[asterisk-users] No Sound on Zap Channels
Jon Weisman
jweisman at ibell.net
Thu Sep 13 09:32:45 CDT 2007
I have a feeling the dchannel is bad. I'll investigate further and post my
findings.
-Jon
----- Original Message -----
From: "Atis" <atis at BEST.eu.org>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Thursday, September 13, 2007 4:12 AM
Subject: Re: [asterisk-users] No Sound on Zap Channels
> On 9/13/07, Hoai-Anh Ngo-Vi <hoaianh at gmx.de> wrote:
>> Have you answered the channel?
>
> Voicemail doesn't require Answer(). It does that itself, as you
> usually get to voicemail after Dial(). It would be silly to require to
> do Answer after each Dial and then send to voicemail.
>
> Regards,
> Atis
>
>>
>> Von: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] Im Auftrag
>> von Jon Weisman
>
>> I've got a strange issue here. When I make a SIP call to say my voicemail
>> app, I hear audio just fine. However when I dial from PSTN into my
>> Asterisk
>> box, I see that its playing the voice files, but I hear nothing, then the
>> call drops. I'm running Fedora Core 6, and Asterisk 1.2.24. CLI output
>> below. T-1 is PRI, showing normal, dchannel is up as well. Any help is
>> greatly appreciated.
>>
>>
>>
>>
>>
>>
>>
>>
>> Thanks,
>>
>>
>> Jon
>>
>>
>>
>>
>>
>>
>>
>>
>> -- Accepting call from '2125551212' to '6465551212' on channel 0/23,
>> span 4
>> -- Executing VoiceMail("Zap/95-1", "u100") in new stack
>> -- Playing 'vm-theperson' (language 'en')
>> -- Playing 'digits/1' (language 'en')
>> -- Playing 'digits/0' (language 'en')
>> -- Playing 'digits/0' (language 'en')
>> -- Playing 'vm-isunavail' (language 'en')
>> -- Playing 'vm-intro' (language 'en')
>> -- Channel 0/23, span 4 got hangup request, cause 34
>> == Spawn extension (default, 6465551212, 1) exited non-zero on
>> 'Zap/95-1'
>> -- Hungup 'Zap/95-1'
>> _______________________________________________
>>
>> Sign up now for AstriCon 2007! September 25-28th.
>> http://www.astricon.net/
>>
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> Atis Lezdins,
> IT Responsible of BEST Riga,
> atis at BEST.eu.org
> ICQ: 142239285
> Skype: atis.lezdins
> Cell Phone: +371 28806004 [Tele2, Latvia]
> Work phone: +1 800 7502835 [Toll free, USA]
> ?BEST? -> www.BEST.eu.org
>
> _______________________________________________
>
> Sign up now for AstriCon 2007! September 25-28th.
> http://www.astricon.net/
>
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list