I am facing one way audio problem in sip trunking between asterisk and
avaya.
+-------------+ +----+
| avaya sip |-------| P1 |
+-------------+ +----+
|
|
|
+-------------+
| Asterisk | WAN
-------------------------------------------------
| | LAN
+-------------+
|
/
+----+ /
| P2 |--+
+----+
When P1 dial P2, P2 hears voice clear but P1 could not hear any voice.
My sip.conf is
[avaya]
type=peer
fromdomain=xx.xx.xx.xx
host=xx.xx.xx.xx
disallow=all
allow=ulaw
dtmfmode=rfc2833
canreinvite=yes
--
Regards,
Shariq Khan
0333-3501125
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users