sr-users December 2011

sr-users@lists.kamailio.org

64 participants
108 discussions

Bug: Dialog State=*Teminated* even when the call is active
by Gnaneshwar Gatla 17 Dec '11

17 Dec '11

Hello, I have come across this behavior in Kamailio 3.1.5 version. The proxy is behind a Firewall and uses the advertised IP to communicate to sipclients. I have used the "dlg_match_mode", as the record_route_preset() does not include the DID during call initiation. modparam("dialog", "dlg_match_mode", 2) I'm also using pua_dialoginfo to notify users about the "Event:dialog". I have used the attached patch to make pua_dialog info aware of the dialog I see that the dialog is being created, and the pua_dialoginfo publishes with state "trying" and "confirmed". When the call is cancelled/408 request is generated, the pua_dialoginfo generates the xml with state=terminated. But when the call is answered, the dialog module updates the dialog to state=4, according to the documentation mentioned below state=4 --> Terminated. I'm not aware of the event types that fires the state to termination (I have not found the documentation about events in the dialog module). When a BYE is received to the same call, the BYE is processed and the dialog is cleared from the memory. But the pua_dialoginfo never gets the callback for the termination and the xml is never updated. I have attached a log for a call, please find the attached document. I have come across a documentation about dialog module: http://www.kamailio.org/dokuwiki/doku.php/modules-new-design:dialog-module-… Regards Gnaneshwar

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Sending SIP request from mi_fifo using TMX module
by Sammy Govind 16 Dec '11

16 Dec '11

Hello list, I'm using Kamailio 3.2 and am trying to send a SIP request from fifo to kamailio using below format. #kamctl fifo t_uac_dlg MESSAGE sip:103@ mydomain.org . . "From: 101(a)mydomain.org To: 103(a)mydomain.org Contact: mydomain.org Content-Type: text/plain; charset=UTF-8" "INVITE" But I see this reply from Kamctl *database engine 'MYSQL' loaded* *Control engine 'FIFO' loaded* *entering fifo_cmd t_uac_dlg INVITE sip:103@ mydomain.org . . From: 101@ mydomain.org To: 103@ mydomain.org Contact: mydomain.org Content-Type: text/plain; charset=UTF-8 INVITE* *400 Bad headers* *FIFO command was:* *:t_uac_dlg:openser_receiver_25561* *INVITE* *sip:103@mydomain.org* *.* *.* *From: 101@ mydomain.org To: 103@ mydomain.org Contact: mydomain.org Content-Type: text/plain; charset=UTF-8* *INVITE* And in log file I see this output: Dec 16 00:57:46 Kamailio3 /usr/local/sbin/kamailio[24541]: ERROR: <core> [parser/msg_parser.c:275]: ERROR: get_hdr_field: bad body for <From: 101(a)mydomain.org To: 103@ mydomain.org Contact: mydomain.org Content-Type: text/plain; charset=UTF-8INVITE#012>(4) Dec 16 00:57:46 Kamailio3 /usr/local/sbin/kamailio[24541]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [From: 101(a)mydomain.o] Can anyone help.. Regards, Sammy

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source port used for forwarded requests
by Ben WIlliams 16 Dec '11

16 Dec '11

Hi, My Kamailio server listens on multiple IPs - say IP1, IP2. If a request arrives on one IP, Kamailio uses the same IP as the source address when forwarding, which is good. But is there any way a UA could generate a request to IP1 that gets forwarded by Kamailio with source address IP2 ? The reason I ask is because Asterisk is setup to trust all calls arriving from IP2 but not IP1. Thanks Ben

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How to get dispatcher list with sercmd command
by Mino Haluz 16 Dec '11

16 Dec '11

Hi, # sercmd dispatcher.list error: 500 - command dispatcher.list not found The module is loaded, dispatcher.list file exists. Am I doing something wrong ? Thanks. Mino.

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SERi - SIP Express Router setup file corrupted
by Tirtha 16 Dec '11

16 Dec '11

Dear Sir, To configure a SIP server in windows platform, I have downloaded the setup file of SIP Express Router from your FTP server as mentioned in your web portal, but unfortunately when I am going to extract that file , a error is coming and showing that the specific file is corrupted. So requesting you please help me in this regard with the proper installation guide line. Thank you. Regards, http://www.matrixventures.com/images/green.jpg Tirtha Ghosh Asst. Manager IT M +91 8017372776 T +91 33 40624172 E <mailto:tirtha.ghosh@matrixinfrastructure.com> eMail Me Matrix Ventures Matrix logo <http://in.linkedin.com/in/tirthaghoshkolkata> LinkedIn | <http://www.facebook.com/home.php#!/profile.php?id=1385652529> Facebook | http://www.matrixventures.com/images/blue.jpg http://www.matrixventures.com/images/blue.jpg Matrix Ventures Pvt Ltd . Saltlake, Sector 1 . Kolkata 700064 . India

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How to enable rtp repacketization
by Arif Tuhin 15 Dec '11

15 Dec '11

I want to enable the rtp repacketization feature of the rtpproxy. I have set force_rtp_proxy("z150"). But the repacketization is not happening. in the log it does show that rtp will be repacketizes but when i see the wireshark capture, its the same. With Best Regards Ariful Hossain Tuhin email: 1. etothepowerpi(a)gmail.com 2. etothepowerpi(a)hotmail.com 3.etothepowerpi(a)yahoo.com skype: freeburn1986

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Kamailio - Asterisk Integration
by Anthony Sanchez 15 Dec '11

15 Dec '11

Is there an update of default Kamailio 3.2.1 configuration file to integrate with Asterisk? Thanks in advance, Tony

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Kamailio - Asterisk Integration
by Anthony Sanchez 15 Dec '11

15 Dec '11

Is there an update of default Kamailio 3.2.1 configuration file to integrate with Asterisk? Thanks in advance, Tony

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Siremis v3.2.0 Released
by Elena-Ramona Modroiu 14 Dec '11

14 Dec '11

Hi, Siremis v3.2.0 is out - the web management interface for Kamailio and SER v3.2.x series. Besides updates to existing views to match the database schema in Kamailio v3.2.x (such as lcr, dialog or siptrace), there are many new features. Among most important: - SQL-based CDR rating engine for billing purposes, with views to manage the rating rules - charts and table summaries to reflect the activity of accounting records - new views to manage records for remote registrations, mtree and global black list - command line tools to add new database table views in a step-by-step wizard fashion - selection of usernames and domains for many views can be done by a select box or picker form with values from subscriber or domain tables More details can be find in the announcement of the release on the web site: * http://asipto.com/u/4t or in the ChangeLog inside the tarball. Siremis can be downloaded from: * http://asipto.com/u/sd Alternative download site, along with the GIT repository for Siremis, is offered at sourceforge.net project: * http://sourceforge.net/projects/siremis/ Regards, Ramona

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Double Invite / Issue
by Brandon Armstead 14 Dec '11

14 Dec '11

Hello, I am running into a problem where I am experiencing duplicated INVITE's being originated from Kamailio proxy. U 2011/12/13 23:58:17.543802 KAMAILIO:5060 -> PSTN:5060 INVITE sip:URI@PSTN SIP/2.0. U 2011/12/13 23:58:17.543840 KAMAILIO:5060 -> PSTN:5060 INVITE sip:URI@PSTN SIP/2.0. As you can see that the timestamp between the two invites is literally a fraction of seconds. There is no failure condition that is causing this with append_branch There is no retransmission timer issue that is causing this (using default timing): modparam("tm", "retr_timer1", 500) modparam("tm", "retr_timer2", 4000) Any thoughts / ideas - as this is causing a race condition in which there is a 200 OK that is being sent back from upstream -- and we are then CANCELING the 2nd INVITE which is essentially causing on overall problem with the call. Thanks for all help / thoughts / input in advance, thanks! Sincerely, Brandon Armstead

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sr-users December 2011