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:pass@192.168.1.130/08@from-sip") 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:pass@192.168.1.130/${EXTEN}@from-sip) [no-callwaiting] include => from-sip include => outgoing EXTENSIONS.CONF on ast551: [from-sip] exten => 19,1,SetGroup(${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:pass@192.168.42.130/${EXTEN}@from-sip) [no-callwaiting] include => from-sip include => outgoing Thanks for any suggestions! Noah
> 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:pass@192.168.1.130/08@from-sip") 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:pass@192.168.1.130/${EXTEN}@from-sip) > > [no-callwaiting] > include => from-sip > include => outgoing > > > EXTENSIONS.CONF on ast551: > [from-sip] > exten => 19,1,SetGroup(${EXTEN}) > exten => 19,2,CheckGroup(1) > exten => 19,103,Goto(19b,1) > exten => 19,3,Dial(SIP/19,20) > exten => 19,4,Voicemail(u18) > exten => 19,5,Hangup > > exten => _6X,1,Dial(IAX2/ast551:pass@192.168.42.130/${EXTEN}@from-sip) > > [no-callwaiting] > include => from-sip > include => outgoingLooks like you're are getting caught with using "friends" instead of peer and user. Try something like this in iax.conf instead: [abc-inc] ; inbound connections from remote site type=user secret=mysecret context=from-site2 disallow=all allow=ulaw ; supports only ulaw deny=0.0.0.0/0.0.0.0 permit=1.2.3.0/255.255.255.0 ; tighten security a little bit [abc-gw] ; outbound connections to remote site type=peer secret=mysecret username=myusername host=1.2.3.4 disallow=all allow=ulaw Then in your dialplan, use something like this to call the remote site: exten => _6X,1,Dial(IAX2/myusername@abc-gw/${EXTEN}) and [from-site2] include => local-extns Note: I type the majority of the above from memory, so there are likely some syntax errors in it. But you should get the picture. Also, a couple of people on the list indicated the friends/user/peer code is now broken in cvs-head (as of the last couple of days), so if you're running current head, that too could be an issue.
> > -- Executing Dial("SIP/68-4ab6", > > "IAX2/ast33:pass at 192.168.1.130/08 at from-sip") in new stack > > Feb 11 14:16:58 WARNING[5653]: channel.c:1898 ast_request: No > > translator path exists for channel type IAX2 (native 0) to 4 > > Feb 11 14:16:58 NOTICE[5653]: app_dial.c:800 dial_exec: Unable to > > create channel of type 'IAX2' (cause 0) > > > > This is particularly confounding because I have all codecs disabled > > except ulaw (all over, sip devices included). Is it trying to do > > native bridging? No lo comprendo. > > > > Here's my info: > > > > ast551: 192.168.1.130 > > ast33: 192.168.42.130 > > Version: CVS-HEAD-11/03/04-14:59:37 (both boxes) > > > > IAX.CONF on ast551: > > [general] > > bindport=4569 > > notransfer=yes > > disallow=all > > allow=ulaw > > > > [ast33] > > type=friend > > auth=md5 > > secret=pass > > context=no-callwaiting > > host=192.168.42.130 > > qualify=yes > > trunk=yes > > disallow=all > > allow=ulaw > > > > > > IAX.CONF on ast33: > > [general] > > bindport=4569 > > disallow=all > > allow=ulaw > > > > [ast551] > > type=friend > > auth=md5 > > secret=pass > > context=no-callwaiting > > host=192.168.1.130 > > qualify=yes > > trunk=yes > > disallow=all > > allow=ulaw > > > > > > EXTENSIONS.CONF on ast33: > > [from-sip] > > exten => 68,1,Dial(SIP/68,20) > > exten => 68,2,Voicemail(u118) > > exten => 68,102,Voicemail(b118) > > exten => 68,103,Hangup > > > > exten => > > _[012345]X,1,Dial(IAX2/ast33:pass at 192.168.1.130/${EXTEN}@from-sip) > > > > [no-callwaiting] > > include => from-sip > > include => outgoing > > > > > > EXTENSIONS.CONF on ast551: > > [from-sip] > > exten => 19,1,SetGroup(${EXTEN}) > > exten => 19,2,CheckGroup(1) > > exten => 19,103,Goto(19b,1) > > exten => 19,3,Dial(SIP/19,20) > > exten => 19,4,Voicemail(u18) > > exten => 19,5,Hangup > > > > exten => _6X,1,Dial(IAX2/ast551:pass at > 192.168.42.130/${EXTEN}@from-sip) > > > > [no-callwaiting] > > include => from-sip > > include => outgoing > > > Looks like you're are getting caught with using "friends" instead of > peer > and user. > > Try something like this in iax.conf instead: > [abc-inc] ; inbound connections from remote site > type=user > secret=mysecret > context=from-site2 > disallow=all > allow=ulaw ; supports only ulaw > deny=0.0.0.0/0.0.0.0 > permit=1.2.3.0/255.255.255.0 ; tighten security a little bit > > [abc-gw] ; outbound connections to remote site > type=peer > secret=mysecret > username=myusername > host=1.2.3.4 > disallow=all > allow=ulaw > > Then in your dialplan, use something like this to call the remote site: > exten => _6X,1,Dial(IAX2/myusername at abc-gw/${EXTEN}) > > and > > [from-site2] > include => local-extns > > Note: I type the majority of the above from memory, so there are likely > some syntax errors in it. But you should get the picture.Hi Rich - I'm amazed you can type all that from memory! Thanks for taking the time to do so.> Also, a couple > of people on the list indicated the friends/user/peer code is now > broken in cvs-head (as of the last couple of days), so if you're > running current head, that too could be an issue.My CVS HEAD version is pretty old (11/04), so hopefully that's not the issue. I tried switching from type=friend over to type=user and type=peer. The results seem to be the same. I still get that error: -- Executing Dial("SIP/69-69ca", "IAX2/ast33-out/08@no-callwaiting") in new stack Feb 11 16:45:48 WARNING[5828]: channel.c:1898 ast_request: No translator path exists for channel type IAX2 (native 0) to 4 Feb 11 16:45:48 NOTICE[5828]: app_dial.c:800 dial_exec: Unable to create channel of type 'IAX2' (cause 0) The part that worries me is the (native 0). Why is it saying that IAX2 is native 0? With all the disallow=all and allow=ulaw, shouldn't it also be type 4? Thanks! Noah
Maybe Matching Threads
- Re: Codec Issue on IAX trunk? (Solved)
- ChanIsAvail for IAX not working again/still? AKA Redundant IAX connections not working
- host and vm on isolated network, there is ip (via dhcp) but not ping
- Set port to which Asterisk should send its answer
- pop3s Authentication Issues, Continued