No. If the phone is connect to sipx, and sipx is doing the trunking, sipx is anchoring the media.
Ensure the ITSP is ending the INVITE on the incoming call to port 5080. On Tue, Oct 25, 2011 at 4:07 PM, Adrien Guillon <aj.guil...@gmail.com>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/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Contract Customers: http://support.myitdepartment.net <http://support.myitdepartment.net>Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services!
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