I have now confirmed this is not a problem with the gateways. I posted a ticket here: http://track.sipfoundry.org/browse/XX-8652 Even with firmware 3.1.3revC this is still happening.
Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/16/2010 10:29 AM, Tony Graziano wrote: > 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/
