sr-users February 2005

sr-users@lists.kamailio.org

215 participants
467 discussions

[Serusers] sequential forking support
by Juha Heinanen 02 Mar '05

02 Mar '05

i just committed some patches to lcr module that introduced two new functions: load_contacts() and next_contacts(). they can be used to implement sequential forking (see README for details). if all contacts in the destination set have the same qvalue, load_contacts() and next_contacts() don't do anything, just check that nothing needs to be done. -- juha

3 8

[Serusers] another cancel test
by Juha Heinanen 02 Mar '05

02 Mar '05

andres, i made another simpler cancel test. first sip:jh@test.fi invites sip:gs@test.fi, and gs starts ringing. then jh sends cancel to r-uri sip:gs@test.fi, which ser.cfg immediately at the start of the script t_relay()s out, i.e., without doing anything to the request. the result is that ser correctly sends a cancel to the ip address/port of gs. my conclusion of this is that cancels don't need to be processed by ser.cfg, just relayed out. perhaps others familiar with tm mystics can confirm or deny this. -- juha

4 8

[Serusers] PHP5 - aggregate_methods() missing, serweb failing
by Dag Bakke 01 Mar '05

01 Mar '05

It appears that aggregate_methods() is missing in PHP5. Thus, I get: Fatal error: Call to undefined function aggregate_methods() in /opt/serweb/html/data_layer.php on line 87 when trying to access the user or admin interface of serweb. This is with serweb cvs as of yesterday and php 5.0.3. I have no idea if there are more issues related to php5 and serweb. But is there a plan/chedule for making serweb working with php5? Or should I just downgrade to php4? Other projects have hit this problem as well: https://sourceforge.net/tracker/?group_id=92482&atid=600780&func=detail&aid… Thank you to everyone contributing to ser and serweb! Dag B.

4 5

[Serusers] Help - ser won't start while installing
by Dana Olson 01 Mar '05

01 Mar '05

I downloaded the 0.8.14 Debian packages and tried installing them. I know that they work, since they worked fine on my other system, but this one (which is almost identical, save for the NIC) doesn't work. Here is what happens in the console: # dpkg -i ser_0.8.14_i386.deb Selecting previously deselected package ser. (Reading database ... 10117 files and directories currently installed.) Unpacking ser (from ser_0.8.14_i386.deb) ... Setting up ser (0.8.14) ... Restarting ser: ser It just hangs there forever. I checked the end of /var/log/messages but there are no messages from ser there. I would try reinstalling Debian from scratch, but the server itself is in another country, and I'm trying to install SER remotely over SSH. I've tried rebooting which didn't help (although I didn't expect it would). I searched on Google and the closest thing I could find was invalid syntax in the ser.cfg script, but this is the initial package install... Please, please help me out. -- Dana

1 3

[Serusers] Turn 3xx into new Invite
by Martin Koenig 01 Mar '05

01 Mar '05

Hello, is it possible to take 3xx-responses in a reply_route and turn them into a new INVITE on Ser using append_branch or simliar? Regards, Martin

3 3

[Serusers] HELP: SER-0.9 Record Route Problem Between SER and Asterisk
by Java Rockx 01 Mar '05

01 Mar '05

Hi All. I'm using ser-0.9 Can anyone take a quick look at this short SIP conversation and tell me if they think the problem is with my ser.cfg or a bug in Asterisk 1.0.2. We use a 3rd party for PSTN gateway functionality. This 3rd party uses a Sonus box behind a SIP proxy. Our SER proxy talks directly to their SIP proxy as needed to complete PSTN calls. The problem is that when a PSTN caller dials a SIP phone and gets sent to voice mail (Asterisk) because of a no answer or busy condition, Asterisk hangs up after the caller leaves a message. When Asterisk hangs up, the BYE from Asterisk is sent to SER, however, SER incorrectly forwards the BYE directly to their Sonus gateway, rather than the their SIP proxy. This causes our PSTN gateway provider to have "open" billing records in their system. If you look at the BYE message from Asterisk to SER you can see that route headers are missing (I think). The final BYE should have been sent to 68.86.100.20, but it was sent to 68.86.100.30 instead. I am record_route()ing all messages except for REGISTER and I have the mhomed=1 parameter set. Can anyone help me put the blame on either my ser.cfg or Asterisk? Regards, Paul IP LEGEND ----------- 68.86.100.30 - 3rd Party Sonus PSTN Gateway 68.86.100.20 - 3rd Party SIP Proxy 24.11.12.24 - Sip Express Router (eth0) 10.255.255.1 - Sip Express Router (eth1) 10.255.255.2 - Asterisk PBX NOTE: I have Asterisk connected to the SER server with a crossover cable. U 2005/02/23 22:24:18.848582 68.86.100.20:5060 -> 24.11.12.24:5060 INVITE sip:4075551212@24.11.12.24:5060 SIP/2.0. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. To: 4075551212 <sip:4075551212@68.86.100.30:5060>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. Max-Forwards: 4. Contact: sip:3211231234@68.86.100.30:5060. Record-Route: <sip:68.86.100.20:5060;lr>. Content-Type: application/sdp. Content-Length: 312. . v=0. o=NexTone-MSW 1234 187 IN IP4 66.236.245.98. s=sip call. c=IN IP4 66.236.245.98. t=0 0. m=audio 16814 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=yes. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. # U 2005/02/23 22:24:18.860022 24.11.12.24:5060 -> 68.86.100.20:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. To: 4075551212 <sip:4075551212@68.86.100.30:5060>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. Content-Length: 0. . # U 2005/02/23 22:24:18.860259 10.255.255.1:1033 -> 10.255.255.2:5060 INVITE sip:699@10.255.255.2:5060 SIP/2.0. Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Via: SIP/2.0/UDP 10.255.255.1;branch=z9hG4bKb929.21080974.0. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. To: 4075551212 <sip:4075551212@68.86.100.30:5060>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. Max-Forwards: 3. Contact: sip:3211231234@68.86.100.30:5060. Record-Route: <sip:68.86.100.20:5060;lr>. Content-Type: application/sdp. Content-Length: 312. . v=0. o=NexTone-MSW 1234 187 IN IP4 66.236.245.98. s=sip call. c=IN IP4 24.11.12.24. t=0 0. m=audio 36574 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=yes. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. # U 2005/02/23 22:24:18.871131 10.255.255.2:5060 -> 10.255.255.1:1033 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=1033. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:699@10.255.255.2>. Content-Length: 0. . # U 2005/02/23 22:24:18.879160 10.255.255.2:5060 -> 10.255.255.1:1033 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=1033. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:68.86.100.20:5060;lr>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:699@10.255.255.2>. Content-Type: application/sdp. Content-Length: 362. . v=0. o=root 550 550 IN IP4 10.255.255.2. s=session. c=IN IP4 10.255.255.2. t=0 0. m=audio 17900 RTP/AVP 97 18 3 4 2 0 8 101. a=rtpmap:97 iLBC/8000. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U 2005/02/23 22:24:18.883882 24.11.12.24:5060 -> 68.86.100.20:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:68.86.100.20:5060;lr>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:699@10.255.255.2>. Content-Type: application/sdp. Content-Length: 363. . v=0. o=root 550 550 IN IP4 10.255.255.2. s=session. c=IN IP4 24.11.12.24. t=0 0. m=audio 36574 RTP/AVP 97 18 3 4 2 0 8 101. a=rtpmap:97 iLBC/8000. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U 2005/02/23 22:24:19.097436 68.86.100.20:5060 -> 24.11.12.24:5060 ACK sip:699@10.255.255.2 SIP/2.0. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=ef5ce1f8a400adab242436d0564cb045. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=ef5ce1f8a400adab242436d0564cb045. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 ACK. Max-Forwards: 4. Contact: sip:3211231234@68.86.100.30:5060. Record-Route: <sip:68.86.100.20:5060;lr>. Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Content-Length: 0. . # U 2005/02/23 22:24:19.098087 10.255.255.1:1033 -> 10.255.255.2:5060 ACK sip:699@10.255.255.2 SIP/2.0. Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Via: SIP/2.0/UDP 10.255.255.1;branch=0. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=ef5ce1f8a400adab242436d0564cb045. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=ef5ce1f8a400adab242436d0564cb045. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 ACK. Max-Forwards: 3. Contact: sip:3211231234@68.86.100.30:5060. Record-Route: <sip:68.86.100.20:5060;lr>. Content-Length: 0. . ### U 2005/02/23 22:24:25.104860 10.255.255.2:5060 -> 10.255.255.1:1033 BYE sip:3211231234@68.86.100.30:5060 SIP/2.0. Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport. Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>,<sip:68.86.100.20:5060;lr>,<sip:3211231234@68.86.100.30:5060>. From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Contact: <sip:699@10.255.255.2>. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 102 BYE. User-Agent: Asterisk PBX. Content-Length: 0. . # U 2005/02/23 22:24:25.108961 24.11.12.24:5060 -> 68.86.100.30:5060 BYE sip:3211231234@68.86.100.30:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:24.11.12.24;r2=on;ftag=as588114d9;lr=on>. Record-Route: <sip:10.255.255.1;r2=on;ftag=as588114d9;lr=on>. Via: SIP/2.0/UDP 24.11.12.24;branch=z9hG4bK231b.07949ea5.0. Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060. From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Contact: <sip:699@10.255.255.2>. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 102 BYE. User-Agent: Asterisk PBX. Content-Length: 0. Route: <sip:3211231234@68.86.100.30:5060>. . # U 2005/02/23 22:24:25.175832 68.86.100.30:5060 -> 24.11.12.24:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 24.11.12.24;branch=z9hG4bK231b.07949ea5.0. Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060. From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 102 BYE. Content-Length: 0. . # U 2005/02/23 22:24:25.176182 10.255.255.1:1033 -> 10.255.255.2:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060. From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 102 BYE. Content-Length: 0. .

4 6

[Serusers] Which SIP messages types use lookup("location")?
by Java Rockx 01 Mar '05

01 Mar '05

Hi All, Can anyone tell me for which message types I should call lookup("location") before t_relay()? ACK BYE CANCEL INVITE NOTIFY REFER SUBSCRIBE

7 10

[Serusers] Can ser Presence Agent support windows messenger 5.0 or above?
by Jeffrey Zhou 01 Mar '05

01 Mar '05

Hi, Can ser Presence Agent support windows messenger 5.0 or above? It seems i can not see the on line contact with windows messenger 5.0 with ser

2 1

[Serusers] fifo db
by AJ Grinnell 01 Mar '05

01 Mar '05

Is the command fifo_db_url meant to enable access to all tables in the ser database or just certaian tables such as grp? Any changes that I make to the grp table take effect immediately, but if I make a change to aliases or domain, SER requires a restart.

2 1

[Serusers] Can I modify a transaction already in progress?
by Paul Belanger 01 Mar '05

01 Mar '05

I need some help understanding transactions, I know how to start transactions, are create a simple stateful proxy. But I'm having some troubles understanding how I can modify a current transaction. Everytime I try I always get the error: 0(10257) ERROR: t_newtran: transaction already in process Which leads me to believe once the transaction is created, I can't modify it. Is this true? ------------------------------- Paul Belanger (mailto:paul.belanger@pronexus.com) Technical Support Specialist Cisco Certified Network Associate Pronexus Inc. - A Powerful Voice in Communication Solutions ------------------------------- Tel: 613.271.8989 ext. 516 Fax: 613.271.8388 http://support.pronexus.com -------------------------------

2 1

Jump to page:

2026

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users February 2005