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Questions tagged [kamailio]

Kamailio (former OpenSER) is a SIP proxy server. DO NOT ask questions relating to installation, use, or configuration of this software. Stack Overflow is a site for programming Q&A so on-topic questions will involve working directly with Kamailio's source code, interacting with its C APIs, or using a programming language such as Python or JavaScript to write routing logic using KEMI.

209 questions
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build SIP server similar iptel.org by kamailio

I want build SIP server like iptel.org. I use this tutorial (http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour) to install Kamailio SIP Server. But I have some problems. Server does not work with UDP.(while I configured…
MJH
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How to get call info from Kamailio

I have setup a Kamailio server and am able to establish calls. I need a way to get call related information like from, to, duration,etc. I have enabled the dialog module in the config but no avail. I am not well versed with config files and I am not…
suchitra nair
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Using Kamailio for Push To Talk Android App

I installed Kamailio, am able to add users and make calls, but I need to use it for my android push to talk client app i.e. upon calling another user(s) (using the same app), instead of ringing their app/phone it should directly transmit voice of…
codebee
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Why the kamailio remove two route header?

There is such a scenario: u1(10.35.146.155) ---- kam proxy1(10.35.148.211) ---- kama proxy2 ---- u2(10.35.148.233, u2 is registered on kam proxy2) The problem occurs in the forwarded ACK. After the ACK is forwarded, there is only one route header…
wangpeng
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Is there any way to answer a call or originate a call by Kamailio?

I know that Kamailio works as a SIP Proxy and I also know that Asterisk/FreeSWITCH or other similar products can do what I'm asking here, but still wondering if it's possible to use Kamailio to answer a call or originate a call? Let's say, "User…
StanZ
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SIP trunking and call routing in Kamailio

I was using freepbx, but because of some limits I installed kamailio on another machine. I want to have a route for outgoing calls to NGN(was peer friend siptrunk in freepbx), which handles call setups started from extensions registered on…
MKeshavarzian
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1 way audio only when registering to OpenSIPs in front of Asterisk

Long time Asterisk user but fairly new to OpenSIPs. I have a SIP phone working with audio both directions when registering to and receiving calls directly from Asterisk. The same phone works with 2 way audio if I register to OpenSIPs and receive a…
Fonewiz
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Freeswitch Guide how to get CDR UUID from client side

My system contain: - Freeswitch server - Sip Client: Web using sipjs , mobile react-native using https://github.com/datso/react-native-pjsip to receive call. My problem is when call done i need to know the uuid of CDR recently add to Postgres DB of…
QViet
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Kamailio kamctlrc password encode

I'm trying to encode the passwords that appear in kamctlrc file. I don't know if it is possible. Is there a way to do it? Best wishes, D.
Daniel Ramos
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How to start Kamailio on boot

I am trying to start the Kamailio Service on Ubuntu when the system boots. I have tried using the commands "service kamailio enable" and "systemctl enable kamailio.service," but the service still does not start on boot. Any help would be…
UsernameGoesHere1
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kamailio add trunk with credentials

recently started with kamailio and looks really good. i have a few extensions and they are talking to each other through rtp proxy. Easy stuff. the tricky bit is to add a provider to talk to the outside world. all the documentation that i have seen…
john
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Bad Media Description when I try make call between two JSSIP clients

I'm trying make a call between two JSSIP clients. Both of them in the same machine on Google Chrome browser (I saw some differences on the Mozilla console). Immediatelly after confirm the call it's closed. In the log message I saw "Bad Media…
Washington Costa
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How to create multiple sip trunking based on pstn grouping using kamailio

I have kamailio as load balancer and two asterisk machine as pstn gateway. In dispatcher.list I have entry for both asterisk machine. Sip trunking is established successfully between kamailio and asterisk.all working fine, Is there any way to create…
dkakoti
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how to make kamailio serial forking?

@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ I am a beginner in kamailio server development, and I want to make serial forking, but that doesn't work. My kamailio server replies Too Many Hops (code: 483)…
GodSon Daher
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