Hello Claudius,Henry is right about the dial plan. Here is what I found from
days of experminting with Callback and DISA and A2billing.Callback does not use
the trunk settings in Asterisk when it places a call out. Neither does
A2billing. However, DISA does. There has been many issues related to this and I
beleive there is a ticket opened for this on Asterisk as to wether this should
be incorporated or not.I got my callback and disa to work nicely after I fixed
the dialplan.If you usually dial out from your softphone with let's say :
416XXXXXXX and then your trunk prefixes 1 to the number then you will encounter
problem with Callback. But the good thing is that callback dose take context
values. For example, in Callback module for callback number you will insert
1${CALLERID(number)} This will make sure that 1 is prefixed to your
number when the call back dials out.If you are using DISA with this then there
is a different storey. For your trunks you can either add your prefix or simply
don't add anything and dial the complete number which might require a 1 such as
1416XXXXXXX.And here is the last tip. Do not use # at the end of destination
number if you are using DISA. This will simply not callout because most of voip
providers do not support # at the end of the number but it is very easy to slip
by your mind when you look at the log.I hope this helps you fix your
system.Regards,BruceP.S. was the verbose set to high when you copied the
log?Date: Wed, 28 Mar 2007 22:57:35 -0500From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: Re: [on-asterisk] RESEND : VoiceMail CallBack Issue
Hi my two cents worth says check the dial plan. The CID from the
initial call will never have a "1" in front of it
If your dial plan needs a "1" (even if its then dropped) to be routed
then that would cause the problem
AS an aside I used to "roll my own dial plans" using "9" + "1" for
long distance on a VoIP line but since a lot of phones keep
track of missed calls and other information as well as Call back and
Call blocking and ....
I have gone for the simplest plan which keeps everything the way it
would be on there phone at home.
The only downside is that the extension range starts at 200 and not
100 that way there is not conflict and all CID s
can be dialed as received.
LOCAL
416NXXXXXX
647NXXXXXX
905NXXXXXX
LONG DISTANCE
1|NXXNXXXXXX
INTERNATIONAL
011.
Henry
Bruce Nik wrote:
Claudius can you please post your CLI
log when you do the Callback/Disa combo or just the callback.
I think I have found one more bit of info to share.
Thanks,
Bruce
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [email protected]
Date: Wed, 28 Mar 2007 04:09:23 +0000
Subject: RE: [on-asterisk] RESEND : VoiceMail CallBack Issue
Thanks Bruce,
I found this problem also being reported on Digium forum and a "patch"
resolved it but I seem not to find the patch in question
http://bugs.digium.com/view.php?id=7961
Hopefully, you could shed some light on this one
Thanks,
Claudius
From: [EMAIL PROTECTED]
To: [email protected]
Date: Tue, 27 Mar 2007 19:08:45 -0400
Subject: RE: [on-asterisk] RESEND : VoiceMail CallBack Issue
Claudius,
I have actually had problems with Callback myself. Exact problem as
yours except for I was using the simple Callback module with DISA.
Apprently lots of people have problem with it, so everyone uses
callback.rb file which is written with Ruby. It's a different module. I
tried it but I didn't get that working either. However, I got the
actuall callback working after removal and re-install but again it
doesn't work with 416 area code and works with 905 area codes. So, I am
puzzled now :) When I get more time, I will work on it and shall post
it here.
To install ruby callback you can do these commands:
yum install ruby
to remove:
yum remove ruby
You have to also add certain lines in your amportal.conf file.
If you browse through trixbox.org you should see my posts on it.
Username: Bruce.
Plz let me know if you get it to work.
Regards,
Bruce
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [email protected]
Date: Tue, 27 Mar 2007 15:54:53 +0000
Subject: RE: [on-asterisk] RESEND : VoiceMail CallBack Issue
Thanks Alex.
Will do and report back to the list if a response is found
Thanks again and have a great one!
Claudius
Date: Tue, 27 Mar 2007
09:41:21 -0400
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [email protected]
Subject: RE: [on-asterisk] RESEND : VoiceMail CallBack Issue
Hi Claudius,
The problem
with an issue like this one is that the FreePBX team has written dozens
of custom macros and contexts to handle call routing within Asterisk.
Unless you get someone who really knows these contexts well, it’s hard
to get help. My suggestion would be to submit a bug to the FreePBX trac
system ( http://freepbx.org/trac/
) with the same level of detail as your original post. One of the devs
will then validate the issue and (hopefully) work on a fix for you.
Cheers,
Alex
___________________________________________
Alex
Robar,
Technical Support, GearyTech Inc.
3075
Fourteenth Avenue, Unit 3, Markham, Ontario L3R 0G9
Markham:
905-513-8000 x 223 Fax: 905-513-8040
Toronto:
416-226-3614 Toll Free: 888-890-3499
[EMAIL PROTECTED]
www.gearytech.com
Strategic
management of technology for business.
From: Claudius Fortis
[mailto:[EMAIL PROTECTED]
Sent: Tuesday,
March 27, 2007 9:22 AM
To:
[email protected]
Subject: RE:
[on-asterisk] RESEND : VoiceMail CallBack Issue
Morning All,
Thought I'd resend, hoping someone would point me to the right
direction...
Thanks,
Claudius
From:
[EMAIL PROTECTED]
To: [email protected]
Date: Sat, 24 Mar 2007 18:49:42 +0000
Subject: [on-asterisk] VoiceMail CallBack Issue
Good Day,
Hoping to get some pointers from here, I thought I'd shoot my question.
On my TrixBox at home, I've noticed the VoiceMail call back is no
longer working; i.e. when listening to messages, one can press 3 for
Advanced Options, then 2 to call back the caller.
This was working fine but I've noticed lately this feature not to be
responding, well, the call will just "silently" disconnect.
Here's the log I'm seeing on the console....
Any help or directions would be greatly appreciated!...I'm no expert
and this things is driving nuts already...
-- Playing
'/var/spool/asterisk/voicemail/default/62255/INBOX/msg0000' (language
'en')
-- Playing 'vm-advopts' (language 'en')
-- Playing 'vm-toreply' (language 'en')
-- Playing 'vm-tocallback' (language 'en')
-- Callback Requested
-- Confirm CID number '4165550123' is
number to use for callback
-- Playing 'vm-num-i-have' (language 'en')
-- Playing 'digits/4'
(language 'en')
-- Playing 'digits/1'
(language 'en')
-- Playing 'digits/6'
(language 'en')
-- Playing 'digits/5'
(language 'en')
-- Playing 'digits/5'
(language 'en')
-- Playing 'digits/5'
(language 'en')
-- Playing 'digits/0'
(language 'en')
-- Playing 'digits/1'
(language 'en')
-- Playing 'digits/2'
(language 'en')
-- Playing 'vm-tocallnum' (language 'en')
-- Playing 'vm-star-cancel' (language 'en')
-- Destination number is CID number '416555012H'
-- Placing outgoing call to extension '416555012H'
in context 'from-internal' from context 'from-internal'
-- Playing 'vm-dialout' (language 'en')
-- Executing Macro("SIP/62255-51ab", "hangupcall") in new stack
-- Executing ResetCDR("SIP/62255-51ab", "w") in new stack
-- Executing NoCDR("SIP/62255-51ab", "") in new stack
-- Executing Wait("SIP/62255-51ab", "5") in new stack
You would notice 2 things: First the CallerID of my caller was
416.555.0123 <10 digit> which is confirmed as above
But, asterisk, when confirming the number, only takes the first 9
digits, thus dropping the last digit, which is then replaced by an H.
From what I understand, the H causes a Hangup
My questions are then:
1. Why Asterisk is stripping the last digit ?
2. Why is it appending H at the end of the number to call ?
3. Where in Asterisk do I have to look to fix this problem?
This is Asterisk 1.2.9.1
Thanks everyone for your input and have a wonderful one!
Claudius
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