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Only can dial IDD mobile number but not dial fixed number

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Only can dial IDD mobile number but not dial fixed number

Postby unicorncn » Mon Aug 09, 2010 10:48 pm

I'm using asterisk dial out to PSTN via T1, then dial IDD via PSTN. But cannot dial fixed number, mobile number is OK. Has asked the PSTN carrier, their setting is OK.

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.
unicorncn
 
Posts: 34
Joined: Fri May 22, 2009 8:20 pm
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Postby williamconley » Tue Aug 10, 2010 4:14 pm

wow. i don't have time to read all that, maybe someone else does. but i can say this: before trying this in Vicidial, hook up a soft phone to vicidial and allow the soft phone to dial directly through your T1 through asterisk.

then contact your Carrier and find out what dial pattern you need to dial to get to land lines. After you test this successfully with the soft phone (without vicidial being involved) so you know the pattern to dial ... are you are ready to attempt to dial in vicidial.

also: if you installed "vnow" which is VicidialNOW, you did not install "vicibox", so i'm guessin that you installed VicidialNOW (but I don't know which version of the VicidialNOW .iso) and the result was VICIdial version 2.0.5 installed onto CentOS (with the kernel version you mentioned). Vicibox is Ubuntu or OpenSuSE.

_____


when you post, please post your entire configuration including (but not limited to) your installation method, vicidial version and build, asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)
Vicidial Installation and Repair, plus Hosting and Colocation
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
williamconley
 
Posts: 20019
Joined: Wed Oct 31, 2007 4:17 pm
Location: Davenport, FL (By Disney!)
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