Hi!
FaberK wrote:
http://pastebin.ca/271763
On the new configuration, I've also changed the codecs, leaving the g711
only.
Unfortunately always the same: calling my number, the call reach the
2600(infact I hear the tone), but is not forwarded to the sip-server.
Have you got a success?
Try to check what dial-peer the call matches - execute 'show call active
voice brief' and under Total call-legs line you'll see info regarding
each call leg like the following:
3F17 : 519908760ms.1 +40 pid:101 Answer 390429782580 connected
dur 19:48:44 tx:113/2108 rx:252/3958
IP aa.bb.cc.dd:36604 rtt:16ms pl:3510/0ms lost:0/1/0 delay:60/60/70ms
g729br8
33EE : 591206120ms.1 +40 pid:101 Answer 000999125 active
dur 00:00:27 tx:1370/45210 rx:1225/40425
IP aa.bb.cc.dd:52190 rtt:0ms pl:4990/3300ms lost:8/543/182
delay:90/70/210ms gsmfr
Here "pid:" shows number of the dial-peer call leg had matched. If it's
OK and if dial-peer was configured correctly with session protocol
sipv2, session target ipv4:XXX.XXX.XXX.115 and session transport tcp,
request should hit the asterisk box assuming IP connectivity is OK.
Yesterday I've been pursuing the similar configuration and imagine my
surprise when it turned out this did not work because of lack of IP
connectivity :)
--
Sincerely,
Andrew Pogrebennyk
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