Getting problems in configuring call files using sip

Asterisk Asterisk Support
1

i have problems in configuring call files by using sip . i want to first dial out of asterisk pbx to an service provider and then want to dial to one of my extentions, defined in extensentions.conf

by using an sample file of call files its giving error saying that call could not be connected

thanks

2

More information is needed. A copy of the call file you are trying to use, and output from a verbose CLI.

Have you tried following the information here as well?:

voip-info.org/tiki-index.php … o-dial+out

3

yes i have tried out the instructions ion the link

I am facing a problem in auto diailing through call files. When i try to
dial having this in my test.call ,

Channel: SIP/914084704700@sip_proxy-out
Callerid: 3301
MaxRetries: 2
RetryTime: 60
WaitTime: 30

Assuming that your local extensions are kept in the

context called [extensions]

Context: test_in
Extension: 1235
Priority: 1

I receive :

*CLI> – Attempting call on SIP/914084704700@sip_proxy-out for
1235@test_in:1 (Retry 1)
> Channel SIP/sip_proxy-out-1ebf was never answered.
Feb 1 10:06:21 WARNING[6019]: cdr.c:548 ast_cdr_disposition: Cause not
handled
Feb 1 10:06:21 NOTICE[6019]: pbx_spool.c:266 attempt_thread: Call failed to
go through, reason 8

even when i can dial out manually through the same context(sip_proxy-out)
in sip.conf.

pl help me out of this problem.

4

Are you saying you are not able to dial out via that SIP proxy via call files or via a SIP endpoint on the Asterisk server? If this is the case, your call file is not the problem and your SIP proxy configuration for that service is. From the information provided it is not possible to determine if you are configured for that SIP proxy properly.

5

we are able to dial through sip exten using the same proxy and same context but

when we try to dial out automatically using call files; messages as descibed in previous mail outcomes.

thanks

6

hi

waiting for yr help

7

This is file in directory /var/spoll/asterisk/outgoing
channel: SIP/00359877054596@musimi
maxretries: 3
retrytime: 60
waittime: 60
callerid: 00359877054596
application: DeadAGI
data: wakeSMS.agi
account:

in sip.conf:

[musimi]
type=friend
secret=password
username=name
fromuser=name

fromdomain=musimi.dk
host=musimi.dk

disallow=all
allow=alaw
canreinvite=yes
context=default
dtmfmode=rfc2833
insecure=very
nat=yes

This way I am making callback, then I can dial number I wish to dodial

8

thank u . i will try