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,BruceFrom: [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 Get news, entertainment and everything you care about at Live.com. Check it out! Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it!ExchangeDefender Message Security: Check Authenticity Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE
