Thanks Bruce..... here's the complete CLI Log ....It actually looks more messed
up than the one I sent out in my original post .......this time 7 digits are
stripped ...Let me know if you need me to reproduce the issue in a specific
way....
NB. The phone number is modified....
Thanks,
Claudius
=====================================
=========================================================================Connected
to Asterisk 1.2.9.1 svn rev 34876 currently running on asterisk1 (pid=
2600)Verbosity is at least 10Core debug is at least 10 -- Executing
Answer("SIP/4518-fbe7", "") in new stack -- Executing Wait("SIP/4518-fbe7",
"1") in new stack -- Executing Macro("SIP/4518-fbe7", "user-callerid|") in
new stack -- Executing GotoIf("SIP/4518-fbe7", "0?report") in new stack
-- Executing GotoIf("SIP/4518-fbe7", "0?start") in new stack -- Executing
Set("SIP/4518-fbe7", "REALCALLERIDNUM=4518") in new stack -- Executing
NoOp("SIP/4518-fbe7", "REALCALLERIDNUM is 4518") in new stack -- Executing
Set("SIP/4518-fbe7", "AMPUSER=4518") in new stack -- Executing
Set("SIP/4518-fbe7", "AMPUSERCIDNAME=Claudius Fortis") in new stack --
Executing GotoIf("SIP/4518-fbe7", "0?report") in new stack -- Executing
Set("SIP/4518-fbe7", "CALLERID(all)=Claudius Fortis <4518>") in new stack --
Executing NoOp("SIP/4518-fbe7", "Using CallerID "Claudius Fortis" <4518>") in
new stack -- Executing Macro("SIP/4518-fbe7", "get-vmcontext|4518") in new
stack -- Executing Set("SIP/4518-fbe7", "VMCONTEXT=default") in new stack
-- Executing GotoIf("SIP/4518-fbe7", "0?200:300") in new stack -- Goto
(macro-get-vmcontext,s,300) -- Executing NoOp("SIP/4518-fbe7", "") in new
stack -- Executing VoiceMailMain("SIP/4518-fbe7", "[EMAIL PROTECTED]") in
new stack -- Playing 'vm-password' (language 'en') -- Playing
'vm-youhave' (language 'en') -- Playing 'digits/17' (language 'en') --
Playing 'vm-Old' (language 'en') -- Playing 'vm-messages' (language 'en')
-- Playing 'vm-onefor' (language 'en') -- Playing 'vm-first' (language 'en')
-- Playing 'vm-message' (language 'en') -- Playing 'vm-received'
(language 'en') -- Playing 'digits/day-3' (language 'en') -- Playing
'digits/mon-10' (language 'en') -- Playing 'digits/20' (language 'en') --
Playing 'digits/h-9' (language 'en') -- Playing 'digits/2' (language 'en')
-- Playing 'digits/thousand' (language 'en') -- Playing 'digits/6'
(language 'en') -- Playing 'digits/at' (language 'en') -- Playing
'digits/6' (language 'en') -- Playing 'digits/30' (language 'en') --
Playing 'digits/3' (language 'en') -- Playing 'digits/p-m' (language 'en')
-- Playing 'vm-from-phonenumber' (language 'en') -- Playing 'digits/6'
(language 'en') -- Playing 'digits/0' (language 'en') -- Playing
'digits/4' (language 'en') -- Playing 'digits/9' (language 'en') --
Playing 'digits/1' (language 'en') -- Playing 'digits/1' (language 'en')
-- Playing 'digits/0' (language 'en') -- Playing 'digits/1' (language 'en')
-- Playing 'digits/2' (language 'en') -- Playing 'digits/3' (language
'en') -- Playing '/var/spool/asterisk/voicemail/default/4518/Old/msg0000'
(language 'en') -- Playing 'vm-advopts' (language 'en') -- Playing
'vm-toreply' (language 'en') -- Playing 'vm-tocallback' (language 'en')
-- Playing 'vm-tohearenv' (language 'en') -- Callback Requested --
Confirm CID number '6049110123' is number to use for callback -- Playing
'vm-num-i-have' (language 'en') -- Playing 'digits/6' (language 'en') --
Playing 'digits/9' (language 'en') -- Playing 'digits/1' (language 'en')
-- Playing 'vm-tocallnum' (language 'en') -- Playing 'vm-star-cancel'
(language 'en') -- Destination number is CID number '-40' -- Placing
outgoing call to extension '-40' in context 'from-internal' from context
'from-internal' -- Playing 'vm-dialout' (language 'en') -- Executing
Macro("SIP/4518-fbe7", "hangupcall") in new stack -- Executing
ResetCDR("SIP/4518-fbe7", "w") in new stack -- Executing
NoCDR("SIP/4518-fbe7", "") in new stack -- Executing Wait("SIP/4518-fbe7",
"5") in new stack -- Executing Hangup("SIP/4518-fbe7", "") in new stack
asterisk1*CLI>
======================================
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Thu, 29 Mar 2007 21:13:49
-0400Subject: RE: [on-asterisk] RESEND : VoiceMail CallBack Issue
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]: [EMAIL PROTECTED]; [EMAIL PROTECTED]: Wed, 28 Mar 2007
04:09:23 +0000Subject: 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=7961Hopefully, you could shed some light on
this one Thanks, Claudius
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 27 Mar 2007 19:08:45
-0400Subject: 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 rubyto remove:yum remove rubyYou 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]: [EMAIL PROTECTED]; [EMAIL PROTECTED]: Tue, 27 Mar 2007
15:54:53 +0000Subject: 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 -0400From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]; [EMAIL PROTECTED]: 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 AMTo: [EMAIL PROTECTED]: 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]: [EMAIL PROTECTED]: Sat, 24 Mar 2007 18:49:42
+0000Subject: [on-asterisk] VoiceMail CallBack IssueGood 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 aboveBut, 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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