Running callweaver version 1.2.0.1 (CallWeaver 1.2.0.1 built on voip, an
x86_64 running Linux on 2008-07-04 15:12:05 UTC) compiled on Suse Linux
Enterprise 10. uname -a reports: Linux voip 2.6.16.60-0.23-smp #1 SMP
Thu May 15 06:38:31 UTC 2008 x86_64 x86_64 x86_64 GNU/Linux
When attempting to establish SIP connection to the outside world I get:
-- Executing [EMAIL PROTECTED]:1] Set("SIP/1000-43d7",
"CALLERID(number)=1888#######")
-- Executing [EMAIL PROTECTED]:2] Set("SIP/1000-43d7",
"TIMEOUT(digit)=5")
-- SIP/1000-43d7 digit timeout set to 5
-- Executing [EMAIL PROTECTED]:3] Dial("SIP/1000-43d7",
"SIP/[EMAIL PROTECTED],60")
-- Called [EMAIL PROTECTED]
-- SIP/jnctn-b588 is making progress passing it to SIP/1000-43d7
-- SIP/jnctn-b588 answered SIP/1000-43d7
Jul 7 14:27:09 WARNING[1093720384]: chan_sip.c:12377
handle_response_invite: Strange... The other side of the bridge don't
have udptl struct
Jul 7 14:27:09 WARNING[1093720384]: chan_sip.c:12377
handle_response_invite: Strange... The other side of the bridge don't
have udptl struct
Jul 7 14:27:09 WARNING[1093720384]: chan_sip.c:12377
handle_response_invite: Strange... The other side of the bridge don't
have udptl struct
== Spawn extension (outgoing, 483####, 3) exited non-zero on
'SIP/1000-43d7'
The call is connected but no audio is heard.
My sip.conf looks like:
[general]
register => XXXXX:[EMAIL PROTECTED]
context=jnctn
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
registertimeout=30
registerattempts=0
[jnctn]
fromdomain=jnctn.net
host=sip.jnctn.net
insecure=very
username=xxxxx
secret=XxXxXxXx
type=peer
My extensions.conf looks like:
exten => _NXXXXXX,1,Set(CALLERID(number)=1888#######)
exten => _NXXXXXX,2,Set(TIMEOUT(digit)=5)
exten => _NXXXXXX,3,Dial(SIP/[EMAIL PROTECTED],60)
exten => _NXXXXXX,104,Congestion()
exten => _NXXXXXX,105,Busy()
The "WARNING[1093720384]: chan_sip.c:12377 handle_response_invite:
Strange... The other side of the bridge don't have udptl struct" can be
gotten rid of by putting a "t38udptlsupport=yes" line in sip.conf but
there still isn't any audio from the call.
The dialplan and everything else works just fine if I dial using IAX2,
but not with SIP.
Can anyone help?
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