Rich Adamson a �crit :
[...]
(Given you mentioned g729, its
likely you're trying to use sip across dsl broadband.)
Yes
If you have a nat/firewall function between * and the sip phone, look there first.
No nat/firewall between
In followup posts, you might mention how your stuff
is configured and include appropriate portions of your sip.conf.
Setup is
<--------- SIP phones -------->-* <------- SIP softphones ------> | --- asterisk --- internet <------- IAX softphones ------> | on nat/firewall <H323 softphones through GnuGK>-* box
What I face is that a SIP call to our GW has from time to time the behaviour to "loose" audio. Hanging up and retrying can work, but mostly we wait or use an IAX GW and try again and then it work. Can also take few hours before it work again.
Our provider see the calls like connected with the right codec, and I can see them too in * logs.
This is the sip.conf relevant part from mainphone BT ATA 186 In this one, prefered codec are G729, Ulaw and Alaw.
[106]
type=friend ; either "friend" (peer+user), "peer" or "user"
context=default
qualify=yes
username=106 ; usually matches the [section] title
secret=<secret>
host=192.168.10.6
canreinvite=no ; allow RTP voice traffic to bypass Asterisk
dtmfmode=rfc2833 ; either RFC2833 or INFO for the BudgeTone
disallow=all ; need to disallow=all before we can use allow=
allow=ulaw ; Note: In user sections the order of codecs
allow=alaw ; listed with allow= does NOT matter!
allow=g729 ; Pass-thru only unless g729 license obtained
allow=gsm ; allways allow gsm
allow=ilbc
Thanks for your help
-- Daniel _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
