Re: [SR-Users] One way communications in Kamailio 3.1.4

2011-09-16 Thread Daniel-Constantin Mierla

Hello,

On 9/14/11 6:13 PM, CaptWho wrote:

Thanks Timo, I'll look into that, just have to figure out what you just said.
:confused: I'm at the bottom of the learning curve and I got dumped into
this after the guy that was dealing with it disappeared.
it will help also to watch the sip signaling traffic, using ngrep or 
wireshark.


For example with ngrep, on the same server with kamailio:

ngrep -d any -qt -W byline port 5060

will capture all the incoming and outgoing sip messages. You can see if 
there are some retransmissions, where the messages are sent, thus being 
able to detect if there is anything wrong in routing. If you do nat 
traversal, then also sdp and some headers must be updated in the proxy 
server.


Cheers,
Daniel




Timo Klecker wrote:

Hello,

the disconnect after 32 seconds is sure to be a lost ACK. Maybe the ACK is
not send E2E but using the Kamailio. In this case you would have to
redirect
it.

In the content of INVITE and 200, 183 or ACK you should be able to see the
IPs for RTP data. Compare these with your phones.
O=root 12312331 12312331 IN IP4 XXX.XXX.XXX.XXX

Be sure, the RTP ports are open on your firewall, too. You can see the
ports
in the content, too.
M=audio X RTP/AVP 8 101

Greetings
Timo



--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda


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[SR-Users] One way communications in Kamailio 3.1.4

2011-09-14 Thread CaptWho

I'm running 3.1.4 on centos and I'm having some trouble with voice only going
one way.  

Both extensions will ring each other, but after they connect, voice will
only travel in one direction.  One extension hears fine, but can't talk.  

I've totally opened up the firewalls (including port 5060) and I'm still
having the trouble.  I'm not behind NAT. 

I've tried it using X-Lite, VoIP phones and ATAs in a number of
combinations. The problem is across the board.  X-Lite sometimes
automatically disconnects after 32 seconds.

Does anyone have any suggestions on where to look? ANY help appreciated. 
This has been going on for weeks.

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Re: [SR-Users] One way communications in Kamailio 3.1.4

2011-09-14 Thread Timo Klecker
Hello,

the disconnect after 32 seconds is sure to be a lost ACK. Maybe the ACK is
not send E2E but using the Kamailio. In this case you would have to redirect
it.

In the content of INVITE and 200, 183 or ACK you should be able to see the
IPs for RTP data. Compare these with your phones.
O=root 12312331 12312331 IN IP4 XXX.XXX.XXX.XXX

Be sure, the RTP ports are open on your firewall, too. You can see the ports
in the content, too.
M=audio X RTP/AVP 8 101

Greetings
Timo

-Ursprüngliche Nachricht-
Von: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von CaptWho
Gesendet: Mittwoch, 14. September 2011 05:01
An: us...@lists.kamailio.org
Betreff: [SR-Users] One way communications in Kamailio 3.1.4


I'm running 3.1.4 on centos and I'm having some trouble with voice only
going one way.  

Both extensions will ring each other, but after they connect, voice will
only travel in one direction.  One extension hears fine, but can't talk.  

I've totally opened up the firewalls (including port 5060) and I'm still
having the trouble.  I'm not behind NAT. 

I've tried it using X-Lite, VoIP phones and ATAs in a number of
combinations. The problem is across the board.  X-Lite sometimes
automatically disconnects after 32 seconds.

Does anyone have any suggestions on where to look? ANY help appreciated. 
This has been going on for weeks.

--
View this message in context:
http://old.nabble.com/One-way-communications-in-Kamailio-3.1.4-tp32460597p32
460597.html
Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.


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