Summary: | ASTERISK-09559: Transfer button on GXP200 cause channel deleting | ||
Reporter: | Yann JOUANIN (yannj) | Labels: | |
Date Opened: | 2007-05-31 12:18:02 | Date Closed: | 2011-06-07 14:08:17 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | 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. |