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

2006-03-06 Thread 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 '[EMAIL PROTECTED]'.  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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B
From: sip:[EMAIL PROTECTED];tag=AD42A97D-626BB596
To: Douglas Garstang sip:[EMAIL PROTECTED];tag=as6202b08e
CSeq: 2 REFER
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Refer-To: sip:[EMAIL 
PROTECTED];user=phone?Replaces=77a7b64e-f546fcbc-f206df35%40172.31.16.67%3Bto-tag%3Das4744b9fa%3Bfrom-tag%3D200C85AA-7A3B0AE3
Referred-By: sip:[EMAIL PROTECTED]
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 '[EMAIL PROTECTED]'.  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:[EMAIL PROTECTED];tag=AD42A97D-626BB596
To: Douglas Garstang sip:[EMAIL PROTECTED];tag=as6202b08e
Call-ID: [EMAIL PROTECTED]
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
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:[EMAIL PROTECTED]

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:[EMAIL PROTECTED]
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 [EMAIL PROTECTED] 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] Call Transfer - Both legs must reside on Asterisk box to transfer at this time

2006-03-03 Thread 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 '[EMAIL PROTECTED]'.  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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 216.188.128.11;branch=z9hG4bKb3056f7489B0729B
From: sip:[EMAIL PROTECTED];tag=AD42A97D-626BB596
To: Douglas Garstang sip:[EMAIL PROTECTED];tag=as6202b08e
CSeq: 2 REFER
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Refer-To: sip:[EMAIL 
PROTECTED];user=phone?Replaces=77a7b64e-f546fcbc-f206df35%40172.31.16.67%3Bto-tag%3Das4744b9fa%3Bfrom-tag%3D200C85AA-7A3B0AE3
Referred-By: sip:[EMAIL PROTECTED]
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 '[EMAIL PROTECTED]'.  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:[EMAIL PROTECTED];tag=AD42A97D-626BB596
To: Douglas Garstang sip:[EMAIL PROTECTED];tag=as6202b08e
Call-ID: [EMAIL PROTECTED]
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
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:[EMAIL PROTECTED]

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:[EMAIL PROTECTED]
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 [EMAIL PROTECTED] 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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