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].**net<[email protected]> >>> >>> Helpdesk Customers: >>> http://myhelp.myitdepartment.**net<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].**net<[email protected]> > > Helpdesk Customers: > http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net> > Blog: http://blog.myitdepartment.net > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >
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