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.
Questions tagged [kamailio]
209 questions
2
votes
2 answers
VoIP calls doesn't work in different networks (Using PJSIP and Kamailio server)
I have setup kamailio 4.2 on an azure instance as server and for client I am using PJSIP library for Android and iOS applications. The voice calls seem to work well when both the devices are connected to the same network, however, either of the…

Dnyaneshwar Wakchaure
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2
votes
1 answer
How exactly to create simple failover on Kamailio with load balance?
I have 3 locations listed in my dispatcher.list file. 2 are meant to load balance eachother. At the moment it works perfectly as the logs show the calls doing a round robin between the 2 locations.
However, I have been going through online docs and…

Damainman
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2
votes
2 answers
Send rtp packet between two rtpproxy server
Is it possible to send media rtp packets from one rtpproxy server to another rtpproxy server?
In my scenario , i am registering voip account via opensips proxy server. We have rtpproxy and opensips server hosted on same place. opensips changes c=…

kaushik parmar
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2
votes
1 answer
Kamailio-Asterisk - route "FROMASTERISK" not found
I'm trying to implement Kamailio 4.1 with Asterisk 12.1.0 regarging this tutorial:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
And when I try to compile kamaili.cfg, I still got this error:
Apr 19 16:59:31 debian…

Patrik18
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2
votes
3 answers
How to provision a test user in kamailio?
I have just (for the first time) compiled and installed kamailio, following this guide. For configuration, I am following the documentation here
I am trying to test a new SIP user. I have created it with:
» kamctl add test testpasswd
The user is…

blueFast
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2
votes
1 answer
Custom SIP Header In Kamailio
I need to set a custom header in Kamailio 3.3.1 from a Lua script.
I can theoretically set one in the config script like this -
append_hf("X-MyHeader: myvalue\r\n");
but I cannot work out how to call it from a Lua script, which is my preferred…

David Wylie
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1
vote
1 answer
How can I send message from Kamailio SIP server (kamailio.cfg) to the SDN controller using HTTP POST?
I would like to know how can I send certain message from Kamailio SIP server to the SDN controller. It doesn't matter if it is http/xhttp/http_client etc. module.
My idea is:
SIP message comes to the Kamailio.
Kamailio processes message in…

gruneluk
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1
vote
1 answer
Kamailio 5: KEMI & Python - import KSR
I am currently using an older version of Kamailio in production. I want to be able to write some custom logic, and v5 allows me to do that in Python using the KEMI interpreter - so I'm interested in upgrading. However I'm encountering something…

Joel
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vote
1 answer
How to get Kamailio to set `Record-Route` header to internal IP for internal leg of call?
I have Kamailio 5.4.1 (and RTPEngine) running on an internal server with a private IP address 172.31.7.96 and One-to-one NAT to an external IP address. The external IP is 192.0.2.100. (Note: The internal IP addresses are all unedited, but the public…

Moshe Katz
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1
vote
4 answers
Are Kamailio pseudo-variables $var safe to use until the end of a single message handler?
I'm using pseudo-variables $var in Kamailio because according to the documentation, they are faster than $dlg_var, but I'm wondering if it's safe to use them like this:
jansson_get("a", $http_rb, "$var(a)");
$var(i) =…

Loky
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1
vote
1 answer
SipML5 with Kamailio as Sip Server return 488 in make Call
I have setup Kamailio with websocket module. When I register with sipML5 its going well. But returns 488 Not Acceptable Here when I trying to call.
488 means: Some aspect of the session description or the Request-URI is not acceptable, or Codec…

wanz
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1
vote
1 answer
OpenSIPs Control Panel errors
I have setup OpenSIPs control panel and I can successfully complete basic functions like adding users. Problem is I keep getting this error when clicking on most features in the control panel.
MI command failed with code 406
Not sure how to fix, any…

Fonewiz
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1
vote
1 answer
Kamailio - Dispatcher determine availability via http?
We are currently using the dispatcher module in kamailio to get the availability of gateways via the dispatch list.
It uses a health check based on if it can talk to the gateway via SIP by default. However, I would like to know if we can make the…

aturner
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1
vote
1 answer
PSTN to OpenSIPS to next SIP destination
I have worked with Asterisk for years but I am very new to OpenSIPS. What I need is to have calls come in from our DID provider to the OpenSIPS server then redirect them to another SIP URI.
Something like this:
DID Origination Provider -> OpenSIPS…

Fonewiz
- 2,065
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- 17
1
vote
1 answer
How do you do NAT traversal for RTP media using Kamailio for signalling?
There are three devices in question.
A VoIP phone behind NAT
My own Kamailio Server on an EC2 instance.
The Linphone application for android on my phone.
My phone is on mobile data, and since I have an MVNO, it appears to be NATed as well (private…

Anthony A
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