Help needed with extensions

[gkandylakis]

I'm trying to setup GoAutoDial 3.3 and i can't get the extensions to work properly. Here is my carrier setup:

[Voipstunt-SIP]
disallow=all
type=friend
dtmfmode=rfc2833
context=trunkinbound
qualify=1000
canreinvite=no
nat=yes
host=sip. voipstunt. com
allow=alaw
username=*******
fromuser=*******
secret=******
fromdomain=sip. voipstunt. com
insecure=port,invite

and dial plan:

exten => _00XXXXXXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _00XXXXXXXXXXXX.,2,Dial(SIP/${EXTEN:14}@Voipstunt-SIP,,tTo)
exten => _00XXXXXXXXXXXX.,3,Hangup

when i try to manual dial the test number 6969696969 from agent interface i get:

[Aug 9 06:32:18] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000012;2", "8600051,F") in new stack
[Aug 9 06:32:18] > Channel Local/8600051@default-00000012;1 was answered.
[Aug 9 06:32:18] == Starting Local/8600051@default-00000012;1 at default,00306969696969,1 failed so falling back to exten 's'
[Aug 9 06:32:18] == Starting Local/8600051@default-00000012;1 at default,s,1 still failed so falling back to context 'default'
[Aug 9 06:32:18] -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-00000012;1
[Aug 9 06:32:18] -- Executing [i@default:1] Playback("Local/8600051@default-00000012;1", "invalid") in new stack
[Aug 9 06:32:18] -- <Local/8600051@default-00000012;1> Playing 'invalid.gsm' (language 'en')
[Aug 9 06:32:18] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 9 06:32:22] -- Executing [i@default:2] Hangup("Local/8600051@default-00000012;1", "") in new stack
[Aug 9 06:32:22] == Spawn extension (default, i, 2) exited non-zero on 'Local/8600051@default-00000012;1'
[Aug 9 06:32:22] -- Executing [h@default:1] AGI("Local/8600051@default-00000012;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Aug 9 06:32:22] -- <Local/8600051@default-00000012;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Aug 9 06:32:22] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000012;2'
[Aug 9 06:32:22] -- Executing [h@default:1] AGI("Local/8600051@default-00000012;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Aug 9 06:32:22] -- <Local/8600051@default-00000012;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

Carrier needs 0030+phone to make a successfull call. If i try to make call just from the registered SIP phone i get :

[Aug 9 06:36:31] NOTICE[4039]: chan_sip.c:23534 handle_request_invite: Call from '8001' (192.168.1.199:5062) to extension '00306969696969' rejected because extension not found in context 'default'.

Any ideas?

[gkandylakis]

I got this working by changing the dialplan to:

exten => _00XXXXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _00XXXXXXXXXX.,2,Dial(${Voipstunt-SIP}/${EXTEN},60,tTo)
exten => _00XXXXXXXXXX.,3,Hangup

the new error now is this:

[Aug 9 12:10:34] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-0000000b;2", "8600051,F") in new stack
[Aug 9 12:10:34] > Channel Local/8600051@default-0000000b;1 was answered.
[Aug 9 12:10:34] -- Executing [00306969696969@default:1] AGI("Local/8600051@default-0000000b;1", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 9 12:10:34] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=61662862))
[Aug 9 12:10:34] -- <Local/8600051@default-0000000b;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 9 12:10:34] -- Executing [00306969696969@default:2] Dial("Local/8600051@default-0000000b;1", "/00306969696969,60,tTo") in new stack
[Aug 9 12:10:34] WARNING[13100]: channel.c:5711 ast_request: No channel type registered for ''
[Aug 9 12:10:34] WARNING[13100]: app_dial.c:2345 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
[Aug 9 12:10:34] == Everyone is busy/congested at this time (1:0/0/1)
[Aug 9 12:10:34] -- Executing [00306969696969@default:3] Hangup("Local/8600051@default-0000000b;1", "") in new stack
[Aug 9 12:10:34] == Spawn extension (default, 00306969696969, 3) exited non-zero on 'Local/8600051@default-0000000b;1'
[Aug 9 12:10:34] -- Executing [h@default:1] AGI("Local/8600051@default-0000000b;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack
[Aug 9 12:10:34] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 9 12:10:34] -- <Local/8600051@default-0000000b;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Aug 9 12:10:34] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-0000000b;2'
[Aug 9 12:10:34] -- Executing [h@default:1] AGI("Local/8600051@default-0000000b;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Aug 9 12:10:34] -- <Local/8600051@default-0000000b;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[williamconley]

exten => _00XXXXXXXXXX.,2,Dial(${Voipstunt-SIP}/${EXTEN},60,tTo)

Error: You are trying to send the call to a sip account ... but you forgot to mention "SIP". you just put the sip account name in. How would Asterisk know this is not IAX or ZAP? In short: You must put the channel type before the account.

Change to SIP/${Voipstunt-SIP}

Best practice is to create a global variable names VOIPSTUNT (or PRIMARY ... or something similar) and define that as SIP/Voipstunt-SIP ... but that's not a requirement. Mostly you just need to remember that you can not send to an account without specifying the protocol first.

Note that global variables can be used in other places and then changing the global variable changes it everywhere. So when you set up failover and other interesting things, this could be useful. But for now ... add SIP/ and you'll be good.

[gkandylakis]

Thanks for the advice! Worked like a charm! Now i'm getting this:

[Aug 10 06:50:06] == Using SIP RTP CoS mark 5
[Aug 10 06:50:06] -- Executing [00306969696969@default:1] AGI("SIP/8001-00000014", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 10 06:50:06] -- <SIP/8001-00000014>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 10 06:50:06] -- Executing [00306969696969@default:2] Dial("SIP/8001-00000014", "sip//00306969696969,60,tTo") in new stack
[Aug 10 06:50:06] WARNING[9213]: chan_sip.c:5823 create_addr: Purely numeric hostname (), and not a peer--rejecting!
[Aug 10 06:50:06] WARNING[9213]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Subscriber absent)
[Aug 10 06:50:06] == Everyone is busy/congested at this time (1:0/0/1)
[Aug 10 06:50:06] -- Executing [00306969696969@default:3] Hangup("SIP/8001-00000014", "") in new stack
[Aug 10 06:50:06] == Spawn extension (default, 00306969696969, 3) exited non-zero on 'SIP/8001-00000014'
[Aug 10 06:50:06] -- Executing [h@default:1] AGI("SIP/8001-00000014", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Aug 10 06:50:06] -- <SIP/8001-00000014>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0

My new config is:

register => username:password@77.72.169.134:5060/username

[Voipstunt-SIP]
disallow=all
type=peer
dtmfmode=rfc2833
context=trunkinbound
qualify=1000
canreinvite=no
nat=yes
host=77.72.169.134
allow=alaw
username=username
fromuser=username
secret=password
fromdomain=sip. voipstunt. com
insecure=port,invite
allow=gsm
allow=ulaw
allow=g729

exten => _00XXXXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _00XXXXXXXXXX.,2,Dial(sip/${Voipstunt-SIP}/${EXTEN},60,tTo)
exten => _00XXXXXXXXXX.,3,Hangup


sip show peers gives me this:

8001/8001 192.168.1.199 D N 5062 OK (4 ms)
Voipstunt-SIP/username 77.72.169.134 N 5060 OK (110 ms)

[striker]

change the dialplan

exten => _00XXXXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _00XXXXXXXXXX.,2,Dial(sip/Voipstunt-SIP/${EXTEN},60,tTo)
exten => _00XXXXXXXXXX.,3,Hangup

[gkandylakis]

Worked perfectly! You are a life saver!

[urban]

HI i have a Problem
I think that the dial plan is incorrect
Here are the infos

register => username:password@HostIp

[Routes]
type=peer

user=username
secret=**********
host=ip of host
qualify=yes
allow=g729

In Asterisk Cli i see that the carrier is registred
My dial Plan is this

exten => _XXXXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXX.,2,Dial(SIP/SIPRoutes/${EXTEN},60,tTo)
exten => _XXXXXXXXXX.,3,Hangup

In campaing Dial Prefix i set To X
In Xlite im trying to call Swiss Numbers like this 4131 11 22 33 4
From Where im calling i must use 11 digit

can anyone help me

[xenia2608]

Registration string: register => username:Password@ip/siphostname:5060

==Account entry==
[voip]
disallow=all
allow=g729
allow=gsm
allow=ulaw
type=friend
fromuser=username
username=username
secret=password
host=ip/sip host
dtmfmode=rfc2833
context=trunkinbound
trustrpid=yes
sendtrpid=yes
trunk=yes
qualify=yes
insecure=invite,port -------{for asterisk 1.8}
nat=yes

Global string: VOIPTRUNK=sip/voip

DialPlan Entry:

exten => _97.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _97.,2,Dial(${VOIPTRUNK}/${EXTEN:2},,tTo)
exten => _97.,3,Hangup

you need to set campaign dialing prefix to 97 or whatever you have provided instead of 97. and just change your campaign setting to omit prefix to yes.
The above mentioned carrier setting always worked for me.

[urban]

8001/8001 172.16.9.125 D N 63050 OK (4 ms)
Routes 77.72.174.129 N 5060 OK (40 ms)
22 sip peers [Monitored: 2 online, 20 offline Unmonitored: 0 online, 0 offline]
[Nov 20 12:02:03] NOTICE[2361]: chan_sip.c:15566 handle_request_invite: Call from '8001' to extension '9737744733533' rejected because extension not found.


this is the Cli with you Dialplan

what is wrong here

[xenia2608]

put 97+country code as prefix in campaign prefix and manual prefix and set yes to section omit prefix in campaign setting. and for manual use as 97+country code+phone number.

[urban]

hi xenia2608
as i have explained before

I call Swiss number like this using Elastix , I x-lite i use extension 100 , 4144 78 97 89 7 { Example}

So when im trying to call manual from xlite with goautodial it gives me error

campaing dial prefix 97
manual prefix 97

i want to make all calls with diferents prefixess

Like 41 for swiss
377 for monacco
and others

can you give me an example for that ?

[xenia2608]

just follow this
for manual calls from xlite or anysoftphone: 97+41 (country code for swiss)+ phone number and it should work.
as for usa/canada: 97+1+phone numbe
for monacco: 97+377+phone number

[urban]

gi i got this in Cli

[Nov 20 16:01:22] == Using SIP RTP CoS mark 5
[Nov 20 16:01:22] -- Executing [9737744733533@default:1] AGI("SIP/8001-00000002", "agi://127.0.0.1:4577/call_log") in new stack
[Nov 20 16:01:22] -- <SIP/8001-00000002>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 20 16:01:22] -- Executing [9737744733533@default:2] Dial("SIP/8001-00000002", "sip/voip/37744733533,,tTo") in new stack
[Nov 20 16:01:22] == Using SIP RTP CoS mark 5
[Nov 20 16:01:22] ERROR[9858]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("voip", "(null)", ...): Name or service not known
[Nov 20 16:01:22] WARNING[9858]: chan_sip.c:5865 create_addr: No such host: voip
[Nov 20 16:01:22] WARNING[9858]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Subscriber absent)
[Nov 20 16:01:22] == Everyone is busy/congested at this time (1:0/0/1)
[Nov 20 16:01:22] -- Executing [9737744733533@default:3] Hangup("SIP/8001-00000002", "") in new stack
[Nov 20 16:01:22] == Spawn extension (default, 9737744733533, 3) exited non-zero on 'SIP/8001-00000002'
[Nov 20 16:01:22] -- Executing [h@default:1] AGI("SIP/8001-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Nov 20 16:01:22] -- <SIP/8001-00000002>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[xenia2608]

from your cli
[Nov 20 16:01:22] WARNING[9858]: chan_sip.c:5865 create_addr: No such host: voip
resolve: getaddrinfo("voip", "(null)", ...): Name or service not known
please check your voip provider ip or host name they have given to you.