On Tue, Feb 16, 2010 at 7:22 PM, Andrew Cotter < [email protected]> wrote:
> 1. Did you make sure Internet calling was disabled? > Disabled > > > 2. When you created the siptrunk, did you use the ATT template? > Yes - ATT template > One thing they have not giving me (so I put the SIP server IP in) > is the "ITSP server domain name" tech claims she does not know it. Would > that do this? > > I hope not. i think it would be fine. > > 3. Is AT&T behind your firewall or in front? > AT&T Cisco 7604 port 0/3 with IP 172.21.210.9 --- switch --- sipx > (IP 172.21.210.10) > Internal to us at that point. > > > 4. Is the address of your sipx server added as a domain alias? > Yes > > > 5. Under Internet calling, did you ensure only your subnets are listed > under Intranet? > I had a couple of different subnets listed. Must have been > defaults? Removed and have only the 172.21.210.0/24 subnet now. > > > What is checked under NAT Traversal there? > Nothing is checked. > Is AT&T on your network as an IP or do you traverse a firewall? > On our network. > > Did you choose the sipxbridge-1 sbc as the route when you created the gateway? I'm unfamiliar with the "flex ip" product. So I would want to know if they are sending REFER or providing MOH. I would ask them. There should be no NAT compensation from them. They should be sending all calls to sipxbridge on port 5080, this is ONLY for incoming calls to setup the calls, inprogress calls would use 5060 because the call is already established, so that's OK. I would want to make sure they are not doing NAT, MOH or have turned off REFER on their end if they have it on. Then I would do a siptrace of a failed call. Anytime you make changes (like remove intranet subnets) you should send all profiles from the server and restart any services. Note this could break remote workers and calls in progress too. > ------------------------------ > *From:* Tony Graziano [mailto:[email protected]] > *Sent:* Tuesday, February 16, 2010 7:09 PM > *To:* Andrew Cotter > > *Cc:* [email protected] > *Subject:* Re: [sipx-users] Problem with transfers from external calls > > 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.
_______________________________________________ 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/
