Rich Adamson a �crit :
Inline and somewhat confused....
Sorry for that
[...]
Setup is
<--------- SIP phones -------->-*
<------- SIP softphones ------> | --- asterisk --- internet
<------- IAX softphones ------> | on nat/firewall
<H323 softphones through GnuGK>-* box
Is the above "on nat/firewall" supposed to say "no nat/firewall"?
So, what that drawing suggests is that asterisk is routing packets
between the Internet and the 192.168.10.x network. Is there something
else involved? If not, then asterisk apparently is your nat box.
Yes asterisk _is_ the nat/firewall box. This is the meaning to the above "on nat/firewall box" I din't use the good words ;-)
[...]
Do your log records tend to suggest the IAX link is dropping, a SIP
link to somewhere, or what?
Logs suggest (and show) that call where normally answered. No drop, nothing. And the status at the end of the call is answered. Remember that this audio behaviour affect only one party, usually the called party.
[...]
If I recall your original posting, it was oriented around a sip call
using g729 dropping connections. How does that relate to the above
diagram and the sip.conf entry below?
asterisk --- internet --- GW SIP --- internet --- landline net phone (nearest possible from called party)
[sipgateway-gw] type=peer secret=MySecret username=MyUsername host=sip.provider canreinvite=no disallow=all allow=alaw allow=ulaw allow=g729
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
Why is there a "allow=" with nothing after the "=" in the above.
Error from copy/paste. Isn't.
The "canreinvite=no" comment says "allow RTP" which is backwards. What
are you actually expecting?
That all the RTP traffic as going through asterisk. As I'm behind NAT...
Since the explanations for the above stuff is a little thin, I'd have to guess that you might have one or more internal sip phones that don't have "canreinvite=no"
Some of EP don't have this value set, which mean the default will be used. Those EP are defined but not connected (used only for test). Today only my ATA 286 is connected
and at least one Internet sip phone that is
unknown as to how it is configured. That's reading way between the lines
and guessing a lot.
No Internet sip phone. All connections are coming from my intranet.
Thanks for help me to debug this. -- 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
