On Thu, 2009-11-12 at 11:15 -0600, Gabe Casey wrote: > I am having some issues using Asterisk as a PRI gateway with Sipxecs. > For the most part it works for inbound and outbound calling however > when a call is received on a PRI channel and then send to a SipXecs > extension which has a forwarding rule to ring the extension and a > mobile device at the same time > asterisk quickly cancels the call to the extension while allowing the > mobile to ring. > > I have 2 media gateways and 2 sipxecs proxies this behavior is not > happening when the call comes from GW2 then gets forwarded out GW1 (or > vice versa) > > Call --> PRI ---> Asterisk PRI GW 1 ---> Sipxecs (Forward Rule > "simultaneous ring") ---> Asterisk PRI GW 1 ----> (SipX Exten + > Mobile) SipX Exten Rings One Time while mobile rings as expected. > > Some of my calls come in another gateway and when this happens the > call is handled properly: > > Call --> PRI ---> Asterisk PRI GW 2 ---> Sipxecs (Forward Rule > "simultaneous ring") ---> Asterisk PRI GW 1 ----> (SipX Exten + > Mobile) Expected result both extensions ring > > Both Asterisk PRI GWs are set up as unmanaged gateways in sipxecs.
> It seem that asterisk just wants to forward the call to the mobile > device and cancel the extens call > Can anyone advise me on a working config for this ? Your question is almost certainly better asked in an Asterisk forum. It sounds like the problem you have is that your installation of Asterisk is not properly supporting a hairpinned call. When sipXecs forwards a call back to the same gateway it came from, that gateway will see the same INVITE that it sent (that is, the same call-id and from tag values) with some new Via headers, a new Record-Route header, and (importantly) a new request-uri value. Some gateways (apparently including yours) are confused by this. It would be easier for us to confirm that diagnosis if you trace the message flow from sipXecs. See: http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer#Getting_SIP_Messages_to_display when you get the trace data, take a look at it using sipviewer and/or post the trace with a description of your configuration (identify components by IP address), what you were doing, and which call in the trace you're talking about (by call-id or frame number in the trace, preferably). (when you get it, don't paste the messages into the body of the email - we have nice tools for displaying and analyzing logs that are messed up by the formatting and line folding the email programs do - attach the data as a text file) _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
