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I have set up a FreePBX Asterisk Server and I configured paging announcement to a set of extension. When I dial the paging extension the annuncement is played but it stopped after 29 seconds, although the announcement file is longer than 29 seconds.enter image description here

Tech Dog
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  • Does it continue the call or just hangs up? – Anubioz Feb 23 '17 at 11:40
  • @Anubioz Thanks for reply. It does not hang up. From the caller side, the call is still in process. I am using Grandstream phone and Portech speakers. At the Asterisk debug output, I get the following output.. http://tinypic.com/r/2ufu0i0/9 . Also the time is 29 seconds when I counted exactly – Tech Dog Feb 27 '17 at 04:40
  • Your asterisk seem to be mixing sound files with different codecs, which is wrong. The extension `.slin` in file `us-rah-par-small.slin` means that your file is 16 bit Signed Linear PCM encoded (i.e. pure wav). No telephony device or program supports that. Freepbx doesn't support it without ffmpeg. You should delete /var/lib/asterisk/sounds/en/custom/us-rah-par-small.slin file and leave only the ulaw version – Anubioz Feb 27 '17 at 06:11
  • Ok I try this and let you know – Tech Dog Feb 27 '17 at 06:49
  • @Anubioz Hi... I tried with ulaw version but the problem is still there. http://i63.tinypic.com/2yvlb3c.jpg – Tech Dog Feb 27 '17 at 07:19
  • Have you played the ulaw file on an external player to check that it isn't a bad encoder problem? – Anubioz Feb 27 '17 at 08:22
  • @Anubioz Thanks for reply. Yes I played and it is perfectly OK. It is even playing full length in the Web-based GUI of FreePBX, where this file is uiploaded – Tech Dog Feb 27 '17 at 08:48
  • Web-based GUI can't play ulaw. It plays the original mp3 file you had uploaded. You should play the ulaw itself to check if it plays. I had encountered similar problems, when using badly encoded files. Anyway, can you hear the beep.gsm after 29 seconds? Or does it beep after the full announcement length of silence has been waited? – Anubioz Feb 27 '17 at 10:40
  • @Anubioz I downloaded the ulaw file and played it on the Awave studio. It is playing perfectly OK. Is there anything I can check? Thanks for your kind help. The beep is played before the announcement... after 29 seconds the audio is just off – Tech Dog Feb 28 '17 at 06:31
  • @Anubioz I have used TCP dump and Wireshark. The VOIP call flow is in the attached image http://i67.tinypic.com/125qpw7.png – Tech Dog Feb 28 '17 at 09:30
  • Well, it seems like we ruled out network problems. Now you can to create a new extension (Aplications->Extensions), enable Find/Follow me with Ring Time 60, Announcement->your record, "No answer" Destination setting to Announcements->your record. On the "Advanced" tab, find "Optional Destinations" (bottom) & set every single one to Announcements->your recording. Save, apply then call to this extension from an internal phone. It should play the full recording that way. If it doesn't try another audio file with a shorter length to see if call will be routed correctly between destinations... – Anubioz Feb 28 '17 at 10:02
  • @Anubioz Hi.. It played the full announcement that way and the caller listens full recording.... However, the problem is other way around. Since we are making a paging call, the announcement should be played on the remote VoIP Speakers.... We do this by adding the Extension to a Paging group and calling the Paging group extension... then the announcement is not played in full. – Tech Dog Mar 02 '17 at 03:43
  • @Anubioz Hi...Any further help that you can provide? – Tech Dog Mar 06 '17 at 04:55
  • Hi! Sure, you can make everything work by adding a custom context to `/etc/asterisk/custom_extensions.conf` which would connect callee to the working extension with announcment you've created earlier. After that you just add `asterisk -x "channel originate Local/ extension 100@your_context"` to your crontab & it will call that number and play announcement when executed. More info https://www.voip-info.org/wiki/view/Asterisk+cli+originate – Anubioz Mar 06 '17 at 10:44

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