[Asterisk-Users] Call gets disconnected upon connect

[Caleb]

Hi Everybody,

I am trying to use SIP (Sipura 2000) to connect to Asterisk which then
dials out a local number using the Digium E100P. We have purchased the
G729 codec licenses from Digium and loaded them into Asterisk
successfully. However, the call drops immediately after being answered
with the debug error message saying something like: "channel.c:2646
ast_channel_bridge: Didn't get a frame from channel:
SIP/6568543197-9af5"

Any idea why this may be happening? Here is the debug log:

Oct  4 00:53:41 DEBUG[1083546560]: chan_sip.c:5269 check_user_full:
Setting NAT on RTP to 4
Oct  4 00:53:41 DEBUG[1083546560]: chan_sip.c:7087 handle_request:
Check for res for 6568543197
Oct  4 00:53:41 DEBUG[1083546560]: chan_sip.c:1650
update_user_counter: Call from user '6568543197' is 1 out of 0
Oct  4 00:53:41 DEBUG[1083546560]: chan_sip.c:4492 build_route:
build_route: Contact hop: +6568543197
<sip:6568543197 at 192.168.1.103:5060>
    -- Executing SetVar("SIP/6568543197-86c2", "sip_codec=g729") in new stack
    -- Executing Dial("SIP/6568543197-86c2", "Zap/g1/91596323") in new stack
    -- Called g1/91596323
Oct  4 00:53:41 DEBUG[1146877376]: rtp.c:1162 ast_rtp_write: Ooh,
format changed from UNKN to G729A
Oct  4 00:53:41 DEBUG[1146877376]: rtp.c:438 ast_rtp_read: RTP NAT:
Using address 68.2.178.157:16410
Oct  4 00:53:46 DEBUG[1089555136]: chan_zap.c:1179 zt_enable_ec:
Enabled echo cancellation on channel 1
    -- Zap/1-1 is ringing
Oct  4 00:53:46 WARNING[1146877376]: channel.c:1441 ast_indicate:
Unable to handle indication 3 for 'SIP/6568543197-86c2'
Oct  4 00:54:01 DEBUG[1089555136]: chan_zap.c:1163 zt_enable_ec: Echo
cancellation already on
    -- Zap/1-1 answered SIP/6568543197-86c2
Oct  4 00:54:02 DEBUG[1083546560]: chan_sip.c:823 __sip_ack: Stopping
retransmission on 'e737d90b-19e8aa7f at 192.168.1.103' of Response 102:
Found
Oct  4 00:54:04 DEBUG[1146877376]: channel.c:2646 ast_channel_bridge:
Didn't get a frame from channel: SIP/6568543197-86c2
Oct  4 00:54:04 DEBUG[1146877376]: channel.c:2716 ast_channel_bridge:
Bridge stops bridging channels SIP/6568543197-86c2 and Zap/1-1
Oct  4 00:54:04 DEBUG[1146877376]: chan_zap.c:2420 zt_setoption: Set
option AUDIO MODE, value: ON(1) on Zap/1-1
Oct  4 00:54:04 DEBUG[1146877376]: chan_zap.c:1933 zt_hangup: Hangup:
channel: 1 index = 0, normal = 16, callwait = -1, thirdcall = -1
Oct  4 00:54:04 DEBUG[1146877376]: chan_zap.c:2069 zt_hangup: Not yet
hungup...  Calling hangup once with icause, and clearing call
Oct  4 00:54:04 DEBUG[1146877376]: chan_zap.c:1211 zt_disable_ec:
disabled echo cancellation on channel 1
Oct  4 00:54:04 DEBUG[1146877376]: chan_zap.c:2332 zt_setoption: Set
option TDD MODE, value: OFF(0) on Zap/1-1
Oct  4 00:54:04 DEBUG[1146877376]: chan_zap.c:1154 update_conf:
Updated conferencing on 1, with 0 conference users
Oct  4 00:54:04 DEBUG[1146877376]: chan_zap.c:2414 zt_setoption: Set
option AUDIO MODE, value: OFF(0) on Zap/1-1
Oct  4 00:54:04 DEBUG[1146877376]: chan_zap.c:1211 zt_disable_ec:
disabled echo cancellation on channel 1
    -- Hungup 'Zap/1-1'
Oct  4 00:54:04 DEBUG[1146877376]: app_dial.c:975 dial_exec: Exiting
with DIALSTATUS=ANSWER.
  == Spawn extension (6568543197, 6591596323, 2) exited non-zero on
'SIP/6568543197-86c2'
Oct  4 00:54:04 DEBUG[1146877376]: chan_sip.c:1749 sip_hangup:
update_user_counter(6568543197) - decrement inUse counter