vicidial.org

Dear Sir

Please help me dialed Plan . outbound call is not going getting error congestion , Actually not call is not heating carrier GW , Appreciate your help .

register => 0960044:001930011@2Z.Z.163.27:5060 (please ignore IP )

[09606000044]
type=friend
context=trunkinbound
trustrpid=yes
sendrpid=yes
host=202.53.163.27
username=09606000044
fromuser=09606000044
secret=001597530011
qualify=yes
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
nat=yes


Dialed Plan:
exten => _9ZX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9ZX.,2,Dial(SIP/${EXTEN:3}@202.53.163.27,,o)
exten => _9ZX.,3,Hangup


---------------------
linux-wmvr*CLI> sip show registry
Host Username Refresh State Reg.Time
Z.Z.163.27:5060 09606000044 105 Registered Thu, 02 Jan 2014 17:49:35



Connected to Asterisk 1.4.44-vici currently running on linux-wmvr (pid = 2381)
Verbosity is at least 21
[Jan 2 17:28:04] Reliably Transmitting (NAT) to 202.53.163.27:5060:
OPTIONS sip:202.53.163.27;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.0.230:5060;branch=z9hG4bK2fa7f8cd;rport
From: "asterisk" <sip:asterisk@192.168.0.230>;tag=as509e5d84
To: <sip:202.53.163.27;cpd=on>
Contact: <sip:asterisk@192.168.0.230>
Call-ID: 77f431057a1cc3e72b2469f17e2e4c51@192.168.0.230
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 02 Jan 2014 11:28:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Jan 2 17:28:04]
<--- SIP read from X.X.163.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.230:5060;branch=z9hG4bK2fa7f8cd
From: "asterisk" <sip:asterisk@192.168.0.230:5060>;tag=as509e5d84
To: <sip:X.X.163.27;cpd=on@>
Call-Id: 77f431057a1cc3e72b2469f17e2e4c51@192.168.0.230
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
[Jan 2 17:28:04] --- (7 headers 0 lines) ---
[Jan 2 17:28:04] Really destroying SIP dialog '77f431057a1cc3e72b2469f17e2e4c51@192.168.0.230' Method: OPTIONS
[Jan 2 17:28:07] == Parsing '/etc/asterisk/manager.conf': [Jan 2 17:28:07] Found
[Jan 2 17:28:07] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 2 17:28:07] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 2 17:28:07] Really destroying SIP dialog '5b8fdd092cc97a0128ee37d232fb4867@192.168.0.230' Method: REGISTER
linux-wmvr*CLI> sip set debug ip X.X.163.27:5060
SIP Debugging Enabled for IP: X.X.163.27:5060
[Jan 2 17:28:26] NOTICE[26622]: chan_sip.c:15812 handle_request_invite: Call from '101' to extension '8801714396426' rejected because extension not found.



Asterisk 1.4.44-vici
Server : Cluster
ISO ::ViciBox.x86_64-4.0.3.iso
ViciBox Server v.4.0!

SVN version 1.6.21 (r1462351) 2052

DB Schema Version: 1362
Linux version 3.1.10-1.29-pae (geeko@buildhost) (gcc version 4.6.2 (SUSE Linux) )
#1 SMP Fri May 31 20:10:04 UTC 2013 (2529847)

Try disabling the NAT. Also its not recommended to post your original SIP details.

the number which you are dialling is not matching your dialplan

as your dialplan accept number starting with 9 and followed by Z ( 1 - 9) so first digit should be 9 and second digit should be betweek 1 to 9

exten => _9ZX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9ZX.,2,Dial(SIP/${EXTEN:3}@202.53.163.27,,o)
exten => _9ZX.,3,Hangup

and you are dialling 8801714396426 which is not starting with 9 as first digit.

[Jan 2 17:28:26] NOTICE[26622]: chan_sip.c:15812 handle_request_invite: Call from '101' to extension '8801714396426' rejected because extension not found.

Hello Sir , I am (manual dialed) dialing 98801714396426 where carrier except only 01714XXXX as Dial(SIP/${EXTEN:3} , as i think 1st 3 digit are omitting ,

My inbound campagains in manual dial prefix option =9

getting same result congestion / temporary unavailable . Actually call are not heating carrier GW.

@urmi.l » Should I remove this line "nat=yes" NB: Sip details are wrong information (edited ) ..lol

post you cli for one single full call ,without debug

also outout of

sip show peers
sip show registry
dialplan show 98801714396426@default

striker wrote:the number which you are dialling is not matching your dialplan

as your dialplan accept number starting with 9 and followed by Z ( 1 - 9) so first digit should be 9 and second digit should be betweek 1 to 9

exten => _9ZX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9ZX.,2,Dial(SIP/${EXTEN:3}@202.53.163.27,,o)
exten => _9ZX.,3,Hangup

and you are dialling 8801714396426 which is not starting with 9 as first digit.

[Jan 2 17:28:26] NOTICE[26622]: chan_sip.c:15812 handle_request_invite: Call from '101' to extension '8801714396426' rejected because extension not found.

Good Advice.

ashis103 wrote:Hello Sir , I am (manual dialed) dialing 98801714396426 where carrier except only 01714XXXX as Dial(SIP/${EXTEN:3} , as i think 1st 3 digit are omitting ,

My inbound campagains in manual dial prefix option =9

getting same result congestion / temporary unavailable . Actually call are not heating carrier GW.

Bad response, this does not address the problem directly.

You dialed 98801714396426. But 98801714396426 does not match "_9ZX." so it CANNOT succeed. This is why the calls never get to the carrier. The calls never even execute as the EXTENSION does not exist.

Think of it this way: Each "EXTEN" entry is a program which may consist of multiple lines (notice 3 lines which all start with _9ZX ... a three line extension program!). So ... There is a program called "_9ZX." which will activate if an extension is dialed that matches it. But there is no program that matches 98801714396426, so no program will execute. Striker described WHY it does not match. Since this family of programs is called "Extensions" (or the Dialplan Language in Asterisk), no program can be activated when you dial this number. Asterisk cannot execute a program to "run" with the extension you dialed, so it merely terminates the call.

Try:

exten => _9880XX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9880XX.,2,Dial(SIP/${EXTEN:3}@202.53.163.27,,o)
exten => _9880XX.,3,Hangup

Dear williamconley

I m using your advised dial plan .but problem still same . Please advise .

exten => _9880XX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9880XX.,2,Dial(SIP/${EXTEN:3}@202.53.163.27,,o)
exten => _9880XX.,3,Hangup

=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.44-vici currently running on linux-wmvr (pid = 2384)
Verbosity is at least 21
[Jan 6 10:20:38] -- Executing [98801714396426@default:1] AGI("SIP/101-00000030", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 6 10:20:38] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 6 10:20:38] -- Executing [98801714396426@default:2] Dial("SIP/101-00000030", "SIP/01714396426@192.168.0.27||o") in new stack
[Jan 6 10:20:39] -- Called 01714396426@192.168.0.27
[Jan 6 10:20:39] NOTICE[2396]: chan_sip.c:13705 handle_response_invite: Failed to authenticate on INVITE to '"101" <sip:101@192.168.0.230>;tag=as7110fdf5'
[Jan 6 10:20:39] -- SIP/192.168.0.27-00000031 is circuit-busy
[Jan 6 10:20:39] == Everyone is busy/congested at this time (1:0/1/0)
[Jan 6 10:20:39] -- Executing [98801714396426@default:3] Hangup("SIP/101-00000030", "") in new stack
[Jan 6 10:20:39] == Spawn extension (default, 98801714396426, 3) exited non-zero on 'SIP/101-00000030'
[Jan 6 10:20:39] -- Executing [h@default:1] DeadAGI("SIP/101-00000030", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
[Jan 6 10:20:39] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

Also Incoming Call not connected . Please help


[Jan 6 06:27:09] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----21-----CONGESTION---------- completed, returning 0
[Jan 6 06:27:19] NOTICE[2161]: chan_sip.c:15812 handle_request_invite: Call from '09606000044' to extension '09606000044' rejected because extension not found.

Hello .. I have changed dial Plan and carrier , Now IP 2 IP call working but when I m receive call from mobile then dialer(ext) showing dialing and then after 22 sec later disconnect auto from zoiper/dialer, no vioce
and also AUTO 2nd times getting ringing call WITH NO VOICE . sip Showing conjested but every call ringing and connected ,Please advise





[Jan 7 00:18:29] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----0-----CANCEL---------- completed, returning 0
[Jan 7 00:18:42] -- Executing [98801714396426@default:1] AGI("SIP/110-00000 01a", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 7 00:18:42] -- AGI Script agi://127.0.0.1:4577/call_log completed, ret urning 0
[Jan 7 00:18:42] -- Executing [98801714396426@default:2] Dial("SIP/110-0000 001a", "SIP/8801714396426@108.178.17.246||o") in new stack
[Jan 7 00:18:42] -- Called 8801714396426@108.178.17.246
[Jan 7 00:19:01] == Parsing '/etc/asterisk/manager.conf': [Jan 7 00:19:01] F ound
[Jan 7 00:19:01] == Parsing '/etc/asterisk/manager.conf': [Jan 7 00:19:01] F ound
[Jan 7 00:19:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 7 00:19:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 7 00:19:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 7 00:19:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 7 00:19:06] == Parsing '/etc/asterisk/manager.conf': [Jan 7 00:19:06] F ound
[Jan 7 00:19:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jan 7 00:19:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jan 7 00:19:12] WARNING[5112]: chan_sip.c:2071 retrans_pkt: Maximum retries ex ceeded on transmission 7402896554bc5d815897125a07ecd98e@202.53.163.98 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt.
[Jan 7 00:19:14] NOTICE[5112]: chan_sip.c:3265 auto_congest: Auto-congesting SI P/108.178.17.246-0000001b
[Jan 7 00:19:14] -- SIP/108.178.17.246-0000001b is circuit-busy
[Jan 7 00:19:14] == Everyone is busy/congested at this time (1:0/1/0)
[Jan 7 00:19:14] -- Executing [98801714396426@default:3] Hangup("SIP/110-00 00001a", "") in new stack
[Jan 7 00:19:14] == Spawn extension (default, 98801714396426, 3) exited non-z ero on 'SIP/110-0000001a'
[Jan 7 00:19:14] -- Executing [h@default:1] DeadAGI("SIP/110-0000001a", "ag i://127.0.0.1:4577/call_log--HVcauses-- ... ESTION---- ------") in new stack
[Jan 7 00:19:14] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----16-----CONGESTION---------- completed, returning 0
[Jan 7 00:19:46] WARNING[5112]: chan_sip.c:2071 retrans_pkt: Maximum retries exceeded on transmission 179c201d2b3d7ec45ae508a45892d081@202.53.163.98 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt.

Dear All Voice issue is solved due to firewall blocked .Thanks all