Can't make outgoing calls

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Can't make outgoing calls

Postby med_aatif » Mon Oct 19, 2009 11:11 am

Hi

I can't make an outbound call, I'm registred my trunk SIP

CLI> sip show registry

Host Username Refresh State

80.x.x.x:5060 testcarrier 105 Registered

this is my CLI

-- Executing AGI("SIP/100-088bfd38", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/100-088bfd38", "TRUNK/33516440377|55|To") in new stack
Oct 19 14:19:01 WARNING[9960]: channel.c:2621 ast_request: No channel type registered for 'TRUNK'
Oct 19 14:19:01 NOTICE[9960]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'TRUNK' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/100-088bfd38", "") in new stack
== Spawn extension (default, 0033516440377, 3) exited non-zero on 'SIP/100-088bfd38'
-- Executing DeadAGI("SIP/100-088bfd38", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack

any help please
med_aatif
 
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thanks

Postby brett05 » Mon Oct 19, 2009 12:26 pm

read the manual manager it will help you
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Re: Can't make outgoing calls

Postby williamconley » Mon Oct 19, 2009 7:15 pm

med_aatif wrote:TRUNK/33516440377|55|To
Your problem is that you have named your trunk "trunk" but you do not have a global variable named "trunk" to alter it to the SIP context. so it tries to dial using "trunk" which does not happen to be a carrier .. . and fails.

if you are using the "gui" in vicidial 2.0.5, your carrier settings have not been completed correctly. your global variable must match the name of the trunk in use as a variable.
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Postby med_aatif » Tue Oct 20, 2009 3:55 am

I'm still having problem

This is my carrier setting :

Carrier Name: TRUNK

Registration String: register => carrier:test@80.x.x.x:5060

Template ID: NONE

Account Entry:

[carrier]
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=g723.1
type=friend
context=inbound
username=carrier
secret=test
host=80.x.x.x
dtmfmode=rfc2833

Protocol: SIP

Globals String: TRUNK = SIP/carrier

Dialplan Entry:

exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(${TRUNK}/${EXTEN},105,tTor)
exten => _X.,3,Hangup

Server IP: 10.1.6.253

Active: Y
med_aatif
 
Posts: 11
Joined: Fri Sep 25, 2009 2:16 pm

thanks

Postby brett05 » Tue Oct 20, 2009 6:06 am

you have disallow=all all you codec then you have all them all ?
strange ?
do you have codec g729?
do you have codec g723.1?
what you see if you type "sip show channel"?
here you need to see the out use of codec from your provider and your asterisk box ?
why you all only phones 105 to make dialplan ?
set it to all phones ?
what is your asterisk version ?
what is your vicidial version?
what is your OS version?
and try this :

Registration String: register => carrier:test@80.x.x.x:5060

Template ID: NONE

Account Entry:

[carrier]
disallow=all
allow=alaw
allow=ulaw
type=friend
context=inbound
username=carrier
secret=test
host=80.x.x.x
dtmfmode=rfc2833

Protocol: SIP

Globals String: TRUNK=SIP/carrier"no space here "

Dialplan Entry:

exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(${TRUNK}/${EXTEN:1},,tTor)
exten => _X.,3,Hangup
#EXTEN:1 or EXTEN:2 or other this depand in your outgoing call number
you can read more the file extensions.conf in etc/asterisk to understand more the use of the dialplan

i hope i have help you with my poor answer :D thanks
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Location: tunisia

Postby med_aatif » Tue Oct 20, 2009 8:36 am

Thanks brett05 and williamconley

do you have codec g729?
do you have codec g723.1?


I can use codecs g729 and g723.1 with my provider

what you see if you type "sip show channel"?

Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
10.0.5.104 cc101 MDRhMDY3ODk 00101/00002 unkn No Rx: ACK
80.x.x.x carrier 4267ca607ac 00114/00000 unkn No
2 active SIP channels

why you all only phones 105 to make dialplan ?
set it to all phones ?


I don't know how

what is your asterisk version ?

V 1.4

what is your vicidial version?

1.2 (Vicidialnow)

any help will be appriciate
med_aatif
 
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Joined: Fri Sep 25, 2009 2:16 pm

Postby williamconley » Tue Oct 20, 2009 10:07 am

vicidialnow version is not vicidial version. your vicidial version is at the bottom of most screens in the gui, include the build information as it can be essential for future troubleshooting.

if your carrier name is trunk, why is the context named [carrier]?

you should try using the original sip sample and not change the dialplan entry or the global variable entry, or the name of the sip context ... just change the host, user & secret to your settings.

leave everything else alone (except for the minor bug in some systems which requires deleting the 91NXXNXXXXXX dial pattern from extensions.conf because it breaks vicidials auto-generated conf file, which attempts to use the same pattern by default and loses).

alternately you can use dial 8 service instead by changing the 9 to an 8 in the sample dial plan.
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Postby manickam » Thu Oct 22, 2009 10:25 am

hi here is your error
Carrier Name: TRUNK this should be carrier

and verify that your carrier id is named carrier

if it's named something else like blabla then in your global strings put
TRUNK=sip/blabla
:)
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Re: Can't make outgoing calls

Postby THUFIR » Tue Mar 17, 2015 6:58 pm

williamconley wrote:
med_aatif wrote:TRUNK/33516440377|55|To
Your problem is that you have named your trunk "trunk" but you do not have a global variable named "trunk" to alter it to the SIP context. so it tries to dial using "trunk" which does not happen to be a carrier .. . and fails.

if you are using the "gui" in vicidial 2.0.5, your carrier settings have not been completed correctly. your global variable must match the name of the trunk in use as a variable.


in SIP "trunking", the asterisk/vici box is a client to a SIP proxy? The proxy being the carrier who provides the termination?

see also:
sip show registry: Show SIP registration status (when Asterisk registers as a client to a SIP Proxy)


thanks,

Thufir
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Re: Can't make outgoing calls

Postby williamconley » Tue Mar 17, 2015 7:11 pm

THUFIR wrote:in SIP "trunking", the asterisk/vici box is a client to a SIP proxy? The proxy being the carrier who provides the termination?


It's a semantic issue in some cases. But, yes. Some carriers handle it this way or at least "call it" this.

Consider it this way: You are not calling the telco directly. You are using the Carrier as a proxy to get to the telco.

In other cases, there's actually a method referred to as proxy that is a bit complex (and must actually be invoked).

X-Lite (A free, widely used soft phone) configuration has similar terminology whereby your outbound calls can be direct through a domain or a proxy.
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