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?
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.
_____
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.
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