sr-users March 2005

sr-users@lists.kamailio.org

250 participants
627 discussions

[Serusers] multiple INVITE retransmissions
by Tina 30 Mar '05

30 Mar '05

Hello, I ran out into some problem when implementing Call Forwarding feature using SER. My test setup: 192.168.25.101 - XLite ---> INVITE UA(sip:444) 192.168.25.163 - ser; rewrites to Address of record of Asterisk (sip:1111 instead of sip:444) 192.168.25.169 - PBX Asterisk As I see from log files, SER sends INVITE 2 times, then Asterisk sends 3 times 200 status etc . until BYE is sent (here everything correct....) Here is my configuration script: ================================== route{ . . . record_route(); #--rewrite URI if (method=="INVITE"){ #--call forwarding setflag(1); route(1); #--cfw if(!isflagset(1)) { break; } lookup("location") ... }; route[1] { log(1, "forwarding\n"); if (!if_activated("to")) { #--continue regular call set up break; }; if (!lookup_dn()){ sl_send_reply("404","Not Found"); resetflag(1); #--continue! break; }; } Here if_activated checks whether the feature was activated; lookup_dn() rewrites URI to new UA and call append_branch() Any ideas what I am doing wrong? if (!t_relay()) { sl_reply_error(); }; } =========== Any help will be highly appreciated!!! Thanks, Tina, software engineer --------------------------------- Do you Yahoo!? Yahoo! Small Business - Try our new resources site!

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[Serusers] call forwarding - INVITE retransmissions after connection established
by Tina 30 Mar '05

30 Mar '05

Hello, I ran out into some problem when implementing Call Forwarding feature using SER. My test setup: 192.168.25.101 - XLite ---> INVITE UA(sip:444) 192.168.25.163 - ser; rewrites to Address of record of Asterisk (sip:1111 instead of sip:444) 192.168.25.169 - PBX Asterisk As I see from log files, SER sends INVITE 2 times, then Asterisk sends 3 times 200 status etc . until BYE is sent (here everything correct....) Here is my configuration script: ================================== route{ . . . record_route(); #--rewrite URI if (method=="INVITE"){ #--call forwarding setflag(1); route(1); #--cfw if(!isflagset(1)) { break; } lookup("location") ... }; route[1] { log(1, "forwarding\n"); if (!if_activated("to")) { #--continue regular call set up break; }; if (!lookup_dn()){ sl_send_reply("404","Not Found"); resetflag(1); #--continue! break; }; } Here if_activated checks whether the feature was activated; lookup_dn() rewrites URI to new UA and call append_branch() Any ideas what I am doing wrong? if (!t_relay()) { sl_reply_error(); }; } =========== __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Mediaproxy, problems in PSTN-to-SIP
by Lucas Aimaretto 30 Mar '05

30 Mar '05

Hi there, I have the following: SER: version: ser 0.9.0 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 16:46:51 Mar 29 2005 with gcc 2.96 CISCO GATEWAY: IOS (tm) 5300 Software (C5300-IS-M), Version 12.2(26), RELEASE SOFTWARE (fc2) The situation is the following: If I have to UAs, both behind same (linux-iptables) nat, I have a two-way audio communication, with nice quality, with no problems at all. If I place a call to the PSTN, trough my Gateway, from my UA, I have a two-way audio communication, with nice quality, with no problems at all. If I try to receive a call from the PSTN, trough my Gateway, I only have a one-way audio communication, but only in the SIP->PSTN direction. What could this be?. I can send my ser.cfg and some dumps from communications, but I believe the problem is quite clear. In fact, I followed the instructions from onsip.org. - some ngrep's ... U ip_ua:5070 -> ip_ser_mediaproxy:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP ip_ser_mediaproxy;branch=z9hG4 U ip_ser_mediaproxy:5060 -> snom_proxy:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP snom_proxy:5060;branch=z9 U snom_proxy:5060 -> ip_ser_mediaproxy:5060 ACK sip:ip_ser_mediaproxy:5060 SIP/2.0..v: SIP/2.0/UDP 38.117.2 U ip_ser_mediaproxy:5060 -> ip_ua:5070 ACK sip:1991015@ip_ua:5070 SIP/2.0..Record-Route: < The ACK arrives nicely to my UA (ip_ua:5070), but it keeps sending RTP to my GATEWAY's IP and port. - tcpdumping RTP on the UA (private IP:192.168.1.185.15090)(behind NAT) 20:36:37.596851 192.168.1.185.15090 > IP_CISCO.20482: udp 32 - show call active voice brief at my cisco: 2D2B : 104269100hs.1 +273 pid:1991334 Originate 1991015 active dur 00:00:36 tx:162/3202 rx:1790/35800 IP ip_ser_mediaproxy:35234 rtt:0ms pl:33170/140ms lost:0/0/0 delay:69/69/70ms g729r8 - session.py (from media proxy) (look at the ports, its fine and logic according to the tcpdumping of RTP at my LAN). Caller Via Called Status Duration Codec Type Traffic ------------------------------------------------------------------------ --------------------------- IP_CISCO:20482 - ip_ser_mediaproxy:35234 - ?.?.?.?:? inactive 1'04" G729 Audio 11.29k/0/0 I would appreciate some help ( or even IDEAS ) Best regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.4 - Release Date: 27/03/2005

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[Serusers] UPDATE: Getting Started - Issue 03a covers NAT Support
by Simon Miles 30 Mar '05

30 Mar '05

Dear All, We had found that the config files provided didn't work as expected, so we now have tested the mediaproxy version and modified it to ensure that is works. We have also updated the rtpproxy file to reflect these changes. So please download the new version 03a which we hope now works ! ! www.ONsip.org Sorry for the inconvenience but this is why we are a community working together. Thanks Paul, Simon, Greger

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[Serusers] consume_credentials breaks parallel forking using aliases
by Jev 30 Mar '05

30 Mar '05

Hi All, I have found that while using consume_credentials() in my ser config, when I make a call to a one to many alias, only the first URI in the alias will get sent a invite. Removing the consume_credentials() from my config allows parallel forking to work again, but with the Proxy-Authorization: header field still present. Is this a known issue? Thanks, -Jev

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[Serusers] SEMS and silence supresison
by Dave Bath 30 Mar '05

30 Mar '05

Hey guys, This might be the wrong list for this... but I'm trying to find a large scale multi-user conferencing system which supports silence suppression. Asterisk works well enough in terms of supporting a large number of users being able to connect to the same "room" simultaneously, but there are a lot of posts about it not supporting silence suppression... which from empirical evidence appears to be true! Does anyone know if SEMS properly supports silence suppression (i.e. in both directions... if no-one in the conference is saying anything (as unlikely as that might be.. then there's no data sent to UAs) ? On a slightly different note, having not looked into the code yet, does anyone have any ideas how easy it is to implement new codecs in SEMS? Once again, if I've posted to the wrong list, sorry about that.... I'm sure someone will redirect me.. Many thanks for all the work and useful help I've got from this list over the last year or so. Cheers! Dave

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[Serusers] RE: PSTN logic in SER
by Alex Vishnev 30 Mar '05

30 Mar '05

Ozan, Please reply to the list and not me directly. It would help if you provide the exact configuration and sip log/message dump of you successful and unsuccessful attempts. Then people can look and see if there is a problem in SER or just your routing implementation. Thanks Alex _____ From: Ozan Blotter [mailto:cosmocid@ispro.net.tr] Sent: Wednesday, March 30, 2005 12:21 PM To: avishnev(a)optonline.net Subject: PSTN logic in SER Estemeed Alex Vishnev, Now there's no error while starting SER, i've removed "{", it routes call but how. I use Zyxel Prestige P2002 ATA's as VoIP Devices. It has some usual properties like (This is original): SIP Number : 8334843600 SIP Local Port : 5060 SIP Server Address : 212.154.104.198 SIP Server Port : 5060 REGISTER Server Address : 212.154.32.154 REGISTER Server Port : 5060 SIP Service Domain : 212.154.32.154 If i do above configuration in my ATA's and for example i call back my cellular as 905353490056, SER routes call to PSTN Gateway since its prefix is 90. But this time when i call 8334454556 for example, there's no activity, it doesn't ring. I do not apply auth. on SER because all important information is kept on other SIP Server (Sysmaster's VoiceMaster) like billing,customer information etc. We're planning to give free SIP2SIP calls between our customers beginning with prefix 833 or 834. If i change all information above to SER (all 212.154.32.154 to 212.154.104.198) i can call other customers but this time when i try to call my cellular again as example it does not work, i think SER doesn't pass register details to other SIP Server which is acting as PSTN Guard. How can i arrange this ? Thanks again and sorry for taking your time. Ozan Blotter

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[Serusers] strip() problem
by Tina 30 Mar '05

30 Mar '05

Hello, I am trying to activate call forwarding using ser. When subscriber sends INVITE which starts from some prefix (say *22), I strip the prefix and write activation information to mysql. I use ser 08-14 ran into the following problem: =====ser.cfg: if (uri =~ "^sip:\*22") { log(1, "*22\n"); strip(3); route(1); ... route[1] { if (!exec_msg(' echo "SIP_HF_MAX_FORWARDS ${SIP_HF_MAX_FORWARDS}">>/tmp/cfwdebug.log; echo "SIP_HF_VIA ${SIP_HF_VIA}">>/tmp/cfwdebug.log; echo "SIP_HF_CSEQ ${SIP_HF_CSEQ}">>/tmp/cfwdebug.log; echo "SIP_HF_FROM ${SIP_HF_FROM}">>/tmp/cfwdebug.log; echo "SIP_ORUI ${SIP_ORUI}">>/tmp/cfwdebug.log; echo "SIP_HF_CONTENT_LENGTH ${SIP_HF_CONTENT_LENGTH}">>/tmp/cfwdebug.log; echo "SIP_TID ${SIP_TID}">>/tmp/cfwdebug.log; echo "SIP_DID ${SIP_DID}">>/tmp/cfwdebug.log; echo "SIP_RURI ${SIP_RURI}">>/tmp/cfwdebug.log; echo "SIP_HF_TO ${SIP_HF_TO}">>/tmp/cfwdebug.log; echo "SIP_OUSER ${SIP_OUSER}">>/tmp/cfwdebug.log; echo "SIP_HF_CALLID ${SIP_HF_CALLID}">>/tmp/cfwdebug.log; echo "SIP_SRCIP ${SIP_SRCIP}">>/tmp/cfwdebug.log; echo "SIP_HF_CONTENT_TYPE ${SIP_HF_CONTENT_TYPE}">>/tmp/cfwdebug.log; echo "SIP_HF_CONTACT ${SIP_HF_CONTACT}">>/tmp/cfwdebug.log')) { log(1, "script-echo fail\n"); } } now when I am looking to /tmp/cfwdebug.log, it contains: SIP_HF_MAX_FORWARDS 69 SIP_HF_VIA SIP/2.0/UDP 192.168.25.101:5060;rport;branch=z9hG4bK5410DAEAA0744FA0B2BD64F59D465EF8 SIP_HF_CSEQ 1591 INVITE SIP_HF_FROM matuka <sip:tuka@192.168.25.163>;tag=1660681736 SIP_ORUI sip:*226566548@192.168.25.163 SIP_HF_CONTENT_LENGTH 275 SIP_TID 3c8495de485acf4427dcfc7c5a0c0e8a SIP_DID SIP_RURI sip:6566548@192.168.25.163 SIP_HF_TO <sip:*226566548@192.168.25.163> SIP_OUSER *226566548 SIP_HF_CALLID 50748A60-1D39-470B-B9CC-84CCC32B67BD(a)192.168.25.101 SIP_SRCIP 192.168.25.101 SIP_HF_CONTENT_TYPE application/sdp SIP_HF_CONTACT <sip:tuka@192.168.25.101:5060> Any help highly aprreciated! Tina, software engineer --------------------------------- Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more. --------------------------------- Do you Yahoo!? Make Yahoo! your home page

2 1

[Serusers] flatstore
by Steve Blair 30 Mar '05

30 Mar '05

Can someone point me toward documentation about activating the flatstore module? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

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[Serusers] PA module help?...
by serusers＠jtoga.org 30 Mar '05

30 Mar '05

Hi - Trying to get a handle on how to get presence working in SER/pa module. - Does some kind soul have an ngrep output of a typical subsribe-ok-notify-ok exchange with a UA and SER? Even better would be some additional notifies with updates? I think mine is broken (no 'status' section in the notify) and not sure if/why...... - Is there supposed to be tables in mysql related to this presentities? (assuming that mysql is fully engaged), I don't seem to have any nor can I find their creation in a db init script. Thanks in advance.

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sr-users March 2005