[Asterisk-Users] Call Transfer - "Both legs must reside on Asterisk box to transfer at this time"

[Douglas Garstang]

I have a SIP user, 2944093 that dialled 3254102. I'm trying to transfer the call from 3254102 to 3254104. When I try and transfer the call, I get the following on the Asterisk console.

Mar  3 15:14:18 NOTICE[23124]: chan_sip.c:6731 get_refer_info: Supervised transfer requested, but unable to find callid '16749440-c28be02e-64b73be7 at 172.31.16.67'.  Both legs must reside on Asterisk box to transfer at this time.

Below is what my SIP debug console output shows me. IP 216.188.128.11 is the phone that the transferer is on (3254102). It sends a REFER message to Asterisk. Asterisk turns around and says 'Not found' eventhough the destination user, 3254104, is in it's database. I wonder if this is because the REFER has Asterisks's IP address and not the IP address of the phone? How could it have gotten that way? 

Thanks,
Doug.

--- (10 headers 0 lines)---
    -- SIP/3254104-a911 is ringing

<-- SIP read from 216.188.128.11:5060: 
REFER sip:2944093 at 216.188.140.203 SIP/2.0
Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B
From: <sip:3254102 at 216.188.128.11>;tag=AD42A97D-626BB596
To: "Douglas Garstang" <sip:2944093 at 216.188.140.203>;tag=as6202b08e
CSeq: 2 REFER
Call-ID: 798757066df2b4824ef9224626a8f872 at 216.188.140.203
Contact: <sip:3254102 at 216.188.128.11>
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Refer-To: <sip:3254104 at ipt.oneeighty.com;user=phone?Replaces=77a7b64e-f546fcbc-f206df35%40172.31.16.67%3Bto-tag%3Das4744b9fa%3Bfrom-tag%3D200C85AA-7A3B0AE3>
Referred-By: <sip:3254102 at 216.188.128.11>
Max-Forwards: 70
Content-Length: 0


--- (12 headers 0 lines)---
Transfer to 3254104 in From_OneEighty
Transfer from 3254102 in From_OneEighty
Mar  3 14:32:49 NOTICE[16519]: chan_sip.c:6731 get_refer_info: Supervised transfer requested, but unable to find callid '77a7b64e-f546fcbc-f206df35 at 172.31.16.67'.  Both legs must reside on Asterisk box to transfer at this time.
Reliably Transmitting (no NAT) to 216.188.128.11:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B;received=216.188.128.11
From: <sip:3254102 at 216.188.128.11>;tag=AD42A97D-626BB596
To: "Douglas Garstang" <sip:2944093 at 216.188.140.203>;tag=as6202b08e
Call-ID: 798757066df2b4824ef9224626a8f872 at 216.188.140.203
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:2944093 at 216.188.140.203>
Accept: application/sdp
Content-Length: 0

Here's the database entry for the destination number:
/SIP/Registry/3254104                             : 216.188.128.12:5060:3600:3254104:sip:3254104 at 216.188.128.12

As you can see, that isn't what the REFER has. It has 216.188.140.203, which is Asterisks IP address. I don't know if that's the issue or not. Asterisk _IS_ in the RTP path.

Doug.


-----Original Message-----
From: David Thomas [mailto:punknow at gmail.com]
Sent: Friday, March 03, 2006 2:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Hardware Requirements for 1M minutes


Sorry, I saw that right after I posted.

It is per month. And almost all during business hours.

regards,
David

On 3/3/06, Martin Joseph <ast at stillnewt.org> wrote:
>
> On Mar 3, 2006, at 9:49 AM, David Thomas wrote:
>
> > I'm doing an install for a client with the following requirements.
> >
> > - 1 Million minutes of outbound calling
>
> Per what?
>
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