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Summary:ASTERISK-09559: Transfer button on GXP200 cause channel deleting
Reporter:Yann JOUANIN (yannj)Labels:
Date Opened:2007-05-31 12:18:02Date Closed:2011-06-07 14:08:17
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Transfers
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:When trying to use TRNF button on the Grandstream GXP200, the channel is deleted .

here you can find sip debug :

 -- Started music on hold, class 'default', on SIP/204725-082147a8
CSterisk*CLI>
<--- SIP read from 172.25.101.144:5060 --->
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 172.25.101.145:5060;branch=z9hG4bK0cb5d5b9;rport
From: "204906"<sip:204906@172.25.101.145>;tag=as48791aaf
To: <sip:204725@172.25.101.144:5060;user=phone>;tag=c0a80101-2f516a
Call-ID: 2cb2883f0c268de9156360a375357639@172.25.101.145
CSeq: 104 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
   -- Got SIP response 400 "Bad Request" back from 172.25.101.144
set_destination: Parsing <sip:204725@172.25.101.144:5060;user=phone> for address/port to send to
set_destination: set destination to 172.25.101.144, port 5060
Transmitting (no NAT) to 172.25.101.144:5060:
ACK sip:204725@172.25.101.144:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.25.101.145:5060;branch=z9hG4bK0cb5d5b9;rport
Max-Forwards: 70
From: "204906" <sip:204906@172.25.101.145>;tag=as48791aaf
To: <sip:204725@172.25.101.144:5060;user=phone>;tag=c0a80101-2f516a
Contact: <sip:204906@172.25.101.145>
Call-ID: 2cb2883f0c268de9156360a375357639@172.25.101.145
CSeq: 104 ACK
User-Agent: Asterisk PBX SVN-trunk-r66638
Content-Length: 0


---
   -- Stopped music on hold on SIP/204725-082147a8
 == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/204906-b6e02488' in macro 'stdexten'
 == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/204906-b6e02488'
Scheduling destruction of SIP dialog '276baab17dcf4bab@172.25.101.93' in 32000 ms (Method: INVITE)
Comments:By: Joshua C. Colp (jcolp) 2007-06-04 12:49:20

The full sip debug and console output is needed for this issue plus a description of what exactly was going on.

By: Joshua C. Colp (jcolp) 2007-06-19 11:36:54

It's been two weeks and the required information has not been provided. Please reopen if this is still an issue and you have it. Thanks.