[Asterisk-bsd] SIP Clients - No voice.

[Richard Neese]

On August 31, 2004 02:48 pm, Jefferson Carvalho wrote:

ok what codecs are you using? what type of phone ?  and cut and paste one of 
your basic sip setup for one of the phones....


> Thanks Dr. Rich Murphey,
>
> Look what i got!
>
> apollo*CLI> set verbose 9
>     -- Executing Dial("SIP/1260-3e37", "SIP/1261|20|rtT") in new stack
>     -- Called 1261
>     -- SIP/1261-c412 is ringing
>     -- SIP/1261-c412 answered SIP/1260-3e37
>     -- Attempting native bridge of SIP/1260-3e37 and SIP/1261-c412
>     -- Attempting native bridge of SIP/1260-3e37 and SIP/1261-c412
> Aug 31 15:48:34 WARNING[135442432]: chan_sip.c:673 retrans_pkt: Maximum
> retries exceeded on call
> 9F37D538-FB63-11D8-B3A5-005004803F0B at 172.20.1.133 for seqno 2746
> (Non-critical Response)
>
> The other side answer the call , the connection is done .. but i don't
> have voice
> on the other side!!!!!!
>
> Regards,
>
> -Jefferson Carvalho
>
> Dr. Rich Murphey escreveu:
> >Try invoking
> >
> >set verbose 9
> >
> >At the asterisk console prompt and watch the console messages as the calls
> >progress for more info.
> >
> >Rich
> >
> >>-----Original Message-----
> >>From: asterisk-bsd-bounces at lists.digium.com
> >>[mailto:asterisk-bsd-bounces at lists.digium.com] On Behalf Of
> >>Jefferson Carvalho
> >>Sent: Tuesday, August 31, 2004 8:44 AM
> >>To: asterisk-bsd at lists.digium.com
> >>Subject: [Asterisk-bsd] SIP Clients - No voice.
> >>
> >>Hello list,
> >>
> >>I could install and configure asterisk on my
> >>FreeBSD-5.2.1-Release :) I create 2 extensions and my clients
> >>can call each other but there's no sound on those calls !!!!
> >>Why????
> >>Someone could help me ?
> >>
> >>Regards,
> >>
> >>-Jefferson Carvalho
> >>
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