Hi All -
Well, after happily existing in a one office environment with asterisk for a few months, I've now decided to start adding in our other offices with their own * boxes and IAX connections (over VPN). Unfortunately, I'm an idiot and I can't get it to work. I'm having some kind of problem with codecs, I guess, but I don't understand what or why. When trying to use an IAX connection to get to another office, I get:
-- Executing Dial("SIP/68-4ab6", "IAX2/ast33:[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack
Feb 11 14:16:58 WARNING[5653]: channel.c:1898 ast_request: No translator path exists for channel type IAX2 (native 0) to 4
Feb 11 14:16:58 NOTICE[5653]: app_dial.c:800 dial_exec: Unable to create channel of type 'IAX2' (cause 0)
This is particularly confounding because I have all codecs disabled except ulaw (all over, sip devices included). Is it trying to do native bridging? No lo comprendo.
An "iax2 show peers" seems to show that the IAX connection is made between the boxes:
ast33*CLI> iax2 show peers
Name/Username Host Mask Port Status
ast551 192.168.1.130 (S) 255.255.255.255 4569 (T) OK (30 ms)
ast551*CLI> iax2 show peers
Name/Username Host Mask Port Status
ast33 192.168.42.130 (S) 255.255.255.255 4569 (T) OK (30 ms)
Here's my info:
ast551: 192.168.1.130 ast33: 192.168.42.130 Version: CVS-HEAD-11/03/04-14:59:37 (both boxes)
IAX.CONF on ast551: [general] bindport=4569 notransfer=yes disallow=all allow=ulaw
[ast33] type=friend auth=md5 secret=pass context=no-callwaiting host=192.168.42.130 qualify=yes trunk=yes disallow=all allow=ulaw
IAX.CONF on ast33: [general] bindport=4569 disallow=all allow=ulaw
[ast551] type=friend auth=md5 secret=pass context=no-callwaiting host=192.168.1.130 qualify=yes trunk=yes disallow=all allow=ulaw
EXTENSIONS.CONF on ast33: [from-sip] exten => 68,1,Dial(SIP/68,20) exten => 68,2,Voicemail(u118) exten => 68,102,Voicemail(b118) exten => 68,103,Hangup
exten => _[012345]X,1,Dial(IAX2/ast33:[EMAIL PROTECTED]/[EMAIL PROTECTED])
[no-callwaiting] include => from-sip include => outgoing
EXTENSIONS.CONF on ast551: [from-sip] exten => 19,1,SetGroup(${EXTEN}) exten => 19,2,CheckGroup(1) exten => 19,103,Goto(19b,1) exten => 19,3,Dial(SIP/19,20) exten => 19,4,Voicemail(u18) exten => 19,5,Hangup
exten => _6X,1,Dial(IAX2/ast551:[EMAIL PROTECTED]/[EMAIL PROTECTED])
[no-callwaiting] include => from-sip include => outgoing
Thanks for any suggestions! Noah
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