So the question still remains if it happens with firmware 3.13RevC. Its the polycom complaining... 3.2 aint all that. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431
Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: [email protected] <[email protected]> To: [email protected] <[email protected]> Sent: Fri Jul 16 11:27:35 2010 Subject: Re: [sipx-users] Help with Patton gateway EDIT - This also seems to be occuring with my Audiocodes gateways as well so apparently it's not isolated just to the Patton gateways. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/16/2010 10:23 AM, Josh Patten wrote: > I'm forwarding the support request I sent to Patton regarding a > problem with their gateways and sipX. Here is where the engineer said > things are going wrong: > > Line 1068, the smartnode sends BYE to polycom to ip 10.200.24.250 as > showed below: > > 23:39:24 SIP_TR> [STACK] > Stack: to 10.200.24.250 > BYE sip:[email protected];x-sipX-nonat SIP/2.0 > Via: SIP/2.0/UDP 10.200.50.11:5060;branch=z9hG4bKaba231ea2aa038eb1 > Route: > <sip:10.200.24.250:5060;lr;sipXecs-CallDest=LOCL;sipXecs-rs=*auth~.*from~MTRCRjc0MkYtNzZERkIzMTY$60.900_ntap*id~Mjg4ODUtNTg4NA$60$60!df02a35b10ef9ba395dee26f5cb05618> > Max-Forwards: 70 > From: <sip:[email protected];user=phone>;tag=1105402544 > To: "Josh Patten" <sip:[email protected]>;tag=14BF742F-76DFB316 > Call-ID: [email protected] > <mailto:[email protected]> > CSeq: 12761 BYE > User-Agent: Patton SN4524 JO EUI 00a0ba05061C R5.T 2010-05-20 H323 SIP > FXS FXO M5T SIP Stack/4.0.29.29 > Content-Length: 0 > > > Line 1093, Polycom answered back with message error 481 as showed below: > > 23:39:24 SIP_TR> [STACK] < Stack: from 10.200.24.250 > SIP/2.0 481 Call Leg/Transaction Does Not Exist > Via: SIP/2.0/UDP 10.200.50.11:5060;branch=z9hG4bKaba231ea2aa038eb1 > From: <sip:[email protected];user=phone>;tag=1105402544 > To: "Josh Patten" <sip:[email protected]>;tag=14BF742F-76DFB316 > Cseq: 12761 BYE > Call-Id: [email protected] > <mailto:[email protected]> > User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 > Accept-Language: en > Content-Length: 0 > Date: Tue, 13 Jul 2010 23:39:24 GMT > > Do you know why the polycom is sending this message instead to > terminate the call? > > I honestly don't know why that's happening. Could someone on this list > with a little more SIP knowledge point out where it's going wrong? > > I have attached the original debug and the original email dialog. If I > need to get a snapshot let me know and I will. > > -------- Original Message -------- > Subject: Re: [Support #54038]: Consultative (attended) transfer to > auto attendant in sipXecs causes incomplete transfer on phone > Date: Tue, 13 Jul 2010 18:47:06 -0500 > From: Josh Patten <[email protected]> > To: [email protected] > > > > Debug is attached. > > Here is the call scenario: > > 4676 calls 95745699 > Patton strips the 9, dialing 5745699 > Once connected, 4676 initiates a consultative (attended) transfer to > 4310 which is an auto attendant > After connected to the auto attendant, 4676 completes the consultative > transfer. The call is transferred but appears to be on hold on the > phone. The only way to clear this ghost call is to un-hold then end > the call. > Josh Patten > Assistant Network Administrator > Brazos County IT Dept. > (979) 361-4676 > > On 7/13/2010 2:44 PM, Patton Electronics Technical Support wrote: >> ====== Please reply above this line ====== >> Hello Josh, >> >> Thanks for contacting Patton Support. >> >> Please run these debug commands via telnet and send me the output as >> a .txt file so we can see why the call is not being disconnected: >> >> >> enable >> show running-config >> show port fxo detail 5 >> debug fxo >> debug ccfxo >> debug call-router detail 5 >> debug call-control detail 5 >> debug context sip-gateway transport detail 5 >> debug context sip-gateway error detail 5 >> >> I have attached a debugging tutorial for reference. >> >> Regards, >> >> Daniel Lizaola >> Technical Support Engineer >> Patton Electronics Co >> 7622 Rickenbacker Drive >> Gaithersburg MD 20879 USA >> t: +1 301-975-1000 >> f: +1 301-869-9293 >> w: http://www.patton.com >> >> Please consider your environmental responsibility before printing >> this e-mail. >> >> Ticket History *Josh Patten* (Client) Posted On: 09 Jul 2010 09:54 PM >> ------------------------------------------------------------------------ >> >> Here is the dialing scenario laid out in the attached debug: >> >> 3001 dials 95745699 >> Patton gateway strips 9 off and dials 5745699 on FXO hunt group >> Once connected, 3001 performs an attended transfer to 4310, an auto >> attendant, by pressing transfer then dialing 4310 >> Once connected, 3001 presses transfer again to complete the transfer. >> 5745699 is transfered to the auto attendant, but the call on the >> transferring phone is put on hold (even though it is no longer an active >> call). To end this "ghost call" the user has to resume the ghost call >> then hang up. >> >> I have attached a debug and a copy of my configuration. Please let me >> know if you need anything else. >> >> >> >> Attachments aa_debug.txt (496.75 KB) >> SmartNode-4524.cfg (8.64 KB) >> >> >> >> Ticket Details >> Ticket ID: 54038 >> Department: Support for NA/LA/APAC >> Priority: Standard >> Status: Waiting for Response >> > > _______________________________________________ > sipx-users mailing list [email protected] > 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/ _______________________________________________ sipx-users mailing list [email protected] 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/
