I'm running 4.2.1 I have just confirmed this has issues with Aastra phones as well.
I've been saying for a while that FreeSWITCH has issues with the way attended transfers are handled. http://wiki.sipfoundry.org/display/xecsuserV4r2/Custom+FreeSWITCH+programming is a prime example. Fix the FreeSWITCH SIP stack and these issues will probably go away. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/16/2010 4:34 PM, Tony Graziano wrote: > Sipxbridge Is not involved in the call scenario, which he details out. > This surely needs to be addressed though. I'll try to compare a call > tomorrow on a 4.0.4 system with the same gateway. > ============================ > 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: 'Josh Patten'<[email protected]>; [email protected] > <[email protected]> > Sent: Fri Jul 16 17:31:01 2010 > Subject: Re: [sipx-users] Help with Patton gateway > > There was an in issue with sipxbridge and attended transfers back in 4.0.2 I > believe. There was a patch for it and it was fixed in 4.0.4. Not sure if > this is your issue, I don't see what version you are running on this system. > > > > From: [email protected] > [mailto:[email protected]] On Behalf Of Josh Patten > Sent: Friday, July 16, 2010 2:04 PM > To: [email protected] > Subject: Re: [sipx-users] Help with Patton gateway > > > > Also I don't think it's a Polycom problem. This only happens when doing > attended transfer to FreeSWITCH services. Attended transfer to everything > else (including park) works fine. > > > > Josh Patten > Assistant Network Administrator > Brazos County IT Dept. > (979) 361-4676 > > > On 7/16/2010 4:00 PM, Josh Patten wrote: > > http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip550.html > > I don't see it > > > > Josh Patten > Assistant Network Administrator > Brazos County IT Dept. > (979) 361-4676 > > > On 7/16/2010 3:49 PM, Tony Graziano wrote: > > 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]<mailto: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%7E.*from%7 > EMTRCRjc0MkYtNzZERkIzMTY$60.900_ntap*id%7EMjg4ODUtNTg4NA$60$60%21df02a35b10e > f9ba395dee26f5cb05618> > <sip:10.200.24.250:5060;lr;sipXecs-CallDest=LOCL;sipXecs-rs=*auth~.*from~MTR > CRjc0MkYtNzZERkIzMTY$60.900_ntap*id~Mjg4ODUtNTg4NA$60$60!df02a35b10ef9ba395d > ee26f5cb05618> > Max-Forwards: 70 > From:<sip:[email protected] > <mailto:sip%[email protected]> ;user=phone>;tag=1105402544 > To: "Josh Patten"<sip:[email protected] > <mailto: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] > <mailto:sip%[email protected]> ;user=phone>;tag=1105402544 > To: "Josh Patten"<sip:[email protected] > <mailto: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/
