Hello all,
I have an environment setup as follows:
VoIP routers -> Kamailio 4.0 -> Asterisk 11.6
When I register the phones to Asterisk the call quality is excellent. So to be clear, the call comes in the VoIP router, routes through Kamailio to the phone registered to Asterisk.
When I register the phone to Kamailio the call quality is significantly reduced. So clarity here, the call comes in the VoIP router, routes through Kamailio to the phone registered to Kamailio.
Why would the call quality diminish when registered to Kamailio? Where can I start looking for this issue?
Thank you,
Travis R. Dillon
Hello all!
I make kamailio1 remotely register to kamailio2.
Here I set K1 modparam("uac", "reg_timer_interval", 15), and expires in uacreg as 30s.
Then K1 should refresh registration after 15s.
However the callid and from tag of every registration are different, and the Cseq just 10 or 11,never increased.In RFC3261, when UAC want refresh bindings, A UA SHOULD use the same Call-ID for all registrations during a single boot cycle.
Is there any way to reuse the callid and fromtag during refresh registration ? Many thanks!
And how to make the Cseq become 12 or more?
Hi,
I'm currently wracking my brains trying to deduce where an application hang can be coming from.
I have ported the PostGres schema to be compatible with MS Sql Server and have had no issues getting a working dev environment working. However when I try and use SQL Azure as the backend db as opposed to SQL Server 2012 (running locally on my dev machine) Kamailio will seem to hang after about 10 - 15 mins of flawless operation. I have recreated the SQL Azure schema from the working schema running on my dev machine and copied all data in a 1:1 manner so I can be certain it's not a data quality issue. I am using the same FreeTDS/UnixODBC config that I am using for Asterisk realtime (which doesn't exhibit this behaviour). The really strange thing is that I can see the queries being sent by Kmailio in the freetds trace log, and the response from SQL Azure, however its as if the db response handler is deadlocked/hung. The kamailio process continues to work fine, and as long as its not required to commiunicate with the DB all appears well i.e. no error messages in the log file.
This behaviour only happens when im using SQL Azure as opposed to SQL Server, so there must be something causing the issue, that's unique to SQL azure as opposed to SQL Server. As far as I can tell this is not something I can create a backtrace for as no core dumps are produced, as far as I can tell there is no easy way debug what the lock is. If anyone has any suggestions I would be very grateful to hear them.
One thing that occurs to me is that the db_unixodbc module doesn't have half as many parameters as its equilent in Asterisk, and that connection pooling may be in use, this could be a bit of a show stopper AFAIK you have to disable connection pooling with SQL server, certainly any of the guides I have come across configuring Asterisk to work with SQL server suggests that.
Tim Chubb
Developer
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Hi Kamailio,
- I meet an error when I am trying install Kamailio 4.1.4 on Mac OS 10.9.
The error:
touch /usr/local/lib64/kamailio/modules/usrloc.so
install -m 755 usrloc.so /usr/local/lib64/kamailio/modules
touch /usr/local/lib64/kamailio/modules/xhttp.so
install -m 755 xhttp.so /usr/local/lib64/kamailio/modules
touch /usr/local/lib64/kamailio/modules/xhttp_rpc.so
install -m 755 xhttp_rpc.so /usr/local/lib64/kamailio/modules
touch /usr/local/lib64/kamailio/modules/xlog.so
install -m 755 xlog.so /usr/local/lib64/kamailio/modules
touch /usr/local/lib64/kamailio/modules/xprint.so
install -m 755 xprint.so /usr/local/lib64/kamailio/modules
/bin/sh: -c: line 1: syntax error: unexpected end of file
make: *** [install-cfg] Error 2
How can I fix it? I am new in Kamailio, I want to learn Kamailio.
--
Thanks and best regards,
Phạm Sơn Trường
Android, iOS Team Lead at ME Corp.
Email1: truonguit2010(a)gmail.com
Email2: truongps(a)mecorp.vn
Mobile: 01218816208
When an invite packet is recieved by kamailio server, what processing is done
on it ? Which modules and functions are called in its processing and in
which sequence ?
Thanking in advance.
--
View this message in context: http://sip-router.1086192.n5.nabble.com/Processing-of-Invite-Packet-tp12867…
Sent from the Users mailing list archive at Nabble.com.
Hi,
I am trying to configure a SIP proxy architecture that is stateful with
regard to transaction but not the dialogues, i.e. If an INVITE takes a
specific route, the corresponding 180 Ringing, 200 OK and ACK shall
traverse the same set of proxies, but the corresponding BYE can take
another route (and obv it's own 200 OK takes the same route again)
Which modules can help me with that?
In specific which functions / modules are recommended to handle requests
and responses?
Currently, I use "t_relay()" from TM module to forward requests and
"loose_route()" from RR module to handle replies, but it looks like that if
a BYE ends in a proxy which its corresponding INVITE has not traversed, it
is not forwarded by that proxy.
Am I making a mistake in my routing? Isn't there a simple way to do
transaction stateful routing in Kamailio?
Thanks,
Alireza
Hi,
when t_relay() times out, calling sl_reply_error() will send 408 to the client.
But if I don't want to send 408, how do I interpret the error code t_relay() set?
Regards,
Allen
Hi,
If I normally receive a INVITE that has a header 'aaa' with value 'bbb', but it's possible that some INVITEs don't have this header.
How do I determine in the config file if the header exists?
To check the value of the header I do if($hdr(aaa) == "bbb"). But what if the header doesn't exist?
Cheer,
Allen
Hello
I have successfully (I believe) implemented Kamailio 4.1.4 integration with
Freepbx 5.2.11 taking as a guide Daniel's tutorial
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.
I just did not create the voicemail tables because voice mail is handled by
Freepbx. I installed the system in a separate box for testing and connected
to the Freepbx Production server via IAX trunk.
The system is behind a Cisco Firewall and looks like this
Remote User Internet
Internal network
Cisco 7960 ----DSL router ---|------Internet --------|-----Cisco ASA 5500
FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx
Production Server --------|------ PSTN
I have configured the FW to allow UDP and TCP traffic from the corresponding
IP as well as tfpt that is needed for the Ciscos to pick up the
configuration from the server. I have a few remotes Cisco 7960 phones that
can register remotely in Kamailio as long as the user is added with kamctl
add user password and as long as the extension is created in Freepbx.
The problem that I have is when try to make a call from the remote Ciscos
the call is dropped after 30 or 40 seconds. I can see from the logs that the
problem appears to be that the server is not receiving responses from the
phone
06-25 10:57:30] WARNING[1814] chan_sip.c: Retransmission timeout reached on
transmission 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 for seqno 102
(Critical Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
[2014-06-25 10:57:30] WARNING[1814] chan_sip.c: Hanging up call
000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 - no reply to our critical
packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Is this something that we can adjust in kamailio or could it be related to
the FW configuration?? Sorry but I am very new to kamailio and sip.
Thanks
Carlos
Hello,
For a particular gateway, I force the RTPProxy. This work fine but on the
other hand, in my side the RTPPRoxy response to the source IP address of SIP
packet instead of the IP address in "Connection information".
Is there a way to instruct the use of the IP address announce in the SDP
instead of the last IP source of the last SIP packet?
Regards,
Igor.
