[Asterisk-Users] Error in placing call file in directory
Chris Blake
chris at pixcel.co.za
Thu Mar 17 01:18:02 MST 2005
On Wed, 2005-03-16 at 16:20, Razza wrote:
> Chris Blake wrote :
>
> -----%<-----
> >If anyone can help I`ll send the call file to you, or is it ok to
> clutter the list with it ?
> -----%<-----
>
> 'Clutter' the list I'd be interested and at least it is pertinent to *
> ;o)
>
Howdy Razza and Stefan, thanks for replying....
Here is my call file
=======================
# This is a sample file that can be dumped in
#/var/spool/asterisk/outgoing
# to generate a call. Obviously, you MUST specify at least a channel in
# the same format as you would for the "Dial" application. Only one
# channel name is permitted.
Channel: Zap/g4/0117265559
# You may also specify a wait time (default is 45 seconds) for how long
# to wait for the channel to be answered, a retry time (default is 5
# mins) for how soon to retry this call, and a maximum number of retries
# (default is 0) for how many times to retry this call.
MaxRetries: 0
RetryTime: 60
WaitTime: 15
# Once the call is answered, you must provide either an application/data
# combination, or a context/extension/priority in which to start the
# PBX.
Context: ext-local
Extension: 200
Priority: 1
=========================
>From what I have read so far, this should work, but obviously something
is wrong somewhere. I understand the following, correct me if I am wrong
:
In the above example, "Context: ext-local" :
* will look in extensions.conf for this context, but I do not have it
specified there. This context is specified in
extensions_additional.conf.
However, I do have the "#include extensions_additional.conf" in
extensions.conf, so it should pick it up right ?
In any event, I have referenced another context which DOES exist in
extensions.conf but I still get the same result.
Here is how this context is specified in extensions_additional.conf :
[ext-local]
exten => 200,1,Macro(exten-vm,200,200)
exten => 201,1,Macro(exten-vm,201,201)
exten => 202,1,Macro(exten-vm,chris at pixcel.co.za,202)
Also, I notice that although the call is not being made, and I have
specified 0 retries in the call file, my log file keeps getting
cluttered with these entries following me setting "sip debug" on via *`s
CLI :
===========================================
=====Sip read:
OPTIONS sip:192.168.204.95 SIP/2.0
Content-Length: 0
Call-ID: ED3D2AE8-1DD1-11B2-AC8B-860A22D62D71 at 192.168.204.10
From: <sip:202 at 192.168.204.95>;tag=29451415291147145415
CSeq: 3895 OPTIONS
Max-Forwards: 70
To: <sip:192.168.204.95>
Via: SIP/2.0/UDP
192.168.204.10;rport;branch=z9hG4bKc0a8cc0a0131c9b142393a2f0af7ccce00003064
8 headers, 0 lines
Looking for 192.168.204.95 in from-sip-external
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.204.10;branch=z9hG4bKc0a8cc0a0131c9b142393a2f0af7ccce00003064
From: <sip:202 at 192.168.204.95>;tag=29451415291147145415
To: <sip:192.168.204.95>;tag=as2d027310
Call-ID: ED3D2AE8-1DD1-11B2-AC8B-860A22D62D71 at 192.168.204.10
CSeq: 3895 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:192.168.204.95>
Accept: application/sdp
Content-Length: 0
to 192.168.204.10:5060
Destroying call 'ED3D2AE8-1DD1-11B2-AC8B-860A22D62D71 at 192.168.204.10'
=================================================
So it appears that it`s destroying the call, and then retrying it
again...there are no other active lines in use on this box yet.
My permissions on /var/spool/asterisk/outgoing are as follows :
drwx------ 2 asterisk asterisk 4096 Mar 17 11:47 outgoing
Still searching google, wiki archives but nothing found yet....
Any ideas...?
--
Chris Blake
Cell: 082 775 1492
Work: +27 11 782 0840
Fax : +27 11 782 0841
Mail: chris at pixcel.co.za
There is no sin but ignorance. -- Christopher Marlowe
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