On Thu, 2008-09-25 at 15:59 -0400, Stephen D. Miller wrote: > > The "treat long distance prefix as optional" is indeed checked (and the > dialplan has been activated). > > That said, I think my core question is, according the applicable SIP > RFCs, is the "To:" field in the INVITE packet of SIP "allowed" to be > different than the argument of the INVITE field? > > If they can be different, then I have a gateway issue because it's > clearly using the "To:" field to form its dialstring. If not, then we > have a sipX issue as its only modifying/updating the INVITE argument > (not the "To:" field).
Yes, they can be different, and the call should be routed only on the request uri (after the INVITE), _not_ on the To header. > I have verified (via tcpdump/wireshark) that the gateway receives from > sipX a SIP packet of this form (i.e. INVITE line with digit '1' > correctly prepended to dialstring, "To:" field without the digit '1' > prepended to dialstring), so it's apparent what's occurring--I'm seeking > which is deemed correct behavior so I can followup and hopefully get it > resolved. > > Thx/all thoughts welcomed- > > sdm > ________________________________ > > From: Tony Graziano [mailto:[EMAIL PROTECTED] > Sent: Thursday, September 25, 2008 3:35 PM > To: [email protected]; Stephen D. Miller > Subject: Re: [sipx-users] Basic SIP INVITE question... > > > Good question. > > I don't think your issue is a gateway issue, rather go to your > dialplan in questyion, click ADVANCED and see if the > option: > > Treat long distance prefix as optional > > is checked, if not CHECK IT, apply, activate, try. Perhaps leave > your proxy log set to debug and tail the log file with: > > tail /var/log/sipxpbx/sipXproxy.log -f > > Post your results. > > >>> "Stephen D. Miller" <[EMAIL PROTECTED]> 09/25/08 > 03:18PM >>> > > In my long distance dial plan, I have sipX (v3.10) prepend the > necessary > 1 for dialing outbound long distance calls over the PSTN. Using > a > network sniff, I have verified that sipX does indeed do this and > creates/sends the following INVITE to the gateway: > > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Record-Route: > > <sip:172.20.1.101:5060;lr;sipXecs-rs=%2Afrom%7EYmIxY2ZiMjA%60.400_authru > les%2Aauth%7E%21eae23d5a72f9f0d65973ce6895b2cf13> > Max-Forwards: 18 > Contact: <sip:[EMAIL PROTECTED]:25948> > To: "9495551212"<sip:[EMAIL PROTECTED]> > From: "Garage"<sip:[EMAIL PROTECTED]>;tag=bb1cfb20 > Call-Id: MDVlNDM1NmU3ODE5ZmI0NjhmNGQzMzdiODJmNzg2ZDQ. > Cseq: 2 INVITE > > ... > > However, the gateway (with no other dialstring-altering dialplan > of its > own) stubbornly dials the number in the "To:" field--not the > target in > the INVITE line of the packet. This call then fails in the PSTN > since > it's missing the leading digit '1'. > > Am I correct in assuming this is the gateway's problem--and not > one > where sipX should be altering the "To:" field of the INVITE? > > Any info/help is appreciated/ > > sdm > > (Gateway=Grandstream GXW4104) > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: > http://list..sipfoundry.org/mailman/listinfo/sipx-users > <http://list.sipfoundry.org/mailman/listinfo/sipx-users> > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
