Hi all,

on asterisk 1.8.16

[2012-10-20 19:36:17] VERBOSE[743] pbx.c: -- Executing [801@OFFICE-Numbers:2] MusicOnHold("Local/801@OFFICE-Numbers-e54a;2", "") in new stack [2012-10-20 19:36:17] VERBOSE[743] res_musiconhold.c: -- Started music on hold, class 'TOOTAi', on Local/801@OFFICE-Numbers-e54a;2
[2012-10-20 19:36:17] WARNING[742] translate.c: no samples for ulawtolin
[2012-10-20 19:36:21] VERBOSE[742] pbx.c: == Spawn extension (from_to-OFFICE, 801, 23) exited non-zero on 'SIP/8081773619-00002f28' [2012-10-20 19:36:21] VERBOSE[743] res_musiconhold.c: -- Stopped music on hold on Local/801@OFFICE-Numbers-e54a;2

or asterisk 10.8.0

-- Executing [801@macro-GeneralNumbers:1] Set("SIP/105-00000081", "CHANNEL(musicclass)=TOOTAi") in new stack -- Executing [801@macro-GeneralNumbers:2] MusicOnHold("SIP/105-00000081", "") in new stack
    -- Started music on hold, class 'TOOTAi', on SIP/105-00000081
[2012-10-20 22:48:48] WARNING[22435]: translate.c:343 framein: no samples for g722tolin
    -- Stopped music on hold on SIP/105-00000081

This is when calling extension:

exten=>801,1,Set(CHANNEL(musicclass)=TOOTAi)
exten=>801,n,MusicOnHold()
exten=>801,n,Hangup

What does mean those WARNINGS and how to solve this problem?

MeetMe, Voicemail or holding a call are working fine. From what I understand, codecs are used in channels and format for handling files. In both cases, two different servers, asterisk is compiled from tar.gz and in menuselect all codecs and formats are activated.

Is this a bug? Did I forget something?

On a third server I run latest Elastix with an asterisk 1.8.16 version. On this server I have no MusicOnHold at all even during calls. Logs show

VERBOSE[19717] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/104-000000b3 VERBOSE[19717] res_musiconhold.c: -- Stopped music on hold on SIP/104-000000b3

which is MusicOnHold stop immediately.

On all servers wav files are installed, even try with original ones delivered with Asterisk.

Thanks for any hint

Regards
--
Daniel

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to