this is my CLI output: then as the client answer the phone it just stop....like being cut-off no activity.......my asterisk version is asterisk 1.4.23
-- Added extension '8600199' priority 1 to default
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI("SIP/1002-b7906a60", "AGI(
agi://127.0.0.1:4577/call_log") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/AGI(
agi://127.0.0.1:4577/call_log
-- AGI Script AGI(
agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/1002-b7906a60", "sip/12127775678@Prime|55|o") in new stack
-- Called 12127775678@Prime
-- SIP/Prime-09ee94d8 is making progress passing it to SIP/1002-b7906a60
-- SIP/Prime-09ee94d8 answered SIP/1002-b7906a60
-- Attempting native bridge of SIP/1002-b7906a60 and SIP/Prime-09ee94d8
== Spawn extension (default, 912127775678, 2) exited non-zero on 'SIP/1002-b7906a60'
MY SIP DEBUG:
From: "1002" <sip:1002@192.168.2.3>;tag=as622f8fad
To: <sip:16076254179@70.42.72.72>;tag=cba-2833-49876c85
Call-ID:
0686e154512ae7d33d63348f28741891@192.168.2.3
CSeq: 103 INVITE
Contact: <sip:16076254179@70.42.72.72>
Date: Mon, 02 Feb 2009 21:58:42 GMT
Server: BRSIP v2.0.1.2
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
Allow-Events: keep-alive, message-summary
Supported: timer
Content-Type: application/sdp
Content-Length: 214
v=0
o=BRSDP 1583155 1583156 IN IP4 66.162.83.70
s=BRSDP Session
c=IN IP4 66.162.83.70
t=0 0
m=audio 19542 RTP/AVP 18 101
a=ptime:20
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
--- (14 headers 10 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 66.162.83.70:19542
Found description format G729
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:16076254179@70.42.72.72> for address/port to send to
set_destination: set destination to 70.42.72.72, port 5060
Transmitting (no NAT) to 70.42.72.72:5060:
ACK sip:16076254179@70.42.72.72 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK784811f8;rport
From: "1002" <sip:1002@192.168.2.3>;tag=as622f8fad
To: <sip:16076254179@70.42.72.72>;tag=cba-2833-49876c85
Contact: <sip:1002@192.168.2.3>
Call-ID:
0686e154512ae7d33d63348f28741891@192.168.2.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0