FWIW - Firmware 3.3.0 is now posted... though you may still have the same problem.
On Fri, Jul 16, 2010 at 12:36 PM, Josh Patten <[email protected]>wrote: > 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] <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]<sip%[email protected]> >>> ;user=phone>;tag=1105402544 >>> To: "Josh >>> Patten"<sip:[email protected]<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]<sip%[email protected]> >>> ;user=phone>;tag=1105402544 >>> To: "Josh >>> Patten"<sip:[email protected]<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/ >>> >>> >> -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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