Hello,
I have a situation, where provider does not understand reivnites when user
makes a Call Forwarding via Asterisk server, it only understands 302 Moved
messages in CF situations (or in Call tranfer).
So my scenario is like this:
Provider sends Invite to Kamailio, I forward it to Asterisk server. where a
user make a call forwarding to Kamailio (INVITE message), who then forwards
it to provider - my question if, can I change the INVITE that Asterisk send
Kamailio server and change it to 302 Moved message and relay it to provider
(I can detect when there is an CF, because Asterisk sends diversion header
in the INVITE towards Kamailio)?
Hi,
I have 2 kamailio servers behind a load balancer.
How can I use t_suspend on one server and t_continue on the other one. I
already have the index and label of the transaction saved in a common
database shared by the two kamailio servers.
The reason I want to do this is for push notifications for iOS and Android,
and I can't guarantee that the caller and the callee will register on the
same server because of the load balancer.
Thanks
Ahmed
Hi Dears,
I'm trying to configure the SIP Capture module with Kamailio BUT i'm
receiving that Error.
Dec 13 09:20:53 debian /usr/local/sbin/kamailio[5183]: ERROR: sipcapture
[hep.c:84]: hep_msg_received(): ERROR: sipcapture:hep_msg_received: not
supported version or bad length: v:[82] l:[69]
Any ideas how to solve this please ?!
Best Regards and Thanks in advance
Hello,
Kamailio is difficult. Many of us want to use it because is open source and it's flexible. but to tell you the truth after 3-4 hours of playing with it I am frustrated, and I am starting to hate it!
Why has nobody made a kamailio video series on youtube?-example of how to install kamailio-example of how to admistrer kamailio, and what other programs you need to work with;-example how to create users in Kamailio for enterprise use.-example of running kamailio outside enterprise like a SBC with a public ip;-example of how can you run kamailio as a load balance server.-example how to save money on your sip infrastructure using kamailio instead of cisco cube for example or sonus..or any other example you might thing is useful in a sip environment!
I look at youtube and I see Kamailio this and kamailio that, but no actual tutorial! This is 2014, people don’t waste time reading documentation, youtube is much more efficient...I think Kamailio will not have the success of it's potential until senior community members will not take the plunge and broadcast their knowledge on a proper video channel! Asterisk has every scenario up there based on these info I was able to setup in less that 3 days my office and 40 people are making calls......It's unbelievable that after so many years nobody has made a godam explicit tutorial for newbies in kamailio...I am so frustrated right now!!
Hello,
I want to achieve the following schema:
1. Accept an initial invite
2. Create separate dialog to different SIP server (probably with some
headers from point 1)
3. Wait until that different SIP server will complete dialog.
4. Based on result of point 3, transfer invite from 1 to the next steps or
decline a call.
The problem I've experienced - how to create a separate dialog and control
it inside main dialog?
Thanks a lot!
Hello,
I setup the mirror of Kamailio github repository to kamailio.org, can be
seen online at:
- http://git.kamailio.org/gitlist/index.php/kamailio/
Mirroring is triggered by a push to github, so it is pretty much in real
time.
Right now the commit and mirroring flow is:
- commit to git.sip-router.org triggers mirroring (push) to github,
which triggers a mirroring (fetch) on git.kamailio.org
Practically with a commit pushed directly to github, the first part of
the above is skipped.
All seems to be ok, my last commits were done to test the mirroring
including the notification emails - I will send a separate message about
what we can get there.
The missing parts are the deployment of gitweb in order to have the
links from archived email notifications still valid and pointing
git.sip-router.org to kamailio mirror server.
Considering all above, I propose to stop committing to
git.sip-router.org and pushing to github starting either next Monday
(Dec 15) (or Wednesday (Dec 17) if not ready by Monday). Existing
developers have to send me the github username to add it to kamailio
project.
Any suggestions and comments are welcome!
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hello,
Something I've been wondering about meddling with sip uris with Kamailio:
I know it's possible to translate between a number prefix and a domain
using PDT, but is this possible: Having a numeric or alphanumeric value
stored in db, associated to a domain and appended to / removed from a
number during processing a sip message?
For example 112222333(a)mydomain.com --> 333.5555(a)mydomain.com where value
5555 and mydomain.com are associated to one another in db somewhere. In
this example I could have a pdt with prefix 112222 and domain mydomain.com,
but the value 5555 should be stored somewhere as well.
So one part is to find a db table suitable for storing a value and
associating it with a domain, and the other part is to change the sip uris.
I'll appreciate any comments!
cheers,
Olli
Hello. I installed kamailio. All OK.
But when i call to other servers. ostel.co. for example. no ring.
What element permits federation between kamailios? DNS Records?
Thank you.
Hello,
Is is possible to change SIP message type?
Example: Server A sends Kamailio SIP INVITE message, can I change this SIP
INVITE to "302 - MOVED" (and reuse data from SIP INVITE) and send it to
Server B?
Does anybody have any example?
Best Regards.
