the internet calling settings for Use external SBC for Internet Calling is unchecked - see full detail attached.
I'm sorry but i do not understand you last question "Since both phones are remote and on the same local network also, what is the setting in sipx for "
On 03/02/2011 12:37, Tony Graziano wrote:
xlite topology should be: use local address advanced session timers=disabled sip keepalives=enabled rport=enabled account register with domain and receive calls=checked domain=checked What is your setting at System>Internet calling: Use external SBC for Internet Calling set for?Since both phones are remote and on the same local network also, what is the setting in sipx forOn Thu, Feb 3, 2011 at 5:20 AM, xavier houghton <[email protected] <mailto:[email protected]>> wrote:HI List, i have a problem with a UA when recieving inbound calls. The problem seems to be a missed routed ACK coming from SIPproxy. I'm posting it here as it seems to be similar to a couple of bugs in the tracker (XTRN-970 <http://track.sipfoundry.org/browse/XTRN-970> . I had earlier in the week posted a similar issues in sipxuser list I'm calling an internal extention from another internal extention. the scenarion i have is as follows: phone1 - X-lite on LAN10.7.6.214 <http://10.7.6.214>: ext 205 Phone 2 - Intertex ix67 ATA on LAN (using phone port)10.7.6.191 <http://10.7.6.191>: ext 206 My LAN is Behind a NAT. I have an intertex firewall with SIP module off and no rewriting going on to replicate a standard remote setup with ALG off SIPx hosted on VM on public IP address outside of this network. its running 4.2.1-01897 build 34 from ISO. the server is behind a 1:1 NAT internet calling is setup for remote users and sipx knows its behind a nat. the registrations look like this is sipxconfig sip:[email protected] <sip:[email protected]:63854;rinstance=231fc0b019b13367;x-sipX-privcontact=10.7.6.214%3A5546> 179 sip:[email protected] <sip:[email protected]:63996;x-sipX-privcontact=10.7.6.191> The symptoms are that after 60 seconds an inbound call to the UA stops. here's what happens to the call When i call phone 2 from phone 1 the call is setup and there is audio both ways. after 62 seconds the call stops. I've looked at the traces in sipviewr and see that on the call setup SIPX get a 200 OK from phone 2 which is forwarded to Phone 1. phone 1 replies with an ACk to sipproxy. SIPx forwards the ACK to the Phone 2 LAN address, destination 10.7.6.191. as the ACK never gets to the phone2, phone 2 resends the 200Ok a number of times with the same scenario as above happening. after a number of retires phone 2 sends a Bye and the call ends. i have no issues the other way round. that is to say when calling Phone 1 from phone 2 i have no issues had the call up for around an hour. I've included a snapshot attached. I've also included a trace from SIPX. Sub2.xml is the failled call. Sub1.xml the working call for reference if needed. help? _______________________________________________ sipx-dev mailing list [email protected] <mailto:[email protected]> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430sip: [email protected] <mailto:[email protected]>Fax: 434.326.5325 Email: [email protected] <mailto:[email protected]> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426sip: [email protected] <mailto:[email protected]>Helpdesk Contract Customers: http://support.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev/ No virus found in this message. Checked by AVG - www.avg.com <http://www.avg.com> Version: 10.0.1204 / Virus Database: 1435/3419 - Release Date: 02/02/11
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