Unable to create channel of type 'SIP' no outboud dialing

[abhasbajpai]

I am having go autodial ce2.1

my carrier configuration is

register => 1000257:XXXXX@216.128.XX.XX

aabhas]
disallow=all
disallow=all
allow=ulaw
allow=alaw
allow=g729
type=friend
username=1000257
secret=9YsTHuXXXX
fromhost=216.128.XX.XX
host=dynamic
dtmfmode=rfc2833
context=trunkinbound

global string

TESTSIPTRUNK = SIP/aabhas

Dial plan

exten => _XXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXX.,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,tTor)
exten => _XXX.,3,Hangup

i am able to ping 216.128.XX.XX from the server

in sip show registery

216.128.XX.XX:5060 1000257 105 Registered
i am registered

when i dial

12127773456

[Sep 8 11:14:02] -- Executing [12127773456@default:1] AGI("SIP/8003-0000000 4", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 8 11:14:02] -- AGI Script agi://127.0.0.1:4577/call_log completed, ret urning 0
[Sep 8 11:14:02] -- Executing [12127773456@default:2] Dial("SIP/8003-000000 04", "SIP/aabhas/127773456||tTor") in new stack
[Sep 8 11:14:02] WARNING[12265]: app_dial.c:1310 dial_exec_full: Unable to crea te channel of type 'SIP' (cause 20 - Unknown)
[Sep 8 11:14:02] == Everyone is busy/congested at this time (1:0/0/1)
[Sep 8 11:14:02] -- Executing [12127773456@default:3] Hangup("SIP/8003-0000 0004", "") in new stack
[Sep 8 11:14:02] == Spawn extension (default, 12127773456, 3) exited non-zero on 'SIP/8003-00000004'
[Sep 8 11:14:02] -- Executing [h@default:1] DeadAGI("SIP/8003-00000004", "a gi://127.0.0.1:4577/call_log--HVcauses- ... NUNAVAIL-- --------") in new stack
i get this error, i am unable to do outbound or inbound calling
please help

[williamconley]

aabhas]

maybe you should have the other [?

[maykelsoft]

check also your dialer if successfully connected to your carrier.

try this command.

asterisk -rx "sip show peers"

[williamconley]

check also your dialer if successfully connected to your carrier.

try this command.

asterisk -rx "sip show peers"

registration shows successful connection. appearing in peers only shows connection for dynamic connections (if there is an ip, someone has successfully attached ...).

But if the connection has "host=xxx.xxx.xxx.xxx", it will appear in the sip show peers regardless. and he has host= ... so it will appear in peers even if the connection isn't possible.

if you also use "qualify=yes" (or a number), you will see the actual connection status (as in whether the round trip packets are outside the allowable time range) as opposed to the registration status (and he already has registered, so that is confirmed).

[maykelsoft]

thanks william,

i guess you're trying to dial a US movie phone number such as 2127773456...
[Sep 8 11:14:02] -- Executing [12127773456@default:2] Dial("SIP/8003-000000 04", "SIP/aabhas/127773456||tTor") in new stack

however, the first two digits of the number is missing when sending the call.. this might due to EXTEN:2 on your dialplan.

try this dialplan

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@aabhas,,tTor)
exten => _91XXXXXXXXXX,3,Hangup

[williamconley]

good eye. however: that would not stop asterisk from attempting to dial out through the provider. the provider would merely have canceled the call upon receipt.

also: if you use a dial plan with EXTEN:2 and 91 in front of it ... you have to modify your dial prefix in the campaign to "9" (so the 9 can be added, and then later stripped) and then verify that the carrier does not want the "1" (or change the EXTEN:2 to EXTEN:1 so it leaves the "1").

[abhasbajpai]

maybe you should have the other [?

i have put that, sorry for the typo here in posting
my conf is
[aabhas]
disallow=all
disallow=all
allow=ulaw
allow=alaw
allow=g729
type=friend
username=1000257
secret=9YsTHuXXXX
fromhost=216.128.XX.XX
host=dynamic
dtmfmode=rfc2833
context=trunkinbound

as for dialplan i tried with both 1 or 2, but its not going to the carrier, same cause 20 error

[maykelsoft]

if that the case, it is more on carrier issue... have you coordinated this error to your carrier as this is always happening to me everytime i changed carrier, they still not open my account... my 2 cents...

[williamconley]

host=dynamic

you cannot have a host without a host. dynamic is "i don't know the host, they will log in to us with a password and tell us their IP". that is not how hosts work, that's how PHONES work. You must have the ip or domain of your host in host=hostsite.com or host=xxx.xxx.xxx.xxx (ip address).

[scenarist]

I have the same problem like on the first post? This is my configuration of sip carrier


but these are result of commands sip show registry and sip show peers

Code: Select all

go*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
8016/8016                  192.168.1.100    D   N      5060     OK (9 ms)
kom2it/xxxx.at            67.215.65.132        N      5060     UNREACHABLE

Code: Select all

go*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time
sip.kom2it.at:5060              xxxx.at           105 Registered           Wed, 08 Feb 2012 21:31:53

and this is my sip set debug 67.215.65.132

Code: Select all

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2012.02.08 21:08:33 =~=~=~=~=~=~=~=~=~=~=~=

go*CLI> sip set debug ip 67.215.65.132

go*CLI>
SIP Debugging Enabled for IP: 67.215.65.132

go*CLI>
[Feb  8 21:10:25] Reliably Transmitting (NAT) to 67.215.65.132:5060:
OPTIONS sip:sip.kom2it.com;cpd=on SIP/2.0

Via: SIP/2.0/UDP 192.168.1.70:5060;branch=z9hG4bK4cea059d;rport

From: "asterisk" <sip:asterisk@192.168.1.70>;tag=as3e91fa1b

To: <sip:sip.kom2it.com;cpd=on>

Contact: <sip:asterisk@192.168.1.70>

Call-ID: 76a93fa00abaf56134db62df3caf817a@192.168.1.70

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 08 Feb 2012 20:10:25 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Content-Length: 0


and this is output of CLI when I try to make phone call

Code: Select all

[Feb  8 21:20:19]     -- Executing [938762723254@default:1] AGI("SIP/8019-0000000a", "agi://127.0.0.1:4577/call_log") in new stack
[Feb  8 21:20:19]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Feb  8 21:20:19]     -- Executing [938762723254@default:2] Dial("SIP/8019-0000000a", "SIP/38762723254@kom2it||tTo") in new stack
[Feb  8 21:20:19] WARNING[2124]: app_dial.c:1310 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Feb  8 21:20:19]   == Everyone is busy/congested at this time (1:0/0/1)
[Feb  8 21:20:19]     -- Executing [938762723254@default:3] Hangup("SIP/8019-0000000a", "") in new stack
[Feb  8 21:20:19]   == Spawn extension (default, 938762723254, 3) exited non-zero on 'SIP/8019-0000000a'
[Feb  8 21:20:19]     -- Executing [h@default:1] DeadAGI("SIP/8019-0000000a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Feb  8 21:20:19]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0

Any sugestions or advice I would be appreciate...[/code]

[scenarist]

Problem fixed!

instead of sip.kom2it.com must being

sip.kom2it.at

Everything is ok now.!