Call Rejected: CHANUNAVAIL Cause: 20 - Subscriber absent.

[sasukael]

Hi everyone .. im a newbie to this furom ..also in goautodial ...

I followed the instructions... set up sip/softphone(Bria Prof) able to get the call and says "youre currently the only person in this press conference" but get the error

CHANUNAVAIL Cause: 20 - Subscriber absent.


Here is my Carrier:

register =>xxxxx:xxxxx@216.94.155.229:5060/xxxxx
Acc. Ent:
[tagbinet]
disallow=all
type=peer
secret=xxxx
username=xxxx
allow=g729
allow=g723
host=216.94.155.229
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes
outboundproxy=216.94.155.229

Global String : SIPtagbinet = SIP/tagbinet
Dial plan
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${VOIPTRUNK}/${EXTEN:1},,tToR)
exten => _91NXXNXXXXXX,3,Hangup



Asterisk Version: 1.4.27.1-vici

BUILD: 100527-2211

GOautodial VERSION: 2.2.1-260



Model
Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz



CPU Speed
2.39 GHz



Cache Size
4.00 MB



And Here is the ASTERISK FLOW


<--- SIP read from 192.168.1.102:53118 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK3869c0fd;rport=5060
Contact: <sip:cc100@192.168.1.102:53118;rinstance=69c992ae34cc0faa;cpd=on>
To: <sip:cc100@192.168.1.102:53118;rinstance=69c992ae34cc0faa;cpd=on>;tag=9a70f7 09
From: "ACagcW1319446774t001"<sip:0000000000@192.168.1.199>;tag=as5c5a8cda
Call-ID: 01424305465b70227b39d50535ce560f@192.168.1.199
CSeq: 102 INVITE
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 0

<------------->
[Oct 24 16:59:38] --- (9 headers 0 lines) ---
[Oct 24 16:59:38] WARNING[11847]: chan_sip.c:3095 create_addr: No such host: Ecc oCarrier
[Oct 24 16:59:38] Really destroying SIP dialog '409ce67b0162c944760a8cf46235c713 @127.0.0.1' Method: INVITE
[Oct 24 16:59:38] WARNING[11847]: app_dial.c:1296 dial_exec_full: Unable to crea te channel of type 'SIP' (cause 20 - Unknown)
[Oct 24 16:59:38] == Everyone is busy/congested at this time (1:0/0/1)
[Oct 24 16:59:38] -- Executing [918472343399@default:3] Hangup("Local/860005 1@default-86c2,1", "") in new stack
[Oct 24 16:59:38] == Spawn extension (default, 918472343399, 3) exited non-zer o on 'Local/8600051@default-86c2,1'
[Oct 24 16:59:38] -- Executing [h@default:1] DeadAGI("Local/8600051@default- 86c2,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CH ANUNAVAIL----------") in new stack
[Oct 24 16:59:38] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0
[Oct 24 16:59:38] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-86c2,2'
[Oct 24 16:59:38] -- Executing [h@default:1] DeadAGI("Local/8600051@default- 86c2,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-------- -------") in new stack
[Oct 24 16:59:38] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----0--------------- completed, returning 0
[Oct 24 16:59:41] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 24 16:59:42]
<--- SIP read from 192.168.1.102:53118 --->
INVITE sip:0000000000@192.168.1.199 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:53118;branch=z9hG4bK-d8754z-6645e13862360e2e-1--- d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc100@192.168.1.102:53118;rinstance=69c992ae34cc0faa;cpd=on>
To: "S1110241659128600051"<sip:0000000000@192.168.1.199>;tag=as02927624
From: <sip:cc100@192.168.1.102:53118;rinstance=69c992ae34cc0faa;cpd=on>;tag=e704 6b01
Call-ID: 0f93b0e645192d1d34de64e33a94810a@192.168.1.199
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF O
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 238

v=0
o=- 1 3 IN IP4 192.168.1.102
s=CounterPath Bria Professional
c=IN IP4 0.0.0.0
t=0 0
m=audio 35538 RTP/AVP 0 3 96
a=fmtp:96 0-15
a=rtpmap:96 telephone-event/8000
a=sendonly
a=x-rtp-session-id:D243FE8D932C4F348509B77D19C9C834

<------------->
[Oct 24 16:59:42] --- (13 headers 10 lines) ---
[Oct 24 16:59:42] Sending to 192.168.1.102 : 53118 (NAT)
[Oct 24 16:59:42] Found RTP audio format 0
[Oct 24 16:59:42] Found RTP audio format 3
[Oct 24 16:59:42] Found RTP audio format 96
[Oct 24 16:59:42] Found audio description format telephone-event for ID 96
[Oct 24 16:59:42] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x6 (gsm|ulaw) /video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
[Oct 24 16:59:42] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), pee r - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Oct 24 16:59:42] Peer audio RTP is at port 0.0.0.0:35538
[Oct 24 16:59:42]
<--- Transmitting (NAT) to 192.168.1.102:53118 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102:53118;branch=z9hG4bK-d8754z-6645e13862360e2e-1--- d8754z-;received=192.168.1.102;rport=53118
From: <sip:cc100@192.168.1.102:53118;rinstance=69c992ae34cc0faa;cpd=on>;tag=e704 6b01
To: "S1110241659128600051"<sip:0000000000@192.168.1.199>;tag=as02927624
Call-ID: 0f93b0e645192d1d34de64e33a94810a@192.168.1.199
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:0000000000@192.168.1.199>
Content-Length: 0


<------------>
[Oct 24 16:59:42] Audio is at 192.168.1.199 port 14174
[Oct 24 16:59:42] Adding codec 0x4 (ulaw) to SDP
[Oct 24 16:59:42] Adding codec 0x2 (gsm) to SDP
[Oct 24 16:59:42] Adding non-codec 0x1 (telephone-event) to SDP
[Oct 24 16:59:42]
<--- Reliably Transmitting (NAT) to 192.168.1.102:53118 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:53118;branch=z9hG4bK-d8754z-6645e13862360e2e-1--- d8754z-;received=192.168.1.102;rport=53118
From: <sip:cc100@192.168.1.102:53118;rinstance=69c992ae34cc0faa;cpd=on>;tag=e704 6b01
To: "S1110241659128600051"<sip:0000000000@192.168.1.199>;tag=as02927624
Call-ID: 0f93b0e645192d1d34de64e33a94810a@192.168.1.199
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:0000000000@192.168.1.199>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 2863 2864 IN IP4 192.168.1.199
s=session
c=IN IP4 192.168.1.199
t=0 0
m=audio 14174 RTP/AVP 0 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=recvonly

<------------>
[Oct 24 16:59:42]
<--- SIP read from 192.168.1.102:53118 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK3869c0fd;rport=5060
Contact: <sip:cc100@192.168.1.102:53118;rinstance=69c992ae34cc0faa;cpd=on>
To: <sip:cc100@192.168.1.102:53118;rinstance=69c992ae34cc0faa;cpd=on>;tag=9a70f7 09
From: "ACagcW1319446774t001"<sip:0000000000@192.168.1.199>;tag=as5c5a8cda
Call-ID: 01424305465b70227b39d50535ce560f@192.168.1.199
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF O
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 244

v=0
o=- 5 2 IN IP4 192.168.1.102
s=CounterPath Bria Professional
c=IN IP4 192.168.1.102
t=0 0
m=audio 50400 RTP/AVP 0 3 96
a=fmtp:96 0-15
a=rtpmap:96 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:5EF6BA8EE1F74EBC8B2F839347E7B61F

<------------->
[Oct 24 16:59:42] --- (12 headers 10 lines) ---
[Oct 24 16:59:42] Found RTP audio format 0
[Oct 24 16:59:42] Found RTP audio format 3
[Oct 24 16:59:42] Found RTP audio format 96
[Oct 24 16:59:42] Found audio description format telephone-event for ID 96
[Oct 24 16:59:42] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x6 (gsm|ulaw) /video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
[Oct 24 16:59:42] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), pee r - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Oct 24 16:59:42] Peer audio RTP is at port 192.168.1.102:50400
[Oct 24 16:59:42] list_route: hop: <sip:cc100@192.168.1.102:53118;rinstance=69c9 92ae34cc0faa;cpd=on>
[Oct 24 16:59:42] set_destination: Parsing <sip:cc100@192.168.1.102:53118;rinsta nce=69c992ae34cc0faa;cpd=on> for address/port to send to
[Oct 24 16:59:42] set_destination: set destination to 192.168.1.102, port 53118
[Oct 24 16:59:42] Transmitting (NAT) to 192.168.1.102:53118:
ACK sip:cc100@192.168.1.102:53118;rinstance=69c992ae34cc0faa;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK09140f3f;rport
From: "ACagcW1319446774t001" <sip:0000000000@192.168.1.199>;tag=as5c5a8cda
To: <sip:cc100@192.168.1.102:53118;rinstance=69c992ae34cc0faa;cpd=on>;tag=9a70f7 09
Contact: <sip:0000000000@192.168.1.199>
Call-ID: 01424305465b70227b39d50535ce560f@192.168.1.199
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "ACagcW1319446774t001" <sip:0000000000@192.168.1.199>;privacy=o ff;screen=no
Content-Length: 0


---
[Oct 24 16:59:42] > Channel SIP/cc100-00000007 was answered.
[Oct 24 16:59:42] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 24 16:59:42] -- Executing [8600051@default:1] MeetMe("SIP/cc100-0000000 7", "8600051|F") in new stack
[Oct 24 16:59:42]
<--- SIP read from 192.168.1.102:53118 --->
ACK sip:0000000000@192.168.1.199 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:53118;branch=z9hG4bK-d8754z-f27a8460b8515a0c-1--- d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc100@192.168.1.102:53118;rinstance=69c992ae34cc0faa;cpd=on>
To: "S1110241659128600051"<sip:0000000000@192.168.1.199>;tag=as02927624
From: <sip:cc100@192.168.1.102:53118;rinstance=69c992ae34cc0faa;cpd=on>;tag=e704 6b01
Call-ID: 0f93b0e645192d1d34de64e33a94810a@192.168.1.199
CSeq: 2 ACK
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 0


<------------->
[Oct 24 16:59:42] --- (10 headers 0 lines) ---
[Oct 24 16:59:44] Really destroying SIP dialog '6e8888510161442649645dec0ec9d278 @127.0.0.1' Method: REGISTER




Can someone help me configure this ? i really appreciate any help ..... im been troubleshooting this for about two weeks now ......


I really need help ryt now

[williamconley]

Global String : SIPtagbinet = SIP/tagbinet
Dial plan
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${VOIPTRUNK}/${EXTEN:1},,tToR)
exten => _91NXXNXXXXXX,3,Hangup

mistmatch. Try:

Code: Select all

Global String : VOIPTRUNK = SIP/tagbinet

and turn off debugging for now as the errors are appearing in the direct command line without sip debug.

[sasukael]

after i do what u said ....when i try to call it say's

Call Rejected: CHANUNAVAIL
Cause: 16 - Normal call clearing

And here is the ASTERISK FLOW :

[Oct 25 13:32:35] == Parsing '/etc/asterisk/meetme.conf': [Oct 25 13:32:35] Found
[Oct 25 13:32:35] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 25 13:32:35] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Oct 25 13:32:35] Found
[Oct 25 13:32:35] -- Created MeetMe conference 1023 for conference '8600051'
[Oct 25 13:32:35] -- <SIP/cc100-0000000a> Playing 'conf-onlyperson' (language 'en')
[Oct 25 13:32:38] == Parsing '/etc/asterisk/manager.conf': [Oct 25 13:32:38] Found
[Oct 25 13:32:38] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 25 13:32:38] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-de2f,2", "8600051|F") in new stack
[Oct 25 13:32:38] > Channel Local/8600051@default-de2f,1 was answered.
[Oct 25 13:32:38] -- Executing [917138390058@default:1] AGI("Local/8600051@default-de2f,1", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 25 13:32:38] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 25 13:32:38] -- Executing [917138390058@default:2] Dial("Local/8600051@default-de2f,1", "SIP/tagbinet/17138390058||tToR") in new stack
[Oct 25 13:32:38] WARNING[20674]: chan_sip.c:3193 sip_call: No audio format found to offer. Cancelling call to 17138390058
[Oct 25 13:32:38] -- Couldn't call tagbinet/17138390058
[Oct 25 13:32:38] Scheduling destruction of SIP dialog '208e94902e5466101c53082444118d49@192.168.1.199' in 15936 ms (Method: INVITE)
[Oct 25 13:32:38] == Everyone is busy/congested at this time (0:0/0/0)
[Oct 25 13:32:38] -- Executing [917138390058@default:3] Hangup("Local/8600051@default-de2f,1", "") in new stack
[Oct 25 13:32:38] == Spawn extension (default, 917138390058, 3) exited non-zero on 'Local/8600051@default-de2f,1'
[Oct 25 13:32:38] -- Executing [h@default:1] DeadAGI("Local/8600051@default-de2f,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack
[Oct 25 13:32:38] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 25 13:32:38] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-de2f,2'
[Oct 25 13:32:38] -- Executing [h@default:1] DeadAGI("Local/8600051@default-de2f,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Oct 25 13:32:38] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 25 13:32:39] Really destroying SIP dialog 'OTE0NDVmMTc4YzE0YjYwMTkwODYyYTQzMDJmY2IxMjY.' Method: REGISTE




I always got this error even the NUMBER that been dialed is RINGABLE to the softphone not connected to the goautodial

[iulianm]

You posted also here...

SIP/tagbinet/17138390058

17138390058 is thist the number , right ? It doesn't need any 0 in front ?


Also

Code: Select all

WARNING[20674]: chan_sip.c:3193 sip_call: No audio format found to offer. Cancelling call to 17138390058

Are you sure that your provider support g729 and g723?

If you are using g729 , do you have the g729 codec installed on your asterisk server? Because if you don't have the transcoding can't be done.

[sasukael]

yes i was able to register the g729 from the asterisk ....

what so happen now .. when i try to call i always get this operator message "press 1 for english bla bla bla (i pressed 1) and then the operator says again "please enter your card number followed by the country"..

I always get this message from the operator everytime i dial a number ....


What should i do next?


Thank you for helping me this far ....


Im needing help again ..

[williamconley]

LOL. Ask your carrier. Do you have $$ on your account with them? What country are you trying to dial to/from? Ask your carrier for a dial pattern. (NXXNXXXXXX is the US domestic dial pattern).

[sasukael]

what do you mean ... what country are we trying to dial to/from? ... its of course the US or canada .... is my dial plan wrong? ....

yup i was about to ask our provider about that ... and i will feedback later ...

[sasukael]

problem solved ... there was problem registering ID's to our provider ....

thnks a lot for helping me ....



i have a question ..... i cant hear any rings but when i prompts live call ... the customer answers immediately ....

do you have any ideas???

[williamconley]

add "R" to your dial options (and google "BriStuff"). But I do not know if GoAutoDial has BriStuff or not. If it does, that should generate a ring during manual dial.

however: you should NOT be using manual dial. just so you know. you're wasting your server. you're pulling your cool new sports car with a horse. Use auto-dial. Even if 1:1 dialing.

[sasukael]

can u point me where to do that ... the adding "R" option on dial .... is it something like .conf or something?

[williamconley]

Admin->Carriers->Choose your carrier

Dialplan entry will have a line with (this is just an example):

Code: Select all

Dial(${TRUNK}/${EXTEN:1},,tTor)

add "R" to Yours:

Code: Select all

Dial(${TRUNK}/${EXTEN:1},,tTorR)

[sasukael]

yes i do what you said .... yes were able to hear rings but only few .... not all numbers will ring ...

[williamconley]

You could look up the loopback dialplan if you NEED rings (but it significantly increases the load on the server).