Our surrounding:
vicibox2.0.5, kernel 2.6.18-164.el5.vnow, asterisk 1.2.30.2
location in HongKong.
Below is the sip debug info, 01 is mobile, 02 is fixed number. There is a special error code return when dial fixed number:
- Code: Select all
<-- SIP read from 192.168.2.118:44972:
--- (0 headers 1 lines) ---
-- Channel 0/1, span 1 got hangup request, cause 1
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/5800-b7a69940", "") in new stack
== Spawn extension (default, 00808607563889788, 2) exited non-zero on 'SIP/5800-b7a69940'
-- Executing DeadAGI("SIP/5800-b7a69940", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CHANUNAVAIL----------") in new stack
zapata.conf
- Code: Select all
[trunkgroups]
[channels]
context=trunkinbound
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=7.0
txgain=7.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma A102 port 1 [slot:4 bus:4 span:1] <wanpipe1>
switchtype=4ess
pridialplan=unknown
prilocaldialplan=unknow
context=trunkinbound
group=1
signalling=pri_cpe
channel =>1-23
zaptel.conf
- Code: Select all
loadzone=us
defaultzone=us
# Span 1: WCT1/0 "Wildcard TE122 Card 0"
span=1,1,0,esf,b8zs
# termtype: te
bchan=1-23
dchan=24
- Code: Select all
vici*CLI> sip debug peer 5800
SIP Debugging Enabled for IP: 192.168.2.118:44972
== 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
vici*CLI>
<-- SIP read from 192.168.2.118:44972:
INVITE sip:00808613425048025@192.168.1.39 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-877ea5363e7d7c01-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:5800@192.168.2.118:44972>
To: "00808613425048025"<sip:00808613425048025@192.168.1.39>
From: "5800"<sip:5800@192.168.1.39>;tag=2e214051
Call-ID: M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 368
v=0
o=- 2 2 IN IP4 192.168.2.118
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.2.118
t=0 0
m=audio 11302 RTP/AVP 107 0 8 101
a=alt:1 3 : I6VRe4O8 Pmp6KEag 192.168.2.118 11302
a=alt:2 2 : XVR3zCxG p6kaXWej 192.168.11.1 11302
a=alt:3 1 : cN0dDlKM ZCUmHpUD 192.168.252.1 11302
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
--- (12 headers 13 lines) ---
Using INVITE request as basis request - M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.
Sending to 192.168.2.118 : 44972 (NAT)
Reliably Transmitting (NAT) to 192.168.2.118:44972:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-877ea5363e7d7c01-1---d8754z-;received=192.168.2.118;rport=44972
From: "5800"<sip:5800@192.168.1.39>;tag=2e214051
To: "00808613425048025"<sip:00808613425048025@192.168.1.39>;tag=as3b293bb4
Call-ID: M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30418e0b"
Content-Length: 0
---
Scheduling destruction of call 'M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.' in 15000 ms
Found user '5800'
vici*CLI>
<-- SIP read from 192.168.2.118:44972:
ACK sip:00808613425048025@192.168.1.39 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-877ea5363e7d7c01-1---d8754z-;rport
To: "00808613425048025"<sip:00808613425048025@192.168.1.39>;tag=as3b293bb4
From: "5800"<sip:5800@192.168.1.39>;tag=2e214051
Call-ID: M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.
CSeq: 1 ACK
Content-Length: 0
--- (7 headers 0 lines) ---
vici*CLI>
<-- SIP read from 192.168.2.118:44972:
INVITE sip:00808613425048025@192.168.1.39 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-ee65ef665d74306d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:5800@192.168.2.118:44972>
To: "00808613425048025"<sip:00808613425048025@192.168.1.39>
From: "5800"<sip:5800@192.168.1.39>;tag=2e214051
Call-ID: M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="5800",realm="asterisk",nonce="30418e0b",uri="sip:00808613425048025@192.168.1.39",response="4937df67e7b7b4f7afa9b033dd12a07e",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 368
v=0
o=- 2 2 IN IP4 192.168.2.118
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.2.118
t=0 0
m=audio 11302 RTP/AVP 107 0 8 101
a=alt:1 3 : I6VRe4O8 Pmp6KEag 192.168.2.118 11302
a=alt:2 2 : XVR3zCxG p6kaXWej 192.168.11.1 11302
a=alt:3 1 : cN0dDlKM ZCUmHpUD 192.168.252.1 11302
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
--- (13 headers 13 lines) ---
Using INVITE request as basis request - M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.
Sending to 192.168.2.118 : 44972 (NAT)
Found user '5800'
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.118:11302
Found description format BV32
Found description format telephone-event
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 00808613425048025 in default (domain 192.168.1.39)
list_route: hop: <sip:5800@192.168.2.118:44972>
Transmitting (NAT) to 192.168.2.118:44972:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-ee65ef665d74306d-1---d8754z-;received=192.168.2.118;rport=44972
From: "5800"<sip:5800@192.168.1.39>;tag=2e214051
To: "00808613425048025"<sip:00808613425048025@192.168.1.39>
Call-ID: M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:00808613425048025@192.168.1.39>
Content-Length: 0
---
-- Executing Dial("SIP/5800-b7a69940", "Zap/g1/00808613425048025|30|tTo") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/00808613425048025
We're at 192.168.1.39 port 15840
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to 192.168.2.118:44972:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-ee65ef665d74306d-1---d8754z-;received=192.168.2.118;rport=44972
From: "5800"<sip:5800@192.168.1.39>;tag=2e214051
To: "00808613425048025"<sip:00808613425048025@192.168.1.39>;tag=as671c3891
Call-ID: M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:00808613425048025@192.168.1.39>
Content-Type: application/sdp
Content-Length: 214
v=0
o=root 2297 2297 IN IP4 192.168.1.39
s=session
c=IN IP4 192.168.1.39
t=0 0
m=audio 15840 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
== Manager 'sendcron' logged off 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
vici*CLI>
<-- SIP read from 192.168.2.118:44972:
--- (0 headers 1 lines) ---
-- Zap/1-1 is ringing
== Manager 'sendcron' logged off from 127.0.0.1
-- Nobody picked up in 30000 ms
-- Hungup 'Zap/1-1'
-- Executing Hangup("SIP/5800-b7a69940", "") in new stack
== Spawn extension (default, 00808613425048025, 2) exited non-zero on 'SIP/5800-b7a69940'
-- Executing DeadAGI("SIP/5800-b7a69940", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----NOANSWER----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----NOANSWER---------- completed, returning 0
Scheduling destruction of call 'M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.' in 32000 ms
Reliably Transmitting (NAT) to 192.168.2.118:44972:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-ee65ef665d74306d-1---d8754z-;received=192.168.2.118;rport=44972
From: "5800"<sip:5800@192.168.1.39>;tag=2e214051
To: "00808613425048025"<sip:00808613425048025@192.168.1.39>;tag=as671c3891
Call-ID: M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:00808613425048025@192.168.1.39>
Content-Length: 0
---
vici*CLI>
<-- SIP read from 192.168.2.118:44972:
ACK sip:00808613425048025@192.168.1.39 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-ee65ef665d74306d-1---d8754z-;rport
To: "00808613425048025"<sip:00808613425048025@192.168.1.39>;tag=as671c3891
From: "5800"<sip:5800@192.168.1.39>;tag=2e214051
Call-ID: M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.
CSeq: 2 ACK
Content-Length: 0
--- (7 headers 0 lines) ---
vici*CLI>
<-- SIP read from 192.168.2.118:44972:
--- (0 headers 1 lines) ---
== 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
Destroying call 'M2I0NDBhYjBkODk4ZDFjZTFkODBlZmEyN2JiMzVmYWQ.'
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
vici*CLI>
<-- SIP read from 192.168.2.118:44972:
--- (0 headers 1 lines) ---
vici*CLI>
[root@vici ~]#
- Code: Select all
[root@vici ~]# asterisk -vvvvvvvvvvvvvr
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.30.2, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.30.2 currently running on vici (pid = 2297)
Verbosity is at least 21
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
vici*CLI>
<-- SIP read from 192.168.2.118:44972:
INVITE sip:00808607563889788@192.168.1.39 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-f02f3372ab408d4f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:5800@192.168.2.118:44972>
To: "00808607563889788"<sip:00808607563889788@192.168.1.39>
From: "5800"<sip:5800@192.168.1.39>;tag=1f12cd00
Call-ID: ODU2YTZlOTE2ZmY4ZjFiM2M5YzcwNjMxMWIzM2I2NDc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 368
v=0
o=- 7 2 IN IP4 192.168.2.118
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.2.118
t=0 0
m=audio 30150 RTP/AVP 107 0 8 101
a=alt:1 3 : 4mNlKvNB kr5qq0ZE 192.168.2.118 30150
a=alt:2 2 : 1YwBnI00 On9HZHFu 192.168.11.1 30150
a=alt:3 1 : UNm9b2nd 88dPnUpx 192.168.252.1 30150
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
--- (12 headers 13 lines) ---
Using INVITE request as basis request - ODU2YTZlOTE2ZmY4ZjFiM2M5YzcwNjMxMWIzM2I2NDc.
Sending to 192.168.2.118 : 44972 (NAT)
Reliably Transmitting (NAT) to 192.168.2.118:44972:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-f02f3372ab408d4f-1---d8754z-;received=192.168.2.118;rport=44972
From: "5800"<sip:5800@192.168.1.39>;tag=1f12cd00
To: "00808607563889788"<sip:00808607563889788@192.168.1.39>;tag=as1e6b05f9
Call-ID: ODU2YTZlOTE2ZmY4ZjFiM2M5YzcwNjMxMWIzM2I2NDc.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d5ecab3"
Content-Length: 0
---
Scheduling destruction of call 'ODU2YTZlOTE2ZmY4ZjFiM2M5YzcwNjMxMWIzM2I2NDc.' in 15000 ms
Found user '5800'
vici*CLI>
<-- SIP read from 192.168.2.118:44972:
ACK sip:00808607563889788@192.168.1.39 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-f02f3372ab408d4f-1---d8754z-;rport
To: "00808607563889788"<sip:00808607563889788@192.168.1.39>;tag=as1e6b05f9
From: "5800"<sip:5800@192.168.1.39>;tag=1f12cd00
Call-ID: ODU2YTZlOTE2ZmY4ZjFiM2M5YzcwNjMxMWIzM2I2NDc.
CSeq: 1 ACK
Content-Length: 0
--- (7 headers 0 lines) ---
vici*CLI>
<-- SIP read from 192.168.2.118:44972:
INVITE sip:00808607563889788@192.168.1.39 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-c4344c6ec86cc240-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:5800@192.168.2.118:44972>
To: "00808607563889788"<sip:00808607563889788@192.168.1.39>
From: "5800"<sip:5800@192.168.1.39>;tag=1f12cd00
Call-ID: ODU2YTZlOTE2ZmY4ZjFiM2M5YzcwNjMxMWIzM2I2NDc.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="5800",realm="asterisk",nonce="3d5ecab3",uri="sip:00808607563889788@192.168.1.39",response="7ae4aa52c3a05850c72539dac54e13c6",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 368
v=0
o=- 7 2 IN IP4 192.168.2.118
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.2.118
t=0 0
m=audio 30150 RTP/AVP 107 0 8 101
a=alt:1 3 : 4mNlKvNB kr5qq0ZE 192.168.2.118 30150
a=alt:2 2 : 1YwBnI00 On9HZHFu 192.168.11.1 30150
a=alt:3 1 : UNm9b2nd 88dPnUpx 192.168.252.1 30150
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
--- (13 headers 13 lines) ---
Using INVITE request as basis request - ODU2YTZlOTE2ZmY4ZjFiM2M5YzcwNjMxMWIzM2I2NDc.
Sending to 192.168.2.118 : 44972 (NAT)
Found user '5800'
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.118:30150
Found description format BV32
Found description format telephone-event
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 00808607563889788 in default (domain 192.168.1.39)
list_route: hop: <sip:5800@192.168.2.118:44972>
Transmitting (NAT) to 192.168.2.118:44972:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-c4344c6ec86cc240-1---d8754z-;received=192.168.2.118;rport=44972
From: "5800"<sip:5800@192.168.1.39>;tag=1f12cd00
To: "00808607563889788"<sip:00808607563889788@192.168.1.39>
Call-ID: ODU2YTZlOTE2ZmY4ZjFiM2M5YzcwNjMxMWIzM2I2NDc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:00808607563889788@192.168.1.39>
Content-Length: 0
---
-- Executing Dial("SIP/5800-b7a69940", "Zap/g1/00808607563889788|30|tTo") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/00808607563889788
We're at 192.168.1.39 port 10218
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to 192.168.2.118:44972:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-c4344c6ec86cc240-1---d8754z-;received=192.168.2.118;rport=44972
From: "5800"<sip:5800@192.168.1.39>;tag=1f12cd00
To: "00808607563889788"<sip:00808607563889788@192.168.1.39>;tag=as2deca7d6
Call-ID: ODU2YTZlOTE2ZmY4ZjFiM2M5YzcwNjMxMWIzM2I2NDc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:00808607563889788@192.168.1.39>
Content-Type: application/sdp
Content-Length: 214
v=0
o=root 2297 2297 IN IP4 192.168.1.39
s=session
c=IN IP4 192.168.1.39
t=0 0
m=audio 10218 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
vici*CLI>
<-- SIP read from 192.168.2.118:44972:
--- (0 headers 1 lines) ---
-- Channel 0/1, span 1 got hangup request, cause 1
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/5800-b7a69940", "") in new stack
== Spawn extension (default, 00808607563889788, 2) exited non-zero on 'SIP/5800-b7a69940'
-- Executing DeadAGI("SIP/5800-b7a69940", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CHANUNAVAIL---------- completed, returning 0
Scheduling destruction of call 'ODU2YTZlOTE2ZmY4ZjFiM2M5YzcwNjMxMWIzM2I2NDc.' in 32000 ms
Reliably Transmitting (NAT) to 192.168.2.118:44972:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-c4344c6ec86cc240-1---d8754z-;received=192.168.2.118;rport=44972
From: "5800"<sip:5800@192.168.1.39>;tag=1f12cd00
To: "00808607563889788"<sip:00808607563889788@192.168.1.39>;tag=as2deca7d6
Call-ID: ODU2YTZlOTE2ZmY4ZjFiM2M5YzcwNjMxMWIzM2I2NDc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
vici*CLI>
<-- SIP read from 192.168.2.118:44972:
ACK sip:00808607563889788@192.168.1.39 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.118:44972;branch=z9hG4bK-d8754z-c4344c6ec86cc240-1---d8754z-;rport
To: "00808607563889788"<sip:00808607563889788@192.168.1.39>;tag=as2deca7d6
From: "5800"<sip:5800@192.168.1.39>;tag=1f12cd00
Call-ID: ODU2YTZlOTE2ZmY4ZjFiM2M5YzcwNjMxMWIzM2I2NDc.
CSeq: 2 ACK
Content-Length: 0
--- (7 headers 0 lines) ---
== Manager 'sendcron' logged off from 127.0.0.1
vici*CLI>
Has added HongKong in indications.conf.
indications.conf
- Code: Select all
[general]
country=hk ; default location
[hk]
description = HongKong
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 400,200,400,3000
dial = 350+440
busy = 480+620/500,0/500ring = 440+480/400,0/200,440+480/400,0/3000
congestion = 480+620/250,0/250
callwaiting = 440/300,0/10000
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
; RECORDTONE - not specified
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
If there any error or any special need noted? Thanks a lot.