Scenario is now one Polycom 550 registered to sipX, receiving a call from the outside. No change in the gateway-to-sipX messaging (call still answered by VM instead of forwarded). Gateway is .53.11, sipX is .45.46.
0.000000 192.168.53.11 -> 192.168.54.46 SIP/SDP Request: INVITE sip:[email protected]:5060, with session description 0.001530 192.168.54.46 -> 192.168.53.11 SIP Status: 100 Trying 0.269556 192.168.54.46 -> 192.168.53.11 SIP Status: 180 Ringing 4.136557 192.168.54.46 -> 192.168.53.11 SIP/SDP Status: 200 OK, with session description 4.152206 192.168.53.11 -> 192.168.54.46 SIP Request: ACK sip:[email protected]:15060;transport=udp 6.856499 192.168.53.11 -> 192.168.54.46 SIP Request: BYE sip:[email protected]:15060;transport=udp 6.864558 192.168.54.46 -> 192.168.53.11 SIP Status: 200 OK The sipX-to-Poly messaging, Poly is .201.60. TCP ACKs removed for clarity. 0.000021 192.168.54.46 -> 172.21.201.60 SIP/SDP Request: INVITE sip:[email protected];x-sipX-nonat, with session description 0.064324 172.21.201.60 -> 192.168.54.46 SIP Status: 100 Trying 0.230543 172.21.201.60 -> 192.168.54.46 SIP Status: 180 Ringing 3.993589 192.168.54.46 -> 172.21.201.60 SIP Request: CANCEL sip:[email protected];x-sipX-nonat 4.045358 172.21.201.60 -> 192.168.54.46 SIP Status: 200 OK 4.079853 172.21.201.60 -> 192.168.54.46 SIP Status: 487 Request Cancelled 4.080990 192.168.54.46 -> 172.21.201.60 SIP Request: ACK sip:[email protected];x-sipX-nonat I'm new to sipX, but not to SIP. All this messaging looks appropriate to me. The only call forward scenario that doesn't work for me is when an external caller hits a user set to forward to an external number - the call goes to the user's VM instead. Every other internal/external caller/callee combination works as expected. - Jeff On Wed, Sep 19, 2012 at 11:59 AM, Todd Hodgen <[email protected]> wrote: > Have you tried removing all the phones but one on that called user to > ensure its not a bad behaving endpoint?**** > > ** ** > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Jeff Pyle > *Sent:* Wednesday, September 19, 2012 7:16 AM > *To:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] Call forward fails to external number**** > > ** ** > > Adtran TA908e. I manage it. CLEC (network) PRI behind it.**** > > ** ** > > The user's extension is set to ring first for 4 seconds, then forward to a > 10-digit PSTN number. The PRI on the gateway will accept 7 or 10-digit > local number and the gateway is set to pass through whatever it receives. > I can dial a 7 or 10-digit local number just fine from a local user, > including the one I'm using to test this forwarding. If I call this local > user from another local user, the forward works correctly. Just not if an > outside user calls in.**** > > ** ** > > Here is the tshark outbound of the entire call flow from the perspective > of that gateway, starting with the inbound call from the PRI sent to sipX. > 192.168.53.11 is the gateway; 192.168.54.46 is the sipX system. > 2160000000 is not the real DID.**** > > ** ** > > 0.000000 192.168.53.11 -> 192.168.54.46 SIP/SDP Request: INVITE > sip:[email protected]:5060, with session description**** > > 0.002137 192.168.54.46 -> 192.168.53.11 SIP Status: 100 Trying**** > > 0.141508 192.168.54.46 -> 192.168.53.11 SIP Status: 180 Ringing**** > > 0.181202 192.168.54.46 -> 192.168.53.11 SIP Status: 180 Ringing**** > > 0.205498 192.168.54.46 -> 192.168.53.11 SIP Status: 180 Ringing**** > > 0.300094 192.168.54.46 -> 192.168.53.11 SIP Status: 180 Ringing**** > > 4.154583 192.168.54.46 -> 192.168.53.11 SIP/SDP Status: 200 OK, with > session description**** > > 4.171045 192.168.53.11 -> 192.168.54.46 SIP Request: ACK > sip:[email protected]:15060;transport=udp**** > > ** ** > > -{ caller hears VM system }-**** > > ** ** > > 7.896108 192.168.53.11 -> 192.168.54.46 SIP Request: BYE > sip:[email protected]:15060;transport=udp**** > > 7.909495 192.168.54.46 -> 192.168.53.11 SIP Status: 200 OK**** > > ** ** > > There are 4 registered devices on the called user, hence the 4x 180 > Ringing messages.**** > > ** ** > > I would expect to see an INVITE or REFER sent to the gateway at > call-forward time instead of the 200 OK of the VM system. It seems like > something is preventing the system from even trying to send the call.**** > > ** ** > > ** ** > > - Jeff**** > > ** ** > > ** ** > > On Wed, Sep 19, 2012 at 9:56 AM, Tony Graziano < > [email protected]> wrote:**** > > what kind of gateway/who is the telco?**** > > -- > ~~~~~~~~~~~~~~~~~~ > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.465.6833 > ~~~~~~~~~~~~~~~~~~ > Linked-In Profile: > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > Ask about our Internet Fax services! > ~~~~~~~~~~~~~~~~~~**** > > Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab > 2013!**** > > On Sep 19, 2012 9:50 AM, "Jeff Pyle" <[email protected]> wrote:**** > > Hi Tony,**** > > ** ** > > For this testing, no ITSP, just a local PRI gateway. Although we're not > that far yet - sipX never sends the INVITE to gateway for the outbound leg > of the forward, so no SIP trace.**** > > ** ** > > This is day 3 a new install atop Centos 6.3 (not the ISO). Very little > fiddling so far.**** > > ** ** > > ** ** > > - Jeff**** > > ** ** > > ** ** > > ** ** > > On Wed, Sep 19, 2012 at 2:46 AM, Tony Graziano < > [email protected]> wrote:**** > > It will depend entirely upon the itsp or telco provider.**** > > Some itsp's do not actually support "hair pinned" calls.**** > > I ran into an instance recently where the outbound call (forward) was a > local call but we had to use a 10 digit number instead of 7 "only" for the > forward.**** > > A siptrace would be helpful.**** > > -- > ~~~~~~~~~~~~~~~~~~ > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.465.6833 > ~~~~~~~~~~~~~~~~~~ > Linked-In Profile: > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > Ask about our Internet Fax services! > ~~~~~~~~~~~~~~~~~~**** > > Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab > 2013!**** > > On Sep 18, 2012 10:24 PM, "Jeff Pyle" <[email protected]> wrote:**** > > Hello, **** > > ** ** > > What must one do in 4.6 to allow a local user to forward a call to an > external number?**** > > ** ** > > Here is what I have now: User A can dial a 10-digit outside number and it > routes out the gateway correctly. User A can include the outside number in > his call forwarding configuration, and if User B calls User A, the call > forward works correctly. But if User A receives a call from the outside, > the outside caller hits User A's voicemail instead of forwarding to the > outside number.**** > > ** ** > > All other inbound and outbound calling through the gateway seems to work > okay.**** > > ** ** > > As far as I can tell all my permissions and dial-plans are configured and > enabled correctly. sipXproxy.log isn't helping much. What might I check > next?**** > > ** ** > > ** ** > > ** ** > > - Jeff**** > > ** ** > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/**** > > ** ** > > LAN/Telephony/Security and Control Systems Helpdesk:**** > > Telephone: 434.984.8426**** > > sip: [email protected]**** > > ** ** > > Helpdesk Customers: http://myhelp.myitdepartment.net**** > > Blog: http://blog.myitdepartment.net**** > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/**** > > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/**** > > ** ** > > LAN/Telephony/Security and Control Systems Helpdesk:**** > > Telephone: 434.984.8426**** > > sip: [email protected]**** > > ** ** > > Helpdesk Customers: http://myhelp.myitdepartment.net**** > > Blog: http://blog.myitdepartment.net**** > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/**** > > ** ** > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >
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