Unable to create channel of type 'sip'

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Unable to create channel of type 'sip'

Postby Michael_N » Thu Nov 19, 2009 3:58 pm

In asterisk CLI i got this error when i try to make a call :

Unable to create channel of type 'sip' (cause 3 - No route to destination)

Is there something wrong with the settings?

Settings are done in modify carrier webpage in vicidial.

My carrier setting is:

register => xxx:xxx@sip.viptel.dk

Account Entry:
[VIPTEL]
disallow=all
allow=alaw
type=friend
username=xxxx
secret=xxxx
dtmfmode=inband
qualify=1000
host=viptel.dk

GLOBAL STRING : VIPTEL = SIP/xxx:xxxx@sip.viptel.dk

DIALPLAN ENTRY
exten => _X!,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X!,2,Dial(sip/${EXTEN}@VIPTEL,55,o)
exten => _X!,3,Hangup

VERSION: 2.2.0-222
BUILD: 91031-1232
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Postby mflorell » Thu Nov 19, 2009 6:14 pm

You are going to need to post more Asterisk CLI output.

Also, have you ever been able to place a call over this trunk?

Did the carrier include any documentation for setting up their service?
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Postby Michael_N » Fri Nov 20, 2009 5:26 am

mflorell wrote:You are going to need to post more Asterisk CLI output.

Also, have you ever been able to place a call over this trunk?

Did the carrier include any documentation for setting up their service?


This is the cli output when calling number 25883655
from my softphone connected to my vici-server.

The provider only gave me username and password and sipserver

and no calls from asterisk with this trunk


[Nov 20 11:18:55] -- Remote UNIX connection
h-87-109*CLI> sip show registry
Host Username Refresh State Reg.Time
sip.viptel.dk:5060 46949733 22 Registered Fri, 20 Nov 2009 11:18:38
[Nov 20 11:19:01] -- ast_get_srv: SRV lookup for '_sip._udp.sip.viptel.dk' mapped to host sbc01.viptel.dk, port 5060
[Nov 20 11:19:01] == Parsing '/etc/asterisk/manager.conf': [Nov 20 11:19:01] Found
[Nov 20 11:19:01] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 20 11:19:01] == Parsing '/etc/asterisk/manager.conf': [Nov 20 11:19:01] Found
[Nov 20 11:19:01] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 20 11:19:02] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 20 11:19:07] == Parsing '/etc/asterisk/manager.conf': [Nov 20 11:19:07] Found
[Nov 20 11:19:07] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 20 11:19:07] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 20 11:19:07] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 20 11:19:11] -- Executing [25883655@default:1] AGI("SIP/cc101-008871b0", "agi://127.0.0.1:4577/call_log") in new stack
[Nov 20 11:19:11] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 20 11:19:11] -- Executing [25883655@default:2] Dial("SIP/cc101-008871b0", "sip/25883655@VIPTEL|55|o") in new stack
[Nov 20 11:19:11] WARNING[5109]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'sip' (cause 3 - No route to destination)
[Nov 20 11:19:11] == Everyone is busy/congested at this time (1:0/0/1)
[Nov 20 11:19:11] -- Executing [25883655@default:3] Hangup("SIP/cc101-008871b0", "") in new stack
[Nov 20 11:19:11] == Spawn extension (default, 25883655, 3) exited non-zero on 'SIP/cc101-008871b0'
[Nov 20 11:19:11] -- Executing [h@default:1] DeadAGI("SIP/cc101-008871b0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
[Nov 20 11:19:11] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
h-87-109*CLI>
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Postby mflorell » Fri Nov 20, 2009 6:18 am

I am not familiar with the dialing prefixes used over there in Denmark, but I would try different prefixes and see what goes through, maybe adding a 0 or the country code...

You could also contact the carrier and ask how they need to receive the dialstring for calls to go through.
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Postby Michael_N » Fri Nov 20, 2009 7:03 am

mflorell wrote:I am not familiar with the dialing prefixes used over there in Denmark, but I would try different prefixes and see what goes through, maybe adding a 0 or the country code...

You could also contact the carrier and ask how they need to receive the dialstring for calls to go through.


They dont use prefix when calling inside of denmark, a danish nummber is like XXXXXXXX
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Re: Unable to create channel of type 'sip'

Postby williamconley » Fri Nov 20, 2009 9:23 am

Michael_N wrote: register => xxx:xxx@sip.viptel.dk

Account Entry:
[VIPTEL]

GLOBAL STRING : VIPTEL = SIP/xxx:xxxx@sip.viptel.dk

DIALPLAN ENTRY
exten => _X!,2,Dial(sip/${EXTEN}@VIPTEL,55,o)



change global string to VIPTELTRUNK = SIP/VIPTEL

you are not dialing through your viptel sip account setup, you are bypassing it by defining everything in the global string.

also, i recommend not using _X! but rather "_8X." for your exten, which will allow you to put "8" in your dial prefix and have multiple carriers.

The reason for this is testing ... if you have a problem with your dialer, you can often save yourself hours of trouble by having a second carrier (_7X.) and just changing the dial prefix to see if the problem is related to your carrier.

so:

exten => _8X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _8X.,2,Dial(sip/${EXTEN:1}@VIPTEL,55,o)
exten => _8X.,3,Hangup

the "EXTEN:1" drops the "8" so it is not sent to your provider.
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