Following on from an original post it appears that to reduce issues with call monitoring it is worth trying to use the same codec as your outgoing/incoming trunk.
Therefore our sip provider uses G729 and respectively we on said trunk only allow=G729
In general settings I have set the Call Recording Format to G729.
Indeed it is generating G729 recorded files, except for now we have no way of listening to them!
I can see them in the User Portal but clicking on the play button doesnt work, as the Quicktime player wont support the codec, nor does clicking the download button work as for some reason you get a 404 File Not Found.
I can Webmin in and pull the file but again its in .G729 format and after extensive googling I’m getting no joy on any play back ability?
I’m having the same problem. I installed FreePBX using the AsteriskNow bundle. All of the extensions are able to listen to/delete voice mail by dialing *98 on their extension, and emails with a copy of the voice mail attached are being sent correctly. The problem is using the user portal; the voice mails appear in the user portal, but no matter what browser I use (IE 8, Chrome 4, Safari 4, Firefox 3.5; all on Windows XP), I have the following problems:
When I click on the play button, a new blank line appears underneath the voicemail without any options to actually play the message back.
When I click on the download button, I’m redirected to a “404 File not found!” page.
When I select 1 or more voicemails and then click the delete button, the page refreshes and the voicemail messages are still there.
Have you checked the AsteriskNOW forum? There are two issues being presented here. The first (and purpose of the original thread) is how to play g729 files of which I suspect there is not going to be a good solution without some transcoding being done through Asterisk (if it has a license) which is an involved process.
The second is just playing the sound files. FreePBX works if the browsers have the proper plugins (various media players, common). So the issue is either a browser issue or the AsteriskNOW distribution has something configured wrong for FreePBX and if this is the case there may be others using AsteriskNOW running into the same problem on their forum possibly.
My best guess, if a configuration error, is it is related to permissions. In the past AsteriskNOW tried to run httpd as user apache but did not have the correct settings to do that such that there were permission problems. I haven’t followed to see if they are still doing that but I think they are. The simplest fix would be to change your httpd configuration to run apache as user asterisk, and make sure to do an ‘amportal restart’ afterwards so that our scrip fixes all the permissions. Then see if the problem is still there.
The original problem I was having was misaligned out of sync audio monitoring (recording) where audio lagged from the outside party (into our PBX or out) if we either rang them or they rang us audio was recorded but not in sync, the outside party not on our system started to lag behind, and the longer the duration of the call the longer the lag.
So I did some reading up further and see explanations of if your using VM then you may have issues, we are not, the machine is fairly well powered and solely running as asterisk/FreePBX server only.
I read in the FreePBX 2.5 guide (PackT Pub) that issues of lag can be because of difference in monitored recording output to actual voice codec/format.
Well we use open-source (non North Amer) G729 codecs, and allow that only respectively in our trunk setup as our SIP supplier also supports it and the recorded format by FreePBX was output to .wav files, so I thought marry the two up.
i.e. We use the G729 codec and in the FreePBX monitor format option I also selected G729, as it is an option there.
Trouble is the user portal uses quicktime and as a result you cant internally in the webpage listen to the recording like you could a .wav file as quicktime dont support the format, what bewilders me is why FreePBX has an option for G729 monitoring when there seems no possible way to review it if ever required which defeats the object of call recording/monitoring in the first place, or am I wrong?
To be honest we just want synced audio as asked in my other posts for monitoring purposes, this post is really an on-the-way prob in me trying to solve it.