Audio only available when i talk on my phone

Asterisk Asterisk Support
1

Hello,

David if you see this, i’ve got out bound calls working :smile:

anyway new problem,

trying to setup voicemail, i call the voice mail, but i have to speak before i can hear anything.

its as if it only activates sound when i talk. ? Codec issue? or a file format issue, i’ve got it set to wav49, in the voicemail.conf

I’m a little baffled.

And i can’t seem to leave voice mails now…
= Using SIP RTP CoS mark 5
– Called SIP/1001
– SIP/1001-00000048 is ringing
– Nobody picked up in 20000 ms
– Executing [1001@default:3] VoiceMail(“SIP/sipgate-00000047”, “2001@default”) in new stack
– <SIP/sipgate-00000047> Playing ‘vm-intro.gsm’ (language ‘en’)
> doing dnsmgr_lookup for ‘sipgate.co.uk’
> ast_get_srv: SRV lookup for ‘_sip._udp.sipgate.co.uk’ mapped to host sipgate.co.uk, port 5060
– <SIP/sipgate-00000047> Playing ‘beep.gsm’ (language ‘en’)
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/default/2001/tmp/LArf5y format: wav49, 0x825c00
– x=1, open writing: /var/spool/asterisk/voicemail/default/2001/tmp/LArf5y format: gsm, 0x815b30
– x=2, open writing: /var/spool/asterisk/voicemail/default/2001/tmp/LArf5y format: wav, 0x7ffcd8
– x=3, open writing: /var/spool/asterisk/voicemail/default/2001/tmp/LArf5y format: g723sf, 0x822ac8
[Apr 17 14:53:24] WARNING[26365]: translate.c:300 ast_translator_build_path: No translator path from unknown to unknown
[Apr 17 14:53:24] WARNING[26365]: file.c:190 ast_writestream: Unable to translate to format g723sf, source format slin
[Apr 17 14:53:24] WARNING[26365]: app.c:900 __ast_play_and_record: Error writing frame
== Spawn extension (default, 1001, 3) exited non-zero on ‘SIP/sipgate-00000047’

Thank you :smile:

Kind Regards

Joshua Shipman

2

No timing source.

3

ok sorted voicemail.

Timing source for a call ?

so i need like a DAHDI timing source to use.

this would enable me to hear audio before i speak on the phone to activate it?

Thank you by the way.

4

Only hearing outgoing recordings whilst you are speaking suggests that you don’t have internal, timing enabled, or don’t have a timing source. This need not be dahdi in current versions of Asterisk.

However, you also seem to be requesting recording in a format for which you don’t have a codec translation path. Note that you must specify the codec when doing originate, otherwise signed linear will be used.