[asterisk-dev] Recent trunk versions: SIP calls all borked

Brian Capouch brianc at palaver.net
Sat May 10 00:39:02 CDT 2008

There was a long while I wasn't able to do my regular builds.  I have 
encountered a few issues I'd like to vet here before opening up a bug 

The last version I built that works OK is r110444M.  Several recent 
versions show this problem, the grabs below are from r115595M built an 
hour or so ago.

Reversion to previous build fixes things.  It seems to happen anytime I 
try to make a call out over a SIP channel.  That's true for both 
software-only providers as well as my hardware SPA3000.

I get these CLI errors after placing a call to a SIP provider from a SIP 
ATA.  Afterwards the server is hosed, and has to be killed from a shell.

      -- Executing Dial("SIP/spa3k-006a6e30", 
"SIP/12125551212 at proxy01.sipphone.com")
   == Using SIP RTP CoS mark 5
     -- Called 12198666114 at proxy01.sipphone.com
     -- SIP/proxy01.sipphone.com-006ac888 answered SIP/spa3k-006a6e30
     -- Packet2Packet bridging SIP/spa3k-006a6e30 and 
[May 10 00:59:33] ERROR[4713]: chan_sip.c:19213 handle_request_do: We 
could NOT get the channel lock for SIP/proxy01.sipphone.com-006ac888!
[May 10 00:59:33] ERROR[4713]: chan_sip.c:19214 handle_request_do: SIP 
transaction failed: 3ff3ee313033b1872931e36b08b63d3e at proxy01.sipphone.com



This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

More information about the asterisk-dev mailing list