Oh thank goodness, cause I didn't either. Sorry to waste your time on that. I'll try again and look for actual call information in the next log.
Danny From: Tony Graziano [mailto:[email protected]] Sent: Tuesday, August 24, 2010 3:12 PM To: Danny Shay Cc: [email protected] Subject: Re: [sipx-users] Cannot transfer calls coming from site-to-site dial plan I dont see a call in this trace file. On Tue, Aug 24, 2010 at 4:07 PM, Danny Shay <[email protected]<mailto:[email protected]>> wrote: I followed the suggestion Tony gave me: Set the proxy to debug, rotate the logs usr/sbin/logrotate -f /etc/logrotate.conf Make the call. When it fails run these commands... cd /var/log/sipxpbx merge-logs sipviewer merged.xml I copied the merged.xml and have attached it. The call that was answered then transferred was at 2:48 and was 1:12 minutes long, it shows "transferred" in the Call Detail Record but it wasn't. Danny -----Original Message----- From: Tony Graziano [mailto:[email protected]<mailto:[email protected]>] Sent: Tuesday, August 24, 2010 2:06 PM To: Danny Shay; [email protected]<mailto:[email protected]> Subject: Re: [sipx-users] Cannot transfer calls coming from site-to-site dial plan I have it set up and working via the internet with no issues. I use polycom handsets though. Call comes in from PRI gateway, answered, transfers to remote location. I used to do this for a coupke of sites via vpn the same way, but the need for vpn has gone away. I immediately think handsets but I don't use nortel handsets and haven't seen a call trace. I am successful doing this "between" versions also (4.x). I would be careful with the "any number of digits", dialing to certain area codes might be affected! Put the proxy at debug logging and get a call trace. Do you have phones other than nortel to test with? ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected]<mailto:[email protected]> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: Danny Shay <[email protected]<mailto:[email protected]>> To: Tony Graziano <[email protected]<mailto:[email protected]>>; [email protected]<mailto:[email protected]> <[email protected]<mailto:[email protected]>> Sent: Tue Aug 24 14:36:23 2010 Subject: RE: [sipx-users] Cannot transfer calls coming from site-to-site dial plan I was recreating dial plans from the older version of Sipx when I made these dial plans. It was working properly before both units were wiped out and set up with latest iso and updated to the previously mentioned build. Ok here's how everything's set up: Phonesystem.williamspryor.local is 192.168.0.20 Phonesystem.williamstulsa.local is 192.168.1.100 Configuration of "sipx_system2" unmanaged gateway: Name: sipx_system2 Address: williamstulsa.local <-- Changed this per Tony's instruction. Used to be the FQDN of the phone system.) Port is default Transport protocol is Auto Location - all - Shared is checked. Dial Plan site-to-site configuration: Name: Site to Site Dialed Number: Prefix 4 and any number of digits Resulting Call: Dial (blank) and append "matched suffix" This is the same for both units, except the proper "address" is referenced in the rule. Remember, the actual site-to-site is working. It is only when the transfer is attempted that it fails. Also, I tested using a blind transfer and that is not working either. I just tested and this fails in either direction. Direct calls work, but when transfer is attempted it fails. Danny From: Tony Graziano [mailto:[email protected]<mailto:[email protected]>] Sent: Tuesday, August 24, 2010 12:38 PM To: Danny Shay; [email protected]<mailto:[email protected]> Subject: Re: [sipx-users] Cannot transfer calls coming from site-to-site dial plan IT DOES NOT WANT FQDN, it is nice it can resolve, but a site-to-site dial plan using sip wants SIPDOMAIN as the gateway. so, not "sipx.sipdomain.tld" as the gateway name, its "sipdomain.tld". See the wiki. http://wiki.sipfoundry.org/display/xecsuserV4r2/sipXecs+to+sipXecs+Calling On Tue, Aug 24, 2010 at 12:56 PM, Tony Graziano <[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> wrote: Explain your site to site dial plan. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> <[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> To: [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> <[email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>>> Sent: Tue Aug 24 12:34:19 2010 Subject: [sipx-users] Cannot transfer calls coming from site-to-site dialplanPoint-to-point T1 between two sipx pbx's both running 4.2.1-018932 2010-06-25T10:43:40 build33. Can call from one pbx to the other with a site-to-site dial plan and successfully communicate. When the receptionist attempts to transfer the call from her phone to another in the office, the intended phone rings once then stops, and the call comes back to her when she hits end call on the phone. The phones are all LG-Nortel LIP-68xx units with 1.2.49 firmware. Network one is 192.168.0.0/24<http://192.168.0.0/24><http://192.168.0.0/24>, network 2 is 192.168.1.0/24<http://192.168.1.0/24><http://192.168.1.0/24>, nat is not in play. Ping by FQDN from one pbx to the other on each box is successful. Calls coming from audiocodes MP-118FXO transfer correctly. Let me know anything else you need! Thanks, Danny Shay Norlem Technology Consulting -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> Fax: 434.984.8431 Email: [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected]<mailto:[email protected]><mailto:[email protected]<mailto:[email protected]>> Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected]<mailto:[email protected]> Fax: 434.984.8431 Email: [email protected]<mailto:[email protected]> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected]<mailto:[email protected]> Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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