Joe,
I double checked its not a problem with the gateway. I re-emphasize that when the call is answered a2billing charges from the time when I dial into a2billing.php. is there any way
the 'answered time is 66' which is wrong. My mobile shows I am conversation for 40 seconds only.
Log is at
https://www.dropbox.com/s/p8qobrh87ddg7 ... g.txt?dl=0My call started ringing at around
a2billing.php,1: file:Class.RateEngine.php - line:1223 - uniqueid:1439527873.56 - [TRUNK STATUS UPDATE : UPDATE cc_trunk SET inuse = inuse + 1 WHERE id_trunk = '2']
-- AGI Script Executing Application: (DIAL) Options: (SIP/gsm0/9066371511,60,L(4140000:61000:30000))
> Limit Data for this call:
> timelimit = 4140000 ms (4140.000 s)
> play_warning = 61000 ms (61.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000 ms (30.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =
== Using SIP RTP CoS mark 5
-- Called SIP/gsm0/9066371511
-- SIP/gsm0-00000026 is making progress passing it to SIP/6332247668-00000025
> 0x7f99b43a5890 -- Probation passed - setting RTP source address to 192.168.2.242:8000
-- SIP/gsm0-00000026 answered SIP/6332247668-00000025
-- Channel SIP/6332247668-00000025 joined 'simple_bridge' basic-bridge <505c3511-5d1a-460e-bfca-806bbbe6c504>
-- Channel SIP/gsm0-00000026 joined 'simple_bridge' basic-bridge <505c3511-5d1a-460e-bfca-806bbbe6c504>
> Bridge 505c3511-5d1a-460e-bfca-806bbbe6c504: switching from simple_bridge technology to native_rtp
> 0x7f99b43a5890 -- Probation passed - setting RTP source address to 192.168.2.242:8000
[Aug 14 10:21:46] WARNING[15618][C-00000015]: chan_sip.c:7386 sip_write: Can't send 10 type frames with SIP write