Hello,
run asterisk in debug mode to understand why is sending BYE.
Cheers,
Daniel
On 07/08/15 16:40, Loic Chabert wrote:
Hello,
I have set on the right place "route(RTPPROXY")", and now it works for
internal calls and external calls.
Reason: my request passing througt RTPPROXY twice ...
One last problem:
- 102 initiate a call to 101
- 101 refuse call with a 486 response
- as asterisk dialplan said: launch voicemail app
- Sounds files has been read from asterisk, but after 5 secondes,
session has been cut with a BYE request sent by Asterisk.
Please find in attachement pcap trace file (91.x.x.x is wan kamailio
interface, 10.0.247.197 is lan kamailio interface, facing to asterisk
cluster)
Why asterisk send this BYE ? Kamailio does not force him to send this
BYE...
Thanks,
Loic.
2015-08-07 10:34 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com
<mailto:miconda@gmail.com>>:
Hello,
look at the sip traffic and see what is in SDP, if you don't get
audio, maybe the other ip is advertised.
Cheers,
Daniel
On 07/08/15 09:16, Loic Chabert wrote:
Hello Daniel,
I have changed my rtpproxy by rtpengine. I have explicitly define
public and private interfaces, and now it work as expected for
external calls (througth PSTN).
But for now, after this change, internal call (like 100 call
101), does not work any more.
I need more investigation to see what append on my call flow.
I will update you asap.
Thanks,
Regards.
2015-08-07 9:04 GMT+02:00 Daniel-Constantin Mierla
<miconda(a)gmail.com <mailto:miconda@gmail.com>>:
Hello,
On 30/07/15 17:38, Loic Chabert wrote:
Hello everyone,
I'm trying put kamailio in front of asterisk server farm.
Fow now, 2 asterisk servers are running and i'm trying to
make some basic calls between two UACc.
All asterisk servers has been ofuscaded from public internet
using 10.189.122.0/24 <http://10.189.122.0/24> network.
All trafic must be passed throught asterisk so RTPproxy is
used to (and used for rtp bridging).
Kamailio and rtpproxy is running with public IP address, and
private ip address (mhomed=1)
But a wired thing append on my SDP body: c line have two
rtpproxy public addresses concatenate (see my capture attached).
Any reason for this ? Only invite method from my asterisk
contains 2 publics IP addresses concatenated.
Does it mean than rtp_manage as been executed twice ?
It could be that it was executed twice. As pointed in another
response, look at what is received on the network and in the
logs.
You can enable cfgtrace for debugger module in order to see
what actions are executed from configuration files -- it is
good to spot quickly errors in the logic of config file.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio -
http://www.asipto.com
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--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio -
http://www.asipto.com