.... is it codec issue?
On 26/10/2011 04:07, Adrien Guillon wrote: > Hi everyone, > > I have been working on incoming calls from a sip trunk, and debugging > potential issues. Right now, calls are disconnected immediately after > I dial an extension from the AA (when I call externally). I'm pretty > sure the NAT is configured properly, and I'm starting to narrow down > the problem. The NAT uses nf_conntrack_sip rather than explicitly > opening RTP ports. I used tcpdump to monitor incoming calls, and I > find events such as (right before disconnection): > > 19:40:25.689135 IP bm-srv-01.voicenetwork.ca> 123.456.1.12: ICMP > bm-srv-01.voicenetwork.ca udp port 19222 unreachable, length 208 > > I have discussed this with a friend, and one potential issue could be > how the phone network is configured. My phones are firewalled so that > they can only communicate with the SipX server. I am not sure if the > transfer negotiation is attempting to pass the connection directly to > the phone, which then has no path back (and is not really reachable > from the NAT system). > > Any suggestions? > > AJ > _______________________________________________ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
