sr-users June 2013

sr-users@lists.kamailio.org

94 participants
170 discussions

by Sunil Chandrasekharan

Hi All, I am trying to install Kamailio on Amazon EC2. I read out in web about some people already achieved configuring Kamailio on EC2. Kindly support on the pre-requisties to be installed before i go with Kamailio installation.. My requirements ; Kamailio to be installed with Presence module. In the config file , i see that we have to use define with NAT and install RTP proxy. Is it required? Is there any module to be added to get MSRP working, so that i can share a file. Kindly support to get Kamailio running with register,presence,IM and MSRP on Amazon EC

10 years, 10 months

481 dialog/transaction does not exists Error.

by Nandini madhu

Dear All, I have integrated Kamailio (Version 4.0.1) with Asterisk (Version 11.4.0 rc1) server. I should have get VoIP features(call, SMS) with this integrated server. But when i tried to call other end SIP client through this integrated server, I'm getting this '*481 dialog/transaction does not exists*' error (please find the attachment 'call.pcap' below). SMS is through with the kamailio server without any problem (Find the attachment 'SMS.pcap' below). But when it comes to Calling feature, something is wrong and i am unable to find it. So please help me in resolving this issue. My system configuration is as follows: Host Architecture: i686 OS : Ubuntu 12.04 LTS Clients: Wi-Fi enabled VoIP phones. Find the attachments below for the kamailio configuration file and tcpdump based trace file for Calling to give a more complete picture. Any help will greatly appreciate. *Best regards, * Nandini

10 years, 10 months

'481 dialog/transaction does not exists'

by Ravindra Gowda

Dear All, I have integrated Kamailio (Version 4.0.1) with Asterisk (Version 11.4.0 rc1) server. I should have get VoIP features(call, SMS) with this integrated server. But when i tried to call other end SIP client through this integrated server, I'm getting the '*481 dialog/transaction does not exists*' error (please find the attachment 'call.pcap' below). SMS is through with the kamailio server without any problem (Find the attachment 'SMS.pcap' below). But when it comes to Calling feature, something is wrong and i am unable to find it. So please help me in resolving this issue. My system configuration is as follows: Host Architecture: i686 OS : Ubuntu 12.04 LTS Clients: Wi-Fi enabled VoIP phones. Find the attachments below for the kamailio configuration file and tcpdump based trace files for Calling and SMSing process, to give a more complete picture. Any help will greatly appreciate. *Thank you and Best regards,* *R Ravindra **Gowda*

10 years, 10 months

Re: [SR-Users] Does Kamailio provide any API for other program tocreate sip account

by cxw1987

10 years, 10 months

Re: [SR-Users] Does Kamailio provide any API for other program to create sip account

by Xiangwen Chen

Thanks David, My user account system is written in Python, can "kamctl" script be called in a python program? if not, can the user account system directly access the kamailio database, and do an SQL insert to create a new sip account? Thanks in advance!

10 years, 10 months

Does Kamailio provide any API for other program to create sip account

by Xiangwen Chen

Hello, I am developing a sip VoIP app in IOS, which want to provide the feature of allowing user to create sip account from client side. I use Kamailio as sip server. My thought is when user register to the the user account system, the user account system can communicate with Kamailio, create a sip account associated with the user and pass the sip account to the client, the client then can configure itself using the sip account. Does Kamailio provide any API or module for me to do this? Thanks!

10 years, 10 months

dp_translate() pseudovariables

by Victor V. Kustov

Hi all! can I use $dlg_var() in dp_translate()? ok, in common form: why some functions fail to use $dlg_var()? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.9.6-calculate GNU/Linux

10 years, 10 months

Re: [SR-Users] Does Kamailio provide any API for other program to create sip account

by cxw1987

10 years, 10 months

SIP Trunks Location

by Jose Suero

Hi I'm planning to set kamailio in front of an farm of pbx servers (haven't decided on freeswitch or asterisk) there's a million tutorials on how to do this, what I haven't found is what part of my setup actually handles the sip trunks my phone company provides me with. What's the best practice when It comes to this? Is kamailio going to be receiving the calls from the trunk and passing them to the PBX or is it the other way around? please advice Thanks in advance Jose Suero

10 years, 10 months

Kamailio Crashing with excessive Via headers

by Krishna Kurapati

In 4.x, Following message is crashing the Kamailio. MESSAGE sip:50.62.12.174:15060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 50.62.72.174:15060;branch=z9hG4bK3333.defaafa4.0..Via: SIP/2.0/UDP 50.62.72.174:15060;branch=z9hG4bK3333.cefaafa4.0.. Via: SIP/2.0/UDP 50.62.12.174:15060;branch=z9hG4bK3333.befaafa4.0.. Via: SIP/2.0/UDP 50.62.12.174:15060;branch=z9hG4bK3333.aefaafa4.0.. Via: SIP/2.0/UDP 50.62.12.174:15060;branch=z9hG4bK33339efaafa4.0.. Via: SIP/2.0/UDP 50.62.12.174:15060;branch=z9hG4bK3333.8efaafa4.0.. Via: SIP/2.0/UDP 50.62.12.174:15060;branch=z9hG4bK3333.7efaafa4.0.. Via: SIP/2.0/UDP 50.62.12.174;branch=z9hk4bK3333.4b488206.0.. Via: SIP/2.0/UDP 50.62.12.174;rport=5060;branch=z9hG4bK3333.3b488206.0.. To: sip:578626051@50.62.12.174.. From: sip:notifier@abc.com;tag=5571efee096a394dda7d0dffc5bb 32a5-c771..CSeq: 10 MESSAGE..Call-ID: presence-T[578626051]-R[578626051]..Max-Forwards: 9..Co ntent-Length: 110..User-Agent: kamailio (4.0.1 (x86_64/linux))..Content-Type: text/html; chars et=utf-8. stack trace: #0 qm_detach_free (qm=0x2ac39d748010, size=64) at mem/q_malloc.c:269 #1 qm_malloc (qm=0x2ac39d748010, size=64) at mem/q_malloc.c:386 #2 0x0000000000560fa1 in parse_via ( buffer=0x8c6a3b "SIP/2.0/UDP 50.62.72.174:15060;branch=z9hG4bK3333.defaafa4.0\r\nVia: SIP/2.0/UDP 50.62.72.174:15060;branch=z9hG4bK3333.cefaafa4.0\r\nVia: SIP/2.0/UDP 50.62.72.174:15060;branch=z9hG4bK3333.befaafa4.0\r\nVia:"..., end=0x8c6e31 "", vbody=0x2ac39d769e90) at parser/parse_via.c:2540 #3 0x0000000000538e68 in get_hdr_field (buf=<value optimized out>, end=0x8c6e31 "", hdr=0x2ac39d7999c0) at parser/msg_parser.c:140 #4 0x0000000000539ac9 in parse_headers (msg=0x2ac39d78a100, flags=2, next=<value optimized out>) at parser/msg_parser.c:351 #5 0x000000000053af8d in parse_msg (buf=<value optimized out>, len=<value optimized out>, msg=0x2ac39d78a100) at parser/msg_parser.c:650 #6 0x000000000049a2ef in receive_msg ( buf=0x8c6a00 "MESSAGE sip:50.62.72.174:15060;transport=udp SIP/2.0\r\nVia: SIP/2.0/UDP 50.62.72.174:15060;branch=z9hG4bK3333.defaafa4.0\r\nVia: SIP/2.0/UDP 50.62.72.174:15060;branch=z9hG4bK3333.cefaafa4.0\r\nVia: SIP/2.0"..., len=1073, rcv_info=0x7fffb40ab770) at receive.c:144 #7 0x0000000000528666 in udp_rcv_loop () at udp_server.c:557 #8 0x0000000000464daa in main_loop () at main.c:1638 #9 0x0000000000467adc in main (argc=<value optimized out>, argv=<value optimized out>)

10 years, 10 months

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sr-users June 2013