> > On Mon, 2008-07-07 at 16:58 -0700, IT Services wrote: > > 1. call forwarding is not working when forwarding to an > external number. > > The PSTN gateway receives [EMAIL PROTECTED] (the ext. #) rather > > than [EMAIL PROTECTED] (the external number). Call > forwarding > > works when forwarding to another internal extension. > > > > 2. the trace pointed out SSL errors and/or issues. > > > > Any help is appreciated! > > It looks like call forwarding is working in the logs you > sent. Looking at the first call that contains 7934068, which > is Call-ID: > [EMAIL PROTECTED], I > see the following messages: > > The initial INVITE: > > Time: 2008-07-03T19:06:49.814075Z > Frame: 398 _.sipXproxy.xml:1142 > Source: 192.168.1.57:5060 > Dest: sipx.apiwc.local-SipXProxy > > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > > The contacts for 343: > > Time: 2008-07-03T19:06:49.876390Z > Frame: 402 _.sipXproxy.xml:1146 _.sipregistrar.xml:206 > Source: sipx.apiwc.local-SipRegistrar > Dest: sipx.apiwc.local-SipXProxy > > SIP/2.0 302 Moved Temporarily > Contact: > <sip:[EMAIL PROTECTED]:5060?ROUTE=%3Csip%3A192.168.1.49%3A5060%3Blr%3E> > Contact: > <sip:[EMAIL PROTECTED];sipx-noroute=Voicemail?expires=1 0&X-Sipx-Authidentity=%3Csip%3A343%40sipx.apiwc.local%> 3Bsignature%3D486D2349%253A7915ec49f7c847ed9d7b79d2989f9c6f%3E &ROUTE=%3Csip%3A192.168.1.49%3A5060%3Blr%3E>;q=0.9 > Contact: > <sip:[EMAIL PROTECTED] 060%3Blr%3E>;q=0.1 > > 7934068 is routed to the gateway at 192.168.1.40: > > Time: 2008-07-03T19:06:50.233256Z > Frame: 416 _.sipXproxy.xml:1176 _.sipregistrar.xml:211 > Source: sipx.apiwc.local-SipRegistrar > Dest: sipx.apiwc.local-SipXProxy > > SIP/2.0 302 Moved Temporarily > Contact: > <sip:[EMAIL PROTECTED];sipx-noroute=Voicemail?ROUTE=%3Csip% > 3A192.168.1.49%3A5060%3Blr%3E>;q=0.9 > > It gets interesting once the call is sent to the gateway. > The INVITE goes to the gateway at _.sipXproxy.xml:1186 > 2008-07-03T19:06:50.335646Z, the gateway responds 180 at > _.sipXproxy.xml:1188 2008-07-03T19:06:51.413693Z, and then > responds 200 at _.sipXproxy.xml:1190 > 2008-07-03T19:06:51.427104Z. Which is strange, since it > seems unlikely that the callee will answer 0.01 second after > his phone rings.
If the gateway is an FXO gateway, it may not have any answer supervision which means that it has no ability to know when the called party actually answers the line. Such gateways usually answer the call right away and stream whatever is coming from the CO line (which will be ringback until the called party answers). Gateways such as Audiocodes implement special algorithms that delay answering the call until they detect voice or fax tones coming from the CO. > The caller eventually hangs up at _.sipXproxy.xml:1258 > 2008-07-03T19:07:26.136773Z. > _______________________________________________ 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
