Followed your advice and here's an interesting fact I've noticed. Let me give some background fact firstHere's a typical VM config for my extensions4518 => 1234,Claudius Fortis,[EMAIL PROTECTED],[EMAIL PROTECTED],review=yes|callback=from-internal|attach=yes|saycid=yes|envelope=yes|delete=noAs you'd notice, mny VM callback context is from-internal. The same one used for my regular outbound callsSo, I then created a new Call-Out context and update the VM config with it [custom-VMcallback]exten => _X.,1,Dial(Local/[EMAIL PROTECTED],90) Result: ====Now I can see the call being made. I can see the call being sent thru [EMAIL PROTECTED] And since the famous H is still added, my call will then fail at Provider level.... I'm still going to continue to dig into the issue and hopefully will reach a resolution .... Thanks again for your input and have a great one! Claudius
From: [EMAIL PROTECTED]: [email protected]; [EMAIL PROTECTED]: Thu, 29 Mar 2007 23:47:59 -0400Subject: RE: [on-asterisk] RESEND : VoiceMail CallBack Issue 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 IssueHi 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 scan be dialed as received.LOCAL 416NXXXXXX647NXXXXXX905NXXXXXXLONG DISTANCE1|NXXNXXXXXX INTERNATIONAL011.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]: [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 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 Learn more! Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! Explore the seven wonders of the world Learn more! Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! 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