search for: __ast_play_and_record

...son ... is busy. Please leave a message <beep>"), then is disconnected immediately after the beep. I see the following in the logs: channel.c:2992 in set_format: Set channel Mobile/SCP2500-7fc8 to write format slin rtp.c:1089 in ast_rtcp_read: Got RTCP report of 104 bytes app.c:657 in __ast_play_and_record: One waitfor failed, trying another app.c:661 in __ast_play_and_record: No audio available on Mobile/SCP2500-7fc8?? Calling via any other method to leave voicemail works correctly. Also, when making outgoing calls (eg softphone -> asterisk -> bluetoothphone -> pstn), audio from the soft...

...te say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten => stress,1,Answer() exten => stress,2(vm),Voicemail(7777|su) exten => stress,3,Hangup() however, if I use sipp to test this, I get [Jan 23 14:43:51] WARNING[22782]: app.c:599 __ast_play_and_record: No audio available on SIP/sipp-b7c274b0?? I suspect that's because sipp itself is not sending audio. Is there any tricks I can do in the dialplan to get an extension to answer sipp and then send it to voicemail, but play some audio for the voicemail ? Thanks. Julian.

...mmand that I use is: sipp -sn uac_pcap -l 1 -m 1 -s 55 -trace_err 192.168.13.6 But, If I use the file g711a.pcap included in the sources of sipp or if use some file captured for me the result is the same ---> error ... the message in Asterisk is: [Apr 2 02:16:14] WARNING[21197]: app.c:674 __ast_play_and_record: No audio available on SIP/sipp-082402b0?? I've tested compiling the sources but still with the same error. I've changed the xml file but I keep failing. Please, How do I test my voicemail but recording audio? Is there other tool to help me? A lot of thanks. Pepo. -- Lin...

...tro' (language 'en') -- <SIP/<vpuser>-081d2800> Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/1337/tmp/KzD4A1 format: wav, 0x8184358 [Oct 31 08:21:02] WARNING[22354]: app.c:602 __ast_play_and_record: No audio available on SIP/<vpuser>-081d2800?? -- User hung up [Oct 31 08:21:02] NOTICE[22354]: pbx.c:1631 pbx_substitute_variables_helper_full: Error in extension logic (missing '}') == Spawn extension (voicepulse-in, 14259491337, 5) exited non-zero on 'SIP/<vpuser>-0...

...v49, 0x7fb880008408 -- x=1, open writing: /var/spool/asterisk/voicemail/default/1060/tmp/tJ2W4E format: gsm, 0x7fb88000f618 -- x=2, open writing: /var/spool/asterisk/voicemail/default/1060/tmp/tJ2W4E format: wav, 0x7fb8800244d8 [Oct 24 19:46:23] WARNING[3005][C-00000000]: app.c:1384 __ast_play_and_record: No audio available on SIP/1061-00000000?? -- User hung up == Spawn extension (default, stdexten-BUSY, 1) exited non-zero on 'SIP/1061-00000000' == WebSocket connection from '192.168.100.71:42822' closed