Ahh Shoretel. Worked with their product in 2006. The reason why you can't use Polycom phones is that Shoretel is an MGCP based platform with ONLY sip TRUNKING abilities (not SIP handset). Their handsets are quite expensive too from what I remember.
It looks like an InGate may work for you though. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 3/15/2010 2:19 PM, Andrew Cotter wrote: > I belive the transfer issue is due to the REFER command not being supported > and port 5080 not being an option. > > Calls come in to the numbers they are supposed to. Each DID answers just > fine. When a user tries to transfer the call it gets dead air (and > polycom's do a re-invite if I remember correctly). This also effects the > auto attendant so that if a caller get the message and tries to dial the > extension or by name, again dead air and no transfer. > > We have an audiocodes MP118 setup on copper that handles allows transfers to > happen all day long. > > Shoretel > > Andrew > > > >> -----Original Message----- >> From: [email protected] >> [mailto:[email protected]] On Behalf Of >> Josh Patten >> Sent: Monday, March 15, 2010 3:10 PM >> To: [email protected] >> Subject: Re: [sipx-users] One last attempt - AT&T IP Flex >> >> Could you remind me again why you couldn't use something like >> an Ingate to do the translation for you? It's been a while >> since you posted and my memory isn't that good. >> >> Also, the commercial system you are describing almost sounds >> like an Alcatel. I thought Alcatel's configuration UI was >> atrocious but I really liked their phones. >> >> Josh Patten >> Assistant Network Administrator >> Brazos County IT Dept. >> (979) 361-4676 >> >> >> On 3/15/2010 2:06 PM, Andrew Cotter wrote: >> >>> So I have finally gotten word from AT&T labs that they will not be >>> able to support SipX and fix our transfer issue. >>> >>> We have a SIP handoff that is direct (switch in the middle) >>> >> from their >> >>> Cisco router onsite. I asked for them to send signaling on >>> >> port 5080 >> >>> (sipXbridge) but that was a no go. Then I asked if they >>> >> can do some >> >>> sort of NAT translation for incoming data from their end, >>> >> through the >> >>> router, and into port 5080. Again, no go as they tested >>> >> this in the >> >>> labs. B2BUA on the Cisco, nope. >>> >>> So... I am left with probably having to leave my sipX setup, that I >>> have come to know and love, behind. >>> >>> A final question for the masses: >>> >>> Would having AT&T swap out the SIP handoff for a PRI handoff >>> potentially fix my transfer issues if I put a gateway in? If this >>> would work and I can convince AT&T to convert the SIP >>> >> handoff to a PRI >> >>> handoff, what solution would you suggest (patton, >>> >> audiocodes, etc.) to >> >>> handle a single PRI. I have >>> 4 sites spread throughout the US and would need something >>> >> fairly cost >> >>> effective for 2 of them since there are 5 or less employees >>> >> at those sites. >> >>> I am sure I will have more questions if people come back >>> >> saying this >> >>> might resolve the issues. >>> >>> >>> Parting thoughts. >>> In light of the position I am now in I am forced to begin to look >>> elsewhere at commercial products. I wanted to share my thoughts on >>> the comparison of sipx and a well known commercial product >>> >> out there. >> >>> After getting a demo of one solution that the salesperson >>> >> was touting >> >>> as an extremely easy interface, so simple a cave man can >>> >> set it up, I >> >>> was amazed at how much I was left desiring the simplicity of SipX. >>> The screens were cluttered, the interface was fairly well >>> >> organized, >> >>> but the voicemail and admin console still resided on a >>> >> windows machine. Not what I want. >> >>> Yes it was a nice system in terms of failover and distribution, but >>> they pretty much insist that we swap out our phones (polycom) for >>> their own phones. Also, for a VoIP system they almost left me >>> speechless when they said I could only use one SIP trunk >>> >> provider unless I bought an InGate. >> >>> VoIP... SIP... Won't support it? Wow! Don't even get me >>> >> started on >> >>> the Windows application or the Outlook piece that I repeatedly told >>> them we would not be using. >>> >>> Thank you again for everyone's help and suggestions over the past >>> month in trying to make this work. If I can slip in a plug for the >>> project during my talk at the Computerworld OSBC later this >>> >> week I will. >> >>> >>> Andrew >>> >>> _______________________________________________ >>> 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/ >> >> > _______________________________________________ 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/
