On Tue, Feb 16, 2010 at 7:55 PM, Tony Graziano <[email protected]> wrote: > Well, the overall issue is understanding what AT&T is doing. A siptrace is > meaningful. It's as if they are bypassing sipxbridge and sipxbridge manages > REFER and they (AT&T) do not support it. Those functions (transfers to > others, voicemail or MOH ) use REFER. AT&T should only be talking to > sipxbridge, not to the phone directly. Get a siptrace.
AT&T has been tested quite well. You can send a sipx-snapshot but before you do that, please be sure your signaling from AT&T is directed to the port 5080. You may want to run wireshark and make sure that it is so. Ranga > > On Tue, Feb 16, 2010 at 7:40 PM, Andrew Cotter > <[email protected]> wrote: >> >> Yes, Route is set to sipXbridge-1. >> >> So close but yet so far! >> >> Looks like an inbound call from my cell phone to my desk phone hangs up if >> I put myself on hold. Same basic issue as with a transfer. >> >> I am going through piece by piece testing one change at a time. >> >> Andrew >> >> ________________________________ >> From: Tony Graziano [mailto:[email protected]] >> Sent: Tuesday, February 16, 2010 7:31 PM >> To: Andrew Cotter >> Cc: [email protected] >> Subject: Re: [sipx-users] Problem with transfers from external calls >> >> I see the AT&T flex Ip is a local gateway. When you created the siptrunk, >> did you choose sipxbridge as the default route for the trunk? >> >> On Tue, Feb 16, 2010 at 7:19 PM, Tony Graziano >> <[email protected]> wrote: >>> >>> forget the codec statement, my mind was elsewhere. >>> >>> On Tue, Feb 16, 2010 at 7:16 PM, Tony Graziano >>> <[email protected]> wrote: >>>> >>>> You might also check the codec order preference on your phones. G711ulaw >>>> and G711alaw should be first. >>>> >>>> On Tue, Feb 16, 2010 at 7:08 PM, Tony Graziano >>>> <[email protected]> wrote: >>>>> >>>>> 1. Did you make sure Internet calling was disabled? >>>>> 2. When you created the siptrunk, did you use the ATT template? >>>>> 3. Is AT&T behind your firewall or in front? >>>>> 4. Is the address of your sipx server added as a domain alias? >>>>> 5. Under Internet calling, did you ensure only your subnets are listed >>>>> under Intranet? What is checked under NAT Traversal there? Is AT&T on your >>>>> network as an IP or do you traverse a firewall> >>>>> The fact that the call fails when dialing the voicemail user (where the >>>>> refer should be held by sipXbridge, but it sounds like something is wrong >>>>> in >>>>> a configuration, so it helps to be sure by asking all these questions), is >>>>> bothersome. It sounds more like a NAT traversal or basic config issue once >>>>> the phone is removed from this. >>>>> >>>>> >>>>> On Tue, Feb 16, 2010 at 6:51 PM, Andrew Cotter >>>>> <[email protected]> wrote: >>>>>> >>>>>> To possibly rule out the Polycom firmware/bootrom issue, I created a >>>>>> phantom user and vmail box. Calling inbound to the AA and selecting the >>>>>> extension gives me the same dead air issue. >>>>>> >>>>>> btw - I think I said this before in previous emails on the subject, >>>>>> but the transfer issue is not a problem on an audiocodes MP118 fxo setup >>>>>> to >>>>>> the same sipx box. >>>>>> >>>>>> Andrew >>>>>> >>>>>> ________________________________ >>>>>> From: [email protected] >>>>>> [mailto:[email protected]] On Behalf Of Andrew >>>>>> Cotter >>>>>> Sent: Tuesday, February 16, 2010 6:47 PM >>>>>> To: 'Tony Graziano' >>>>>> Cc: [email protected] >>>>>> Subject: Re: [sipx-users] Problem with transfers from external calls >>>>>> >>>>>> I have enabled MOH on the sipXbridge-1 and the only parameter on the >>>>>> phones I can find is "musicOnHold.uri" which is blank in two places. I >>>>>> am >>>>>> looking at the group the phones are in and under "Lines | Registration" >>>>>> as >>>>>> well as "Phones | SIP". >>>>>> >>>>>> I added in the SBC SIP config "Public Port" to be 5080 as well. The >>>>>> AT&T tech was watching traffic and she said she saw 5060 traffic after I >>>>>> would try a transfer. >>>>>> >>>>>> So.... If I originate a call from inside to my cell phone, answer, and >>>>>> then transfer (not sure about blind yet) it works with MOH playing. >>>>>> >>>>>> If I call in from my cell to my desk phone, answer, then try the >>>>>> transfer, my cell drops off the call. >>>>>> >>>>>> Closer, but not 100%. Time for a trace? >>>>>> >>>>>> Bootrom 4.2? >>>>>> >>>>>> Andrew >>>>>> >>>>>> ________________________________ >>>>>> From: Tony Graziano [mailto:[email protected]] >>>>>> Sent: Tuesday, February 16, 2010 6:19 PM >>>>>> To: Andrew Cotter >>>>>> Cc: [email protected]; [email protected] >>>>>> Subject: Re: [sipx-users] Problem with transfers from external calls >>>>>> >>>>>> "If" it were me, I'd be using bootrom 4.2.(whatever) but I don't think >>>>>> it's the bootrom.I've been wrong a couple of times. >>>>>> I would make sure MOH is enabled on sipXbridge and that the MOH field >>>>>> is blanked out in the phone via sipxconfig and resend the profiles. >>>>>> If that does not work, I would do a siptrace and post it here. >>>>>> >>>>>> >>>>>> On Tue, Feb 16, 2010 at 6:15 PM, Andrew Cotter >>>>>> <[email protected]> wrote: >>>>>>> >>>>>>> Polycoms 430 and 550. Testing on the 550. >>>>>>> >>>>>>> bootrom is 4.1.4 and firmware is 3.1.3RevC split. >>>>>>> >>>>>>> Andrew >>>>>>> >>>>>>> ________________________________ >>>>>>> From: Tony Graziano [mailto:[email protected]] >>>>>>> Sent: Tuesday, February 16, 2010 6:13 PM >>>>>>> To: Andrew Cotter >>>>>>> Cc: [email protected]; [email protected] >>>>>>> Subject: Re: [sipx-users] Problem with transfers from external calls >>>>>>> >>>>>>> What phone are you using? If Polycom, what bootrom and firmware? >>>>>>> >>>>>>> On Tue, Feb 16, 2010 at 6:11 PM, Andrew Cotter >>>>>>> <[email protected]> wrote: >>>>>>>> >>>>>>>> Well... Got AT&T on the phone and they made the port change. I >>>>>>>> dropped the >>>>>>>> unmanged gateway and can once again make calls in and out. AT&T >>>>>>>> tech >>>>>>>> confirmed UDP traffic on port 5080. >>>>>>>> >>>>>>>> Transfers still don't work. Any thoughts? >>>>>>>> >>>>>>>> Andrew >>>>>>>> >>>>>>>> > -----Original Message----- >>>>>>>> > From: Tony Graziano [mailto:[email protected]] >>>>>>>> > Sent: Tuesday, February 16, 2010 4:15 PM >>>>>>>> > To: [email protected]; >>>>>>>> > [email protected]; [email protected] >>>>>>>> > Subject: Re: [sipx-users] Problem with transfers from external >>>>>>>> > calls >>>>>>>> > >>>>>>>> > Nothing pretty. >>>>>>>> > ============================ >>>>>>>> > 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: Andrew Cotter <[email protected]> >>>>>>>> > To: 'Tony Graziano' <[email protected]>; >>>>>>>> > [email protected] <[email protected]>; >>>>>>>> > [email protected] <[email protected]> >>>>>>>> > Sent: Tue Feb 16 16:13:19 2010 >>>>>>>> > Subject: RE: [sipx-users] Problem with transfers from external >>>>>>>> > calls >>>>>>>> > >>>>>>>> > OK. My guess is they don't. AT&T is not quite like some of >>>>>>>> > the other players out there. I have played with folks like >>>>>>>> > flowroute, junction, etc. >>>>>>>> > that have interfaces. Nothing like that I have heard or from >>>>>>>> > AT&T. >>>>>>>> > >>>>>>>> > Any route to take if the 800 pound gorilla won't budge and >>>>>>>> > has to send the calls to 5060? >>>>>>>> > >>>>>>>> > Andrew >>>>>>>> > >>>>>>>> > >>>>>>>> > > -----Original Message----- >>>>>>>> > > From: Tony Graziano [mailto:[email protected]] >>>>>>>> > > Sent: Tuesday, February 16, 2010 4:04 PM >>>>>>>> > > To: [email protected]; >>>>>>>> > > [email protected]; [email protected] >>>>>>>> > > Subject: Re: [sipx-users] Problem with transfers from external >>>>>>>> > > calls >>>>>>>> > > >>>>>>>> > > I have no idea if they do, "MOST I >>>>>>>> > > ITSP's do". >>>>>>>> > > ============================ >>>>>>>> > > 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: Andrew Cotter <[email protected]> >>>>>>>> > > To: 'Tony Graziano' <[email protected]>; >>>>>>>> > > [email protected] <[email protected]>; >>>>>>>> > > [email protected] <[email protected]> >>>>>>>> > > Sent: Tue Feb 16 16:01:11 2010 >>>>>>>> > > Subject: RE: [sipx-users] Problem with transfers from external >>>>>>>> > > calls >>>>>>>> > > >>>>>>>> > > They have a control panel? That one is news to me. Guess my >>>>>>>> > > salesperson is about to get a call! >>>>>>>> > > >>>>>>>> > > Andrew >>>>>>>> > > >>>>>>>> > > > -----Original Message----- >>>>>>>> > > > From: Tony Graziano [mailto:[email protected]] >>>>>>>> > > > Sent: Tuesday, February 16, 2010 1:36 PM >>>>>>>> > > > To: [email protected]; >>>>>>>> > > > [email protected]; [email protected] >>>>>>>> > > > Subject: Re: [sipx-users] Problem with transfers from >>>>>>>> > external calls >>>>>>>> > > > >>>>>>>> > > > Or log into their control panel and set it yourself. >>>>>>>> > > > ============================ >>>>>>>> > > > 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: 'Picher, Michael' <[email protected]>; >>>>>>>> > > > [email protected] >>>>>>>> > > > <[email protected]> >>>>>>>> > > > Sent: Tue Feb 16 12:52:59 2010 >>>>>>>> > > > Subject: Re: [sipx-users] Problem with transfers from >>>>>>>> > external calls >>>>>>>> > > > >>>>>>>> > > > I know that 5080 is the default port for sipXbridge. To >>>>>>>> > > make sure my >>>>>>>> > > > calls are coming on in 5080, do I need to request that from >>>>>>>> > > > AT&T? >>>>>>>> > > > >>>>>>>> > > > Andrew >>>>>>>> > > > >>>>>>>> > > > > -----Original Message----- >>>>>>>> > > > > From: Picher, Michael [mailto:[email protected]] >>>>>>>> > > > > Sent: Tuesday, February 16, 2010 12:47 PM >>>>>>>> > > > > To: Andrew Cotter; [email protected] >>>>>>>> > > > > Subject: RE: [sipx-users] Problem with transfers from >>>>>>>> > > external calls >>>>>>>> > > > > >>>>>>>> > > > > Make sure that Internet Calling check box is off and that >>>>>>>> > > > your calls >>>>>>>> > > > > are actually coming in on 5080 and not 5060. >>>>>>>> > > > > >>>>>>>> > > > > > -----Original Message----- >>>>>>>> > > > > > From: [email protected] >>>>>>>> > > [mailto:sipx-users- >>>>>>>> > > > > > [email protected]] On Behalf Of Andrew Cotter >>>>>>>> > > > > > Sent: Tuesday, February 16, 2010 5:04 AM >>>>>>>> > > > > > To: [email protected] >>>>>>>> > > > > > Subject: [sipx-users] Problem with transfers from >>>>>>>> > external calls >>>>>>>> > > > > > >>>>>>>> > > > > > Good morning, >>>>>>>> > > > > > >>>>>>>> > > > > > Almost there with our system, but it appears that I have >>>>>>>> > > > > one issue to >>>>>>>> > > > > > resolve. AT&T finally got around to putting in their IP >>>>>>>> > > > > Flex product >>>>>>>> > > > > > and it works well except for transfers. >>>>>>>> > > > > > >>>>>>>> > > > > > The problem shows up in two places which seem like the >>>>>>>> > > same issue. >>>>>>>> > > > > > First, when an outside call comes in to the AA I try >>>>>>>> > > > either dial by >>>>>>>> > > > > > name or dialing the extension. Both end up recognizing >>>>>>>> > > > the tones, >>>>>>>> > > > > > announce that I will be transferred, and then dead air. >>>>>>>> > > > > Phone to be >>>>>>>> > > > > > transferred to never rings. >>>>>>>> > > > > > Second is a DID call to a phone and that phone tries to >>>>>>>> > > > > transfer the >>>>>>>> > > > > > call to another phone. Internal transfers work and >>>>>>>> > > transfers work >>>>>>>> > > > > when >>>>>>>> > > > > > coming in from the copper lines using an AudioCodes MP-118 >>>>>>>> > > > > > fxo >>>>>>>> > > > > gateway. >>>>>>>> > > > > > >>>>>>>> > > > > > Lay of the land.... >>>>>>>> > > > > > >>>>>>>> > > > > > Sipx is set as 172.21.210.10 and is using 4.0.4 >>>>>>>> > > > > > >>>>>>>> > > > > > Cisco 7604 router has three ports, WAN, LAN, SIP. While >>>>>>>> > > > > setting this >>>>>>>> > > > > > up with AT&T it was my understanding that I should have an >>>>>>>> > > > > internal IP >>>>>>>> > > > > > on the SIP port. We had them set it as 172.21.210.9. >>>>>>>> > > > > Seems to work >>>>>>>> > > > > > except for this. >>>>>>>> > > > > > >>>>>>>> > > > > > We setup sipx to use an unmanaged gateway that points to >>>>>>>> > > > > the public IP >>>>>>>> > > > > > AT&T gave us to connect to. >>>>>>>> > > > > > >>>>>>>> > > > > > Attached is a merged trace file that I have been >>>>>>>> > looking at in >>>>>>>> > > > > > sipviewer. >>>>>>>> > > > > > Just wish I knew what I was looking for in this. The two >>>>>>>> > > > > lines that >>>>>>>> > > > > > from a newbie perspective look like possible culprates >>>>>>>> > > > are the "407 >>>>>>>> > > > > > Proxy Authentication Required" and "408 Request timeout". >>>>>>>> > > > > > >>>>>>>> > > > > > Everything I have been reading points be back to >>>>>>>> > REFER, but I am >>>>>>>> > > > > stuck. >>>>>>>> > > > > > Any pointers or thoughts? >>>>>>>> > > > > > >>>>>>>> > > > > > >>>>>>>> > > > > > BTW - sipviewer seems like a great debug tool! >>>>>>>> > > > > > >>>>>>>> > > > > > >>>>>>>> > > > > > 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/ >>>>>>>> > > > >>>>>>>> > > >>>>>>>> > >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> ====================== >>>>>>> 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/ >>>>>>> >>>>>>> Why do mathematicians always confuse Halloween and Christmas? >>>>>>> Because 31 Oct = 25 Dec. >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> ====================== >>>>>> 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/ >>>>>> >>>>>> Why do mathematicians always confuse Halloween and Christmas? >>>>>> Because 31 Oct = 25 Dec. >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> ====================== >>>>> 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/ >>>>> >>>>> Why do mathematicians always confuse Halloween and Christmas? >>>>> Because 31 Oct = 25 Dec. >>>>> >>>> >>>> >>>> >>>> -- >>>> ====================== >>>> 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/ >>>> >>>> Why do mathematicians always confuse Halloween and Christmas? >>>> Because 31 Oct = 25 Dec. >>>> >>> >>> >>> >>> -- >>> ====================== >>> 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/ >>> >>> Why do mathematicians always confuse Halloween and Christmas? >>> Because 31 Oct = 25 Dec. >>> >> >> >> >> -- >> ====================== >> 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/ >> >> Why do mathematicians always confuse Halloween and Christmas? >> Because 31 Oct = 25 Dec. >> > > > > -- > ====================== > 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/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > > > _______________________________________________ > 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/ > -- M. Ranganathan _______________________________________________ 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/
