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.
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