checkout http://sip.fontventa.com/content/view/16/45/
Dan On Sat, Jun 20, 2009 at 1:34 AM, Amos Shahar <[email protected]> wrote: > > > hi, > I'm trying to save video on asterisk voicemail. > enabled GSM (audio) and H263 (video) for the user 1234 in the sip.conf > and set videosupport=yes > > in voicemail.conf I tried both: > format=wav|h263 > and > format=wav > > [1234] > type=friend > host=dynamic > nat=yes > context=inbound > insecure=invite > username=1234 > secret=1234 > disallow=all > allow=gsm > allow=h263 > > x-lite has gsm and H263 only. > > when calling the VM (same occure when using "record") I'm getting the > following error: > -- Executing VOICEMAIL("SIP/1234-08f52068", "1...@inbound") > -- <SIP/1234-08f52068> Playing 'vm-intro.gsm' (language 'en') > -- <SIP/1234-08f52068> Playing 'beep.gsm' (language 'en') > -- Recording the message > -- x=0, open writing: > /var/spool/asterisk/voicemail/inbound/1234/tmp/5sY6L3 format: wav, 0x8eb8758 > -- x=1, open writing: > /var/spool/asterisk/voicemail/inbound/1234/tmp/5sY6L3 format: h263, > 0x906a228 > [Jun 20 01:32:52] WARNING[26485]: translate.c:281 > ast_translator_build_path: No translator path from unknown to unknown > [Jun 20 01:32:52] WARNING[26485]: file.c:184 ast_writestream: Unable to > translate to format h263, source format slin > [Jun 20 01:32:52] WARNING[26485]: app.c:763 __ast_play_and_record: Error > writing frame > > > Any idea? > Thanks > Amos > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-video mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-video >
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