Re: [SR-Users] Kamailio Dispatcher and FreeSwitch, Too many hops.

2011-10-09 Thread Morten Isaksen
>From Kamailio. 2011/10/8 Henrik Aagaard Sørensen : > From Kamailio or FreeSwitch? > > On Sat, Oct 8, 2011 at 10:28 PM, Morten Isaksen wrote: >> >> Can you capture one of the calls that fails with tcpdump. >> >> Also try to add some xlog lines in the configuration file for debuging. >> >> What do

Re: [SR-Users] Audio quality issue

2011-10-09 Thread Austin Einter
Hi All Thanks for your kind answer. The call flow looks as below I have two doubts here 1. My UA is just behind the Modem, and in Kamailio config file I have enabled WITH_NAT, will this lead to any kind of problem 2. In kamailio proxy I am using force_rtp_proxy and unforce_rtp_proxy instead of

Re: [SR-Users] Kamailio Dispatcher and FreeSwitch, Too many hops.

2011-10-09 Thread Henrik Aagaard Sørensen
When trying to dial 101 this is a tshark output on the Kamailio: 0.00 71.12.95.46 -> 215.183.255.142 SIP Status: 480 Temporarily Unavailable 0.000196 215.183.255.142 -> 71.12.95.46 SIP Request: ACK sip:1...@sip.my-domain.com 0.000255 215.183.255.142 -> 95.214.24.165 SIP Status: 480 Tem

[SR-Users] a=nortpproxy:yes

2011-10-09 Thread Austin Einter
If the SIP UA is not behind the NAT, and Kamailio insersts a=nortpproxy:yes in SDP body, can it cause one sided audio. I am using PJSIP as my UA, kamailio 3.1.5 , rtp proxy 1.2.1 as intermediate proxy for signalling and media. In kamailio config, I have enabled WITH_NAT. On call setup, only one wa