sr-users June 2013

sr-users@lists.kamailio.org

94 participants
171 discussions

Installing Kamailio on EC2
by Sunil Chandrasekharan 26 Jun '13

26 Jun '13

Hi All, I have just started and installed Kamailio on EC2 Ubuntu instance. I am going on with the steps provided in this link http://www.kamailio.org/wiki/install/4.0.x/git i have gone ahead and installed Kamailiio. I have just reached to Section no. 8 on the link. But i am not sure if there is anything extra i need to take care. Kindly let me know if any extra parameters / details need to be checked for Kamailio on EC2. On Wed, Jun 26, 2013 at 6:00 PM, aft <aftnix(a)gmail.com> … [View More]

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Installing Kamailio on EC2
by Sunil Chandrasekharan 26 Jun '13

26 Jun '13

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 … [View More]

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481 dialog/transaction does not exists Error.
by Nandini madhu 26 Jun '13

26 Jun '13

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,… [View More]

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'481 dialog/transaction does not exists'
by Ravindra Gowda 26 Jun '13

26 Jun '13

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, … [View More]

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Re: [SR-Users] Does Kamailio provide any API for other program tocreate sip account
by cxw1987 26 Jun '13

26 Jun '13

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Re: [SR-Users] Does Kamailio provide any API for other program to create sip account
by Xiangwen Chen 26 Jun '13

26 Jun '13

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!

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Does Kamailio provide any API for other program to create sip account
by Xiangwen Chen 26 Jun '13

26 Jun '13

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!

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dp_translate() pseudovariables
by Victor V. Kustov 26 Jun '13

26 Jun '13

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

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Re: [SR-Users] Does Kamailio provide any API for other program to create sip account
by cxw1987 26 Jun '13

26 Jun '13

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SIP Trunks Location
by Jose Suero 25 Jun '13

25 Jun '13

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

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sr-users June 2013