[Asterisk-bsd] SIP Clients - No voice.

[Jefferson Carvalho]

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
>>
>>_______________________________________________
>>Asterisk-BSD mailing list
>>Asterisk-BSD at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-bsd
>>
>>
>>    
>>
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