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
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