You need to use sipxbridge or use another SBC. AT&T is dropping the call
because they don't support REFER. You need to follow the steps for
configuring sipxbridge AND a sip trunk for AT&T, not an unmanaged gateway.
============================
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: [email protected] <[email protected]>
Sent: Tue Feb 16 05:04:27 2010
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
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