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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