Douglas Garstang
2006-Mar-03 15:19 UTC
[Asterisk-Users] Call Transfer - "Both legs must reside on Asterisk box to transfer at this time"
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@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@216.188.140.203 SIP/2.0
Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B
From: <sip:3254102@216.188.128.11>;tag=AD42A97D-626BB596
To: "Douglas Garstang"
<sip:2944093@216.188.140.203>;tag=as6202b08e
CSeq: 2 REFER
Call-ID: 798757066df2b4824ef9224626a8f872@216.188.140.203
Contact: <sip:3254102@216.188.128.11>
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Refer-To:
<sip:3254104@ipt.oneeighty.com;user=phone?Replaces=77a7b64e-f546fcbc-f206df35%40172.31.16.67%3Bto-tag%3Das4744b9fa%3Bfrom-tag%3D200C85AA-7A3B0AE3>
Referred-By: <sip:3254102@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@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@216.188.128.11>;tag=AD42A97D-626BB596
To: "Douglas Garstang"
<sip:2944093@216.188.140.203>;tag=as6202b08e
Call-ID: 798757066df2b4824ef9224626a8f872@216.188.140.203
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:2944093@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@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@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@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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>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Douglas Garstang
2006-Mar-06 08:15 UTC
[Asterisk-Users] Call Transfer - "Both legs must reside on Asterisk box to transfer at this time"
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@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@216.188.140.203 SIP/2.0
Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B
From: <sip:3254102@216.188.128.11>;tag=AD42A97D-626BB596
To: "Douglas Garstang"
<sip:2944093@216.188.140.203>;tag=as6202b08e
CSeq: 2 REFER
Call-ID: 798757066df2b4824ef9224626a8f872@216.188.140.203
Contact: <sip:3254102@216.188.128.11>
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Refer-To:
<sip:3254104@ipt.oneeighty.com;user=phone?Replaces=77a7b64e-f546fcbc-f206df35%40172.31.16.67%3Bto-tag%3Das4744b9fa%3Bfrom-tag%3D200C85AA-7A3B0AE3>
Referred-By: <sip:3254102@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@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@216.188.128.11>;tag=AD42A97D-626BB596
To: "Douglas Garstang"
<sip:2944093@216.188.140.203>;tag=as6202b08e
Call-ID: 798757066df2b4824ef9224626a8f872@216.188.140.203
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:2944093@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@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@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@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?
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users