a some delay while acquiring the lock to modify the usrloc's location table. We are in a thought to use "posix semaphores" (USE_POSIX_SEM), will this improve any performance for usrloc location table. Thoughts and Ideas will be appreciated. -- Ramu Yadav

4 5

[Users] NEW FEATURE: better timers
by Bogdan-Andrei Iancu 02 Feb '07

02 Feb '07

Hi everybody, following to some discussions / requests from users, openser devel branch has now core support for timers will resolution less than a second. It uses time functions that can give a microsecond accuracy, but for performance reasons, the utimer (microtimer) uses a ticking period of 100 milliseconds. Of course, this can be changed at compile time. First "beneficiary" of this timer, is TM, to gain more accuracy for retransmissions: * retransmission timers use the the core utimer (with microsecond resolution), having 100 milliseconds intervals. * this will increase the retransmissions precision to an error of maximum 100 milliseconds. * T1 (first retransmission) is done at 500 milliseconds (instead of 1 sec), as the RFC3261 suggests. * T1 and T2 (see RFC3261) can be configured via module parameters - the inter-steps are automatically computed on startup based on the RFC3261 specifications. Along with increasing the timer accuracy, the TM timer lists were improved for faster operations: * faster timers lists - use shortcut pointers (based on timeout values) for faster walking through the list -> faster insert and expire * for elements to be deleted from the timer lists, do not overwrite the timeout value as this will trigger more processing when inserting new elements into the list The performance tests are not completed - I need more machines to stress the proxy :D -, but I will come back with results during the testing phase. regards, bogdan

1 0

[Users] removed avp_write(); avp_aliases moved in core
by Daniel-Constantin Mierla 02 Feb '07

02 Feb '07

Hello, two major changes were applied to avpops module in development branch (1.2.0): - avp_write() has been removed - same functionality is provided by assignment operator in routing blocks, together with latest commit of PV transformations - avp_aliases parameter has been removed from avpops module and introduced as core parameter. you can use it directly in core, in global parameters section in form of: avp_aliases="uuid=I:660;email=s:email_addr;fwd=i:753" The value is the same as it was for avpops parameter. Now the AVP aliases can be used directly in routing blocks. Cheers, Daniel

1 0

[Users] new feature: pseudo-variable transformations
by Daniel-Constantin Mierla 02 Feb '07

02 Feb '07

Hello, currently in CVS trunk of development version is a new feature 'transformations'. A transformations is basically a function that can be applied to a pseudo-variable (PV). In this way, the value of PV is not altered but the result is something different. They are intended to facilitate access to different attributes of PV (like strlen of value, parts of value, substrings) or complete different value of PV (encoded in hexa, md5 value, escape/unescape PV value for DB operations...). The format is '{id[,parameters]}' and they can be used inside parenthesis of pseudo-variables (note that even short name pseudo-variables must be enclosed in '(' and ')' to be able to apply transformations) For example: - $(ru{s.len}) - returns the lenth of R-URI - $(avp(i:3){s.md5}) - returns the md5 string over value of $avp(i:3) The following classes of transformations are implemented 1) string - the name of transformation starts with 's.'. Available transformations in this class: - {s.len} - returns strlen of PV value - {s.md5} - returs md5 over PV value - {s.substr,offset,length} - returns substring starting at offset having size of 'length'. If offset is negative, then it is counted from the end of PV value, -1 being the last char. In case of positive value, 0 is first char. Length must be positive, in case of 0, substring to the end of PV value is returned. offset and length can be PV as well. Example: "abcd"{s.substr,1,0} = "bcd" - {s.select,index,separator} - returns a field from PV value. The field is selected based on separator and index. The separator must be a character used to identify the fields. Index must be a integer value or a PV. If index is negative, the count of fields starts from end of PV value, -1 being last field. If index is positive, 0 is the first field. Example: "12,34,56"{s.select,1,,} = "34" ; "12,34,56"{s.select,-2,,} = "12" - {s.encode.hexa} - returns encoding in hexa of PV value - {s.decode.hexa} - returns decoding from hexa of PV value - {s.escape.common} - returns escaped string of PV value. Characters escaped are ''', '"', '\' and 0. Useful when doing DB queries (care should be taken for non latin character set) - {s.unescape.common} - returns unescaped string of PV value. Reverse of above transformation. 2) uri - the name of transformation starts with 'uri.'. The PV value is considered to be a SIP URI. This transformation returns parts of SIP URI (see struct sip_uri). Available transformations in this class: - {uri.user} - returns the user part - {uri.host} or {uri.domain} - returns the domain part - {uri.passwd} - returns the password - {uri.port} - returns the port - {uri.params} - returns the URI parameters in a string - {uri.param,name} - returns the value of parameter with name 'name' - {uri.headers} - returns URI headers - {uri.transport} - returns the value of transport parameter - {uri.ttl} - returns the value of ttl parameter - {uri.uparam} - returns the value of user parameter - {uri.maddr} - returns the value of maddr parameter - {uri.method} - returns the value of method parameter - {uri.lr} - returns the value of lr parameter - {uri.r2} - returns the value of r2 parameter 3) parameters list - the name of the transformation starts with 'param.'. The PV value is considered to be a string like "name1=value1;name2=value2;...". The transformations returns the value for a specific parameter, or the name of a parameter at a specific index. Available transformations in this class: - {param.value,name} - returns the value of parameter 'name' Example: "a=1;b=2;c=3"{param.value,c} = "3" - {param.name,index} - returns the name of parameter at position 'index'. Example: "a=1;b=2;c=3"{param.name,1} = "b" Within a PV, many transformation can be applied, being executed from left to right. Example: $var(x) = "a=1;b=22;c=333"; $(var(x){param.value,$(var(x){param.name,1})}{s.len}) = 2 - the length of the value of parameter at position 1 (remember 0 is first position, 1 is second position) The transformations can be used anywhere, being considered parts of PV -- in xlog, avpops or other modules' functions and parameters, in right side assignment expressions or in comparisons. Documentation page for transformations was opened at: http://openser.org/dokuwiki/doku.php/transformations:devel Feedback is welcome! Cheers, Daniel

1 0

[Serusers] meeting (was: Re: [Serdev] SER's core design features(process model/parser/lumps/script) - was: So who/what is SER for, anyway?
by Jiri Kuthan 02 Feb '07

02 Feb '07

We could use the IETF meeting in Prague in March. Actually, I have anticipated that and there is a hotel room reserved already :-) Could folks willing to show up and debate roadmap or whatever else appears important technically speak up now? -jiri At 12:08 25/01/2007, Olle E Johansson wrote: >As I am fighting the same issues and try to make chan_sip easier for >the administrator, would it be >a good idea with a joint Open Source SIP developer meeting somewhere >in Europe this spring? > >We have a lot of common issues, need to settle with the same >terminology and propably fight >the same fights with RFCs to implement... > >A lot of of users use both Asterisk and SER/OPENSER, so they need >interoperability. > >For my work on a new chan_sip, please check http://www.codename- pineapple.org > >/O >_______________________________________________ >Serdev mailing list >Serdev(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serdev -- Jiri Kuthan http://iptel.org/~jiri/

7 9

[Serusers] www_challenge() error in auth module
by raj gopal 02 Feb '07

02 Feb '07

Hi, I can not use www_challenge(realm,qop)function in ser.cfg file even though auth module is loaded. The readme file has documentation for it. When I use the following in config file and start SER, the following error is thrown if (!radius_www_authorize("")) { www_challenge("", "1"); break; }; parse error (197,49-50): unknown command, missing loadmodule? Can any one help me to find out whats wrong with this module/function. -Venkat

1 0

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