[Asterisk-Users] SIP registration

[Tais M. Hansen]

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On Thursday 18 September 2003 19:04, Hielke Christian Braun wrote:
> try to change  [siptestphone] to [atrg613test] in sip.conf. Maybe
> that helps.

It didn't. And now something else is weird. Asterisk fails sending audio to my 
SIP phone. Found this in my logs:

Sep 19 11:08:52 WARNING[950291]: File channel.c, Line 1819 
(ast_channel_make_compatible): No path to translate from 
SIP/sc.sc.sc.sc-de54(
4) to H323/ip$hc.hc.hc.hc:1244/14060(8)
Sep 19 11:08:58 WARNING[147466]: File chan_sip.c, Line 443 (retrans_pkt): 
Maximum retries exceeded on call [hex]@
as.as.as.as for seqno 102 (Request)
Sep 19 11:09:04 WARNING[147466]: File chan_sip.c, Line 443 (retrans_pkt): 
Maximum retries exceeded on call [hex]@
as.as.as.as for seqno 102 (Request)

What on earth is this? Codec?

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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