sr-users December 2012

sr-users@lists.kamailio.org

71 participants
113 discussions

by Mino Haluz

Hi, what things should I set in order to set my custom RPID. What I am doing: remove_hf("Remote-Party-ID"); append_rpid_hf("", ";party=calling;id-type=subscriber;screen=no"); but RPID is only removed, second command does not append any rpid. Thanks Mino

12 years

New developer: Konstantin Mosesov

by Daniel-Constantin Mierla

Hello, I want to announce that a new developer got GIT write access to repository: Konstantin Mosesov - he has contributed patches to app_python and joins the team to help maintaining and developing this module. His git commit id is: ez My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

12 years

How to use LIMIT 1 in db_query api

by Krishna Kurapati

Hi, I need a way to limit the results to just one row. The only way I know I could do is to call db_query followed by fetch_result of 1 row. However this is not optimal as all the rows are fetched to dynamic memory with db_query. Is there an alternative approach? Thanks Krish Kura

12 years

Help

by ameneh pourmoghadas

Dear Admin, My name is Ameneh from India. I am new in SIP. I have installed asterisk 1.8 on my ubuntu system and could have registered the internal users in my LAN to talk over x-lite. Now my question is how can I make a call to a mobile in India or a PSTN number in another country that would require a public SIP I think. how can I make my SIP server as public? would you please help me? thanks and regards, Ameneh

12 years

CTL module and alternatives for remote execution mi commands.

by Pavel Zalunin

Hi. In our project we need to query current users, dialogs and ability to end dialog using some rpc. Requirements are - do this remotely. I already tried to use xmlrpc module. When we had 42 users online it returned 17kb broken xml file. Then I tried to use binrpc protocol from CTL module. Usage ends with returning broken message too (in case, when message is biger than 21kb). Are there any other more stable solutions to query mi information using some tcp protocol? P.S. When I discovered sources of CTL module I found usage of writev - here is the problem about partial packets, when buffer is longer, than 21kb it don't pushes all vector buffers into socket. And in general, this module is using async sockets, AFAIK writev don't gives any guarantees, that it will write all vectors in async sockets. P.P.S. Of couse, last resort solution will be writing proxy for using mi_fifo on kamailio side to forward all records to other servers, but if there are more pretty solutions, please give me know:) Thanks. Pavel.

12 years

OFFTOPIC - Realtime testers needed - Asterisk Path header support (Oolong branch)

by Olle E. Johansson

While this is not Kamailio - it's related. If you want to use Kamailio in front of Asterisk, you do need SIP Path header support in Asterisk to do it right. Especially if you have one Kamailio handling Internet-facing communication and one handling internal communication. Please test! /O Vidarebefordrat brev: > Från: "Olle E. Johansson" <oej(a)edvina.net> > Ämne: Realtime testers needed - Path header support (Oolong branch) > Datum: 7 december 2012 11:07:09 CET > Till: Asterisk Developers Mailing List <asterisk-dev(a)lists.digium.com> > Kopia: "Olle E. Johansson" <oej(a)edvina.net> > > Friends, > > I have taken Klaus Darillion's patch for Path header support in Asterisk and created branches for 1.4, 1.8 and trunk. I need help testing it with realtime support by someone who are more used to the realtime databases than I am. I also need help completing the LDAP schemas with the new attribute. > > General testing is also encouraged. > > I have been using this patch without realtime for a couple of months. So far no Squirrels where hurt in the progress. And no bugs hit me in the back. > > Read more about the patch here: > http://svnview.digium.com/svn/asterisk/team/oej/oolong-path-support-trunk/R… > > Klaus' original review: > https://reviewboard.asterisk.org/r/991/ > > The SIP Path header support is documented in RFC 3327 (that has a strange title). > > Basically, this is used when you have a proxy between a phone and your Asterisk. Asterisk needs to remember the path to the device when sending an INVITE, so we save both the contact and the path (basically a route) to the phone when Asterisk receives a REGISTER. When sending an INVITE (or anything based on the registry) a route header is added and the message finds the proper way to the device. > > Path header support exists in the path module in Kamailio if you want to test. > http://kamailio.org/docs/modules/3.3.x/modules_k/path.html > > Looking forward to your feedback! > > /O

12 years

Kamailio\ Message 300 Contacts

by Fatima Chahrour~Vanrise Support

Dears, In the redirect message 300, is there any way to control the maximum number of contacts(IP's) returned in the message? (am only getting 5 IP's) Also, if I have one gateway with 5 IP's and trying the same call 3 times; the order of the IP's returned in the message 300 is different in each time. Why is that? Thanks in advance, F Chahrour

12 years

Kamailio 3.X.X + BLF configuration trouble

by Hemanshu Patel

Hello everyone, Till now we were working with asterisk to handle BLF. But then we decided to move everything to kamailio and hence i started working on BLF with kamailio. I have already configured kamailio for Call handling and presence information. >From different pages on Internet, I have found some information about configuration of kamailio+blf. And i did so, but somehow things are not working as they should. I have done following changes to kamailio.cfg file loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_mwi.so" loadmodule "dialog.so" loadmodule "pua.so" loadmodule "pua_dialoginfo.so" modparam("presence", "fallback2db", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_url", "mysql://xxxxxxxxx:xxxxxxxxxxxxxx@localhost/openser") modparam("dialog", "db_mode", 1) modparam("pua", "db_url", "mysql://xxxxxxxxx:xxxxxxxxxxxxxx@localhost/openser") Now when my phone(yealink VP530), sends subscribe request it looks as below: SUBSCRIBE sip:1001@172.16.27.66 SIP/2.0 Via: SIP/2.0/UDP 172.16.27.61:5063;branch=z9hG4bK10464096 From: "1000" <sip:1000@172.16.27.66>;tag=393923564 To: <sip:1001@172.16.27.66> Call-ID: 1572901975(a)172.16.27.61 CSeq: 1 SUBSCRIBE Contact: <sip:1000@172.16.27.61:5063> Accept: application/dialog-info+xml Max-Forwards: 70 User-Agent: VP530P 23.70.0.40 Expires: 120 Event: dialog Content-Length: 0 In response to this message, Kamailio returns 202 with following information SIP/2.0 202 OK Via: SIP/2.0/UDP 172.16.27.61:5063;branch=z9hG4bK245273058 From: "1000" <sip:1000@172.16.27.66>;tag=393923564 To: <sip:1001@172.16.27.66>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8ee2 Call-ID: 1572901975(a)172.16.27.61 CSeq: 2 SUBSCRIBE Expires: 120 Contact: <sip:172.16.27.66:5060> Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Followed by a NOTIFY NOTIFY sip:1000@172.16.27.61:5063 SIP/2.0 Via: SIP/2.0/UDP 172.16.27.66;branch=z9hG4bK3ccd.adfbc416.0 To: sip:1000@172.16.27.66;tag=2058189864 From: sip:1001@172.16.27.66;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6cd8 CSeq: 1 NOTIFY Call-ID: 2040123403(a)172.16.27.61 Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:172.16.27.66:5060> Subscription-State: active;expires=170 During subscribe message asked for XML based reply, yet NOTIFY doesnt send any XML. and hence there is no impact of this message on device. Nothing is changed.Also there's nothing when i call from this device to another, ideally i should get some NOTIFY messages So, i think i am doing something wrong in Configuration. Can anyone please help me in it. If you need any other info, i'll provide you. -- Regards, Hemanshu Patel Senior Software Engg P Help the environment – please don't print this email unless you really need to!

12 years

Internal/External SIP setup

by Zero Aggression

Hello, I am a bit new to VoIP, and I just need a quick yes/no (and maybe pointers) answer. Is the following possible (and smart) to do with Kamailio? Kamailio server. Dual-homed (internal network, Internet). Internal users connect to Kamailio to make calls to internal and Internet SIP peers. Users have NO internet connection themselves (they can only reach the server). Can Kamaillio (when dual-homed) transport all the data/voice between Internet and intranet? Can it also be used to carry all traffic between intranet users (if there is no direct route between them)? Thanks so much. Zane

12 years

db_cluster configuration

by Juan José Ivars

Hello, i am configuring db_cluster module in kamailio 3.3 I have configured in roun-robin read between two MySQL servers but the load is not distributed equally My configuration is that: modparam("db_cluster", "connection", "con1=>mysql://USER1:PASS1@1.1.1.1/DB") modparam("db_cluster", "connection", "con2=>mysql://USER2:PASS2@2.2.2.2/DB") modparam("db_cluster", "cluster", "cls1=>con1=9r9p;con2=9r9p") And the auth module with: modparam("auth_db", "db_url", "cluster://cls1") But i have enabled the mysql logs and the major part of the querys are targeted to the second server Is there any cache by peer? What am i doing wrong? Thanks in advance.

12 years

← Newer
1
...
6
7
8
9
10
11
12
Older →

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 December 2012