HELP HELP, Cannot Answer calls on extensions from QUEUE

Asterisk Asterisk Support
1

my incoming calls are queued and then rings and available local extension, except when I try to answer the call is hungup. Why?

Extensions can call each other fine. I have asterisk 1.2.13 with trixbox.

Two days of work no luck. Please help me.

2

read the sticky at the top of the forum, then come back and post your logs for a failed call.

3

Here are my logs:

[general]
port = 5060 ; Port to bind to (SIP is 5060)
externip=xxx.xx.xx.xxx
localnet=192.168.2.3/255.255.255.0
bindaddr =0.0.0.0; Address to bind to (all addresses on machine)
context=default
nat=yes
; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
#include additional_a2billing_sip.conf

outgoing:

allow=g729
context=from-trunk
host=did.voip.com
insecure=very
nat=yes
type=peer

Incoming
allow=g729
context=from-trunk
insecure=very
nat=yes
type=peer

ERROR LOG

INVITE sip:112@192.168.2.100:56224;rinstance=a6b6a1943fa1a97f SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK3dbc0ed3
From: “6015556666” sip:6015556666@192.168.2.3;tag=as3cb5dd51
To: sip:112@192.168.2.100:56224;rinstance=a6b6a1943fa1a97f
Contact: sip:6015556666@192.168.2.3
Call-ID: 07e37600752c40c627dddd092876e05d@192.168.2.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Dec 2006 12:17:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 2460 2460 IN IP4 192.168.2.3
s=session
c=IN IP4 192.168.2.3
t=0 0
m=audio 12330 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

Dec 15 07:17:34 VERBOSE[3062] logger.c:
<-- SIP read from 192.168.2.100:56224:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK3dbc0ed3
Contact: sip:112@192.168.2.100:56224;rinstance=a6b6a1943fa1a97f
To: sip:112@192.168.2.100:56224;rinstance=a6b6a1943fa1a97f;tag=b27ff165
From: "6015556666"sip:6015556666@192.168.2.3;tag=as3cb5dd51
Call-ID: 07e37600752c40c627dddd092876e05d@192.168.2.3
CSeq: 102 INVITE
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0

Dec 15 07:17:34 VERBOSE[3062] logger.c: — (9 headers 0 lines) —
Dec 15 07:17:34 DEBUG[3062] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on ‘07e37600752c40c627dddd092876e05d@192.168.2.3’ Request 102: Found
Dec 15 07:17:36 VERBOSE[3062] logger.c:
<-- SIP read from 64.154.41.60:51668:
BYE sip:5555551212@74.244.73.149:5060 SIP/2.0
Via: SIP/2.0/UDP 64.154.41.60:5060
From: sip:6015556666@64.154.41.60;tag=7E5080DB-1EDB
To: sip:5555551212@74.244.73.149;tag=as3edb454a
Date: Fri, 15 Dec 2006 12:15:37 GMT
Call-ID: CE7144F1-8B6C11DB-B189F654-8C59149@64.154.41.60
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 6
Timestamp: 1166185056
CSeq: 102 BYE
Content-Length: 0

Dec 15 07:17:36 VERBOSE[3062] logger.c: — (11 headers 0 lines) —
Dec 15 07:17:36 VERBOSE[3062] logger.c: Sending to 64.154.41.60 : 5060 (NAT)
Dec 15 07:17:36 VERBOSE[3062] logger.c: Transmitting (NAT) to 64.154.41.60:51

4

Ok, this really pisses me off. After rebuild asterisk box reconfig, etc the issue was I must press a # sign once the call is answered for it to connect. That’s it.

Ok, I am better now… Hope this helps someone else