search for: ast_play_and_record

...-- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/105/INBOX/msg0001 format: g729, 0x8140f88 Jun 1 10:08:45 WARNING[15148]: channel.c:2326 set_format: Unable to find a codec translation path from g729 to slin Jun 1 10:08:45 WARNING[15148]: app.c:621 ast_play_and_record: Unable to set to linear mode, giving up Obviously I don't have codec_g729 installed. The real question is, why does it need to convert to slinear? Thanks! -- Kristian Kielhofner

...-- Playing 'digits/7' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message Feb 8 14:07:53 DEBUG[4195]: app.c:549 ast_play_and_record: play_and_record: <None>, /var/spool/asterisk/voicemail/voicepulse_connect_context/7777/INBOX/msg0001, 'wav49|gsm|wav' Feb 8 14:07:53 DEBUG[4195]: app.c:566 ast_play_and_record: Recording Formats: sfmts=wav49 -- x=0, open writing: /var/spool/asterisk/voicemail/voicepulse_connect...

...ly the sample barebones sip setup from the O'Reilly site (onlamp.com). The exact problem is that I can get sound out of asterisk to my sip extensions, but asterisk is not able to get any sound from the sip devices. The console error I get is: Sep 24 09:33:03 WARNING[1111635520]: app.c:599 ast_play_and_record: No audio available on SIP/04-e2f4?? -- User hung up I've only tried soft phones so far, but I have tried several different ones, and all of them seem to have the same problem. I'm pretty sure the problem is with my * config, or some missing libraries on my server. Other than th...

...f time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP phone connected to Box A, upon leaving voicemail I get in the following: -- x=0, open writing: /mnt/nfs/123/INBOX/msg0004 format: wav49, (nil) Mar 21 17:28:18 WARNING[8576]: app.c:706 ast_play_and_record: Error creating writestream '/mnt/nfs/123/INBOX/msg0004', format 'wav49' Mar 21 17:28:18 WARNING[8576]: app_voicemail.c:787 base_encode: Failed to open log file: /mnt/nfs/123/INBOX/msg0004.wav: No such file or directory -- Executing Hangup("Zap/5-1", "") i...

...5 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v oicemail/default/300/INBOX/msg0000.WAV: No such file or directory -- x=0, open writing: /var/spool/asterisk/voicemail/default/300/INBOX/msg0000 format: wav49, (n il) Mar 24 12:48:35 WARNING[344081]: app.c:701 ast_play_and_record: Error creating writestream '/var/spo ol/asterisk/voicemail/default/300/INBOX/msg0000', format 'wav49' Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:784 base_encode: Failed to open log file: /var/spoo l/asterisk/voicemail/default/300/INBOX/msg0000.WAV: No such file or directory...

I recently setup app_meetme with the 'i' option. My boss wants users to say their name and go directly into the conference instead of reviewing the recording. If anyone else is interested in this behavior becoming an option, has a suggestion what letter to use as the option (I was thinking 'i' -- with review and 'I' -- without review), or anything else, I'd appreciate

...=> 312,102,Voicemail(b312) exten => 312,103,Hangup exten => 313,1,Dial(SIP/313,10) exten => 313,2,Voicemail(u313) exten => 313,102,Voicemail(b313) exten => 313,103,Hangup Voicemail works, but i can not leave a message from a sip phone: an 7 08:25:32 WARNING[393234]: app.c:615 ast_play_and_record: No audio available on SIP/313-47b0?? -- User hung up Urgent handler but i can do that from a zap device. I use asterisk debian's packages from testing. ii asterisk 1.0.2-2 Open Source Private Branch Exchange (PBX) ii asterisk-doc 1.0.2-2 Documentation for asteri...

...: wav49, 0x814cb60 -- x=1, open writing: /home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005 format: gsm, 0x814d068 -- x=2, open writing: /home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005 format: wav, 0x8144980 Mar 6 18:41:45 WARNING[3539]: app.c:619 ast_play_and_record: No audio available on SIP/69.70.x.x-08149a98?? -- User hung up == Spawn extension (ser, 900, 1) exited non-zero on 'SIP/69.70.x.x-08149a98' Destroying call 'ixiXpRvNGSyIBxmn@192.168.1.103' If I use rewritehostport instead of forward, the call does not reach asterisk: failu...

I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com

I've beaten myself bloody dealing with this one... No luck so far. In summary, incoming calls from Gizmo establish, but neither get nor send sound. Outbound calls to Gizmo work fine (well a bit choppy but work) My thought is that the SIP connection is being made fine, but the RTP is getting stopped / blocked / misdone somewhere. Here is the thing: Asterisk 2.5 on Linux (No hardware

...;en') pbx*CLI> Sip read: 0 headers, 0 lines -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/internalextensions/230/INBOX/msg0008 format: wav49, 0x8144680 Jun 24 00:56:31 WARNING[7507]: app.c:619 ast_play_and_record: No audio available on SIP/233-a3ba?? -- User hung up -- Executing GotoIf("SIP/233-a3ba", "1?menuinternal|t|2") in new stack -- Goto (menuinternal,t,2) -- Executing Wait("SIP/233-a3ba", "4") in new stack -- Executing Goto("SIP/233-a3b...