Dialplan for uk without any prefixes from my SIP provider

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Dialplan for uk without any prefixes from my SIP provider

Postby techster » Thu Nov 26, 2009 12:17 pm

Hi Guys

Following is the config:
ViciBox 2.0.5 installed;
asterisk version: 1.2.30.4
Phone used: XLITE 4 beta@windows 7;couldnt get it to work in linux so resorted to windows
Test campaign:def campaign
Leads loaded:1000;hopper has 6 in manual dialing; 5 in tapered
User used:100
Phone: cc100/test

now the problem:
I hear "This is not a valid extension"

My vici admin carrier entry is:
Registration String: 21100XXXXXX:XXXXX@193.47.84.4:5060
Template:SIP_generic
Account entry:
[21100XXXXXX]
allow=ulaw
allow=alaw
allow=GSM
type=friend
username=21100XXXXXX
secret=XXXXX
host=dynamic
dtmfmode=rfc2833
context=trunkinbound

Protocol:SIP
Globals String: SIPtrunk=SIP/ 21100XXXXXX:XXXXX@193.47.84.4:5060
Dialplan Entry: exten => _901144XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _901144XXXXXXXXXX,2,Dial(sip/${EXTEN:1}@SIPtrunk,55,o)
exten => _901144XXXXXXXXXX,3,Hangup

In a normal circumstance using xlite as a standalone for dialing to uk I wouild dial 44YYYYYYYYYY where y=landlinenumber of 10 digits.

CLI output:
Code: Select all
active console is [dsp]
  == 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
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
       > Channel SIP/cc100-081c1868 was answered.
    -- Executing MeetMe("SIP/cc100-081c1868", "8600051|F") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '8600051'
    -- Playing 'conf-onlyperson' (language 'en')
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing MeetMe("Local/8600051@default-2cba,2", "8600051|F") in new stack
       > Channel Local/8600051@default-2cba,1 was answered.
  == Starting Local/8600051@default-2cba,1 at default,9441383882838,1 failed so falling back to exten 's'
  == Starting Local/8600051@default-2cba,1 at default,s,1 still failed so falling back to context 'default'
    -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-2cba,1
    -- Executing Playback("Local/8600051@default-2cba,1", "invalid") in new stack
    -- Playing 'invalid' (language 'en')
Nov 26 22:42:54 WARNING[9135]: file.c:1045 ast_waitstream: Unexpected control subclass '-1'
  == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-081c1868'
    -- Executing DeadAGI("SIP/cc100-081c1868", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1
  == 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
    -- Timeout on Local/8600051@default-2cba,1
  == CDR updated on Local/8600051@default-2cba,1
    -- Executing Goto("Local/8600051@default-2cba,1", "#|1") in new stack
    -- Goto (default,#,1)
    -- Executing Playback("Local/8600051@default-2cba,1", "invalid") in new stack
    -- Playing 'invalid' (language 'en')
    -- Executing Hangup("Local/8600051@default-2cba,1", "") in new stack
  == Spawn extension (default, #, 2) exited non-zero on 'Local/8600051@default-2cba,1'
    -- Executing DeadAGI("Local/8600051@default-2cba,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
    -- Hungup 'Zap/pseudo-611069140'
  == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-2cba,2'
    -- Executing DeadAGI("Local/8600051@default-2cba,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1
exserve*CLI>


SIP is registered:
Code: Select all
sip show registry
Host                            Username       Refresh State               
193.47.84.4:5060                21100XXXXXX        105 Registered         
exserve*CLI>

techster
 
Posts: 124
Joined: Tue Nov 24, 2009 7:55 am

Postby gmcust3 » Thu Nov 26, 2009 12:50 pm

9441383882838 shld be 91441383882838
GoAutoDial CE
VERSION: 2.4-309a
BUILD: 110430-1642
No other software installed on the box.
I've read the manager manual.
gmcust3
 
Posts: 1148
Joined: Sat Oct 24, 2009 1:15 pm

Postby techster » Thu Nov 26, 2009 1:02 pm

gmcust3 wrote:9441383882838 shld be 91441383882838


That's it??

I will try it in few hours when I'm back in office.

BTW, what does it signify? What are these codes for? Do we have more such codes?
techster
 
Posts: 124
Joined: Tue Nov 24, 2009 7:55 am

Postby techster » Fri Nov 27, 2009 12:28 am

techster wrote:
gmcust3 wrote:9441383882838 shld be 91441383882838


That's it??

I will try it in few hours when I'm back in office.

BTW, what does it signify? What are these codes for? Do we have more such codes?


It still says "Its not a valid extension".
techster
 
Posts: 124
Joined: Tue Nov 24, 2009 7:55 am

Few updates

Postby techster » Fri Nov 27, 2009 2:37 am

Few updates:

I Changed prefix in campaign to X from 9.

Updated dialplan as under:
exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(${SIPtrunk}/${EXTEN:1},,tTor)
exten => _X.,3,Hangup

Now,I dont get error saying bad extension but the call disconnects (Cant hear anything but cli shows:

Code: Select all
exserve*CLI> sip show registry
Host                            Username       Refresh State               
193.47.84.4:5060                21100XXXXXX        105 Registered         
exserve*CLI> console active
No device [active] exists
Nov 27 13:04:02 WARNING[7567]: chan_oss.c:356 find_desc: could not find <active>
  == 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
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
       > Channel SIP/cc100-081d6b60 was answered.
    -- Executing MeetMe("SIP/cc100-081d6b60", "8600051|F") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '8600051'
    -- Playing 'conf-onlyperson' (language 'en')
  == Manager 'sendcron' logged off 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
    -- Executing MeetMe("Local/8600051@default-6ae7,2", "8600051|F") in new stack
       > Channel Local/8600051@default-6ae7,1 was answered.
    -- Executing AGI("Local/8600051@default-6ae7,1", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing Dial("Local/8600051@default-6ae7,1", "SIP/21100XXXXXX:XXXXXX@193.47.84.4:5060/41796473347||tTor") in new stack
    -- Called 21100XXXXXX:XXXXXX@193.47.84.4:5060/41796473347
Nov 27 13:04:16 WARNING[5809]: chan_sip.c:3680 process_sdp: Unknown SDP media type in offer: video 38974 RTP/AVP 34 103 99
    -- Got SIP response 500 "Service Unavailable" back from 193.47.84.4
    -- SIP/193.47.84.4:5060/41796473347-0821be60 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Hangup("Local/8600051@default-6ae7,1", "") in new stack
  == Spawn extension (default, 441796473347, 3) exited non-zero on 'Local/8600051@default-6ae7,1'
    -- Executing DeadAGI("Local/8600051@default-6ae7,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----38-----CONGESTION----------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----38-----CONGESTION---------- completed, returning 0
  == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-6ae7,2'
    -- Executing DeadAGI("Local/8600051@default-6ae7,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Hungup 'Zap/pseudo-1555968266'
  == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-081d6b60'
    -- Executing DeadAGI("SIP/cc100-081d6b60", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
exserve*CLI>



Any heads up??
techster
 
Posts: 124
Joined: Tue Nov 24, 2009 7:55 am

Postby techster » Fri Nov 27, 2009 6:27 am

:cry: It wont just work.. It just wont work!!

I banged head for hours at stretch.. looking through manuals, forums etc etc.

Learnt few thigns about asterisk as well.

Tried many combinations but the call isn't getting transferred. I dialed using xlite normally (connecting to terrasip directly) and it works like charm but it wont work through vicidial.

I'm sure I'm missing something here.. any help? Any angel out there?? :roll:
techster
 
Posts: 124
Joined: Tue Nov 24, 2009 7:55 am

Postby gmcust3 » Fri Nov 27, 2009 6:35 am

Try this :

register =>classic:system@XXX.171.XXX.46:5060

[ISA203]
disallow=all
allow=g729
allow=g711
allow=ulaw
type=friend
username=classic
fromuser=classic
secret=system
host=XXX.171.XXX.46
dtmfmode=rfc2833

exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,2,Dial(${SIP9}/${EXTEN:2},,tTor)
exten => _9X.,3,Hangup

Under campaign, use 9144 .

Omit Phone Code:YES
GoAutoDial CE
VERSION: 2.4-309a
BUILD: 110430-1642
No other software installed on the box.
I've read the manager manual.
gmcust3
 
Posts: 1148
Joined: Sat Oct 24, 2009 1:15 pm

Postby techster » Fri Nov 27, 2009 7:00 am

gmcust3 wrote:Try this :

register =>classic:system@XXX.171.XXX.46:5060

[ISA203]
disallow=all
allow=g729
allow=g711
allow=ulaw
type=friend
username=classic
fromuser=classic
secret=system
host=XXX.171.XXX.46
dtmfmode=rfc2833

exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,2,Dial(${SIP9}/${EXTEN:2},,tTor)
exten => _9X.,3,Hangup

Under campaign, use 9144 .

Omit Phone Code:YES


in exten block you have mentioned SIP9; in my case I have global string as SIPtrunk, will use SIPtrunk?
techster
 
Posts: 124
Joined: Tue Nov 24, 2009 7:55 am

Postby techster » Fri Nov 27, 2009 7:34 am

Code: Select all
=========================================================================
Connected to Asterisk 1.2.30.4 currently running on exserve (pid = 5808)
Verbosity is at least 21
    -- Remote UNIX connection
exserve*CLI> console
active console is [dsp]
Nov 27 18:01:29 NOTICE[5823]: chan_sip.c:11518 handle_request: Unknown SIP command 'PUBLISH' from '70.168.1.5'
    -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 70.168.1.5
  == 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
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
       > Channel SIP/cc100-081a5410 was answered.
    -- Executing MeetMe("SIP/cc100-081a5410", "8600051|F") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '8600051'
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Playing 'conf-onlyperson' (language 'en')
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing MeetMe("Local/8600051@default-0849,2", "8600051|F") in new stack
       > Channel Local/8600051@default-0849,1 was answered.
    -- Executing AGI("Local/8600051@default-0849,1", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing Dial("Local/8600051@default-0849,1", "SIP/terrasip/44441314492368||tTor") in new stack
    -- Called terrasip/44441314492368
    -- SIP/terrasip-081d7550 is making progress passing it to Local/8600051@default-0849,1
    -- Got SIP response 500 "Service Unavailable" back from 193.47.84.4
    -- SIP/terrasip-081d7550 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Hangup("Local/8600051@default-0849,1", "") in new stack
  == Spawn extension (default, 9144441314492368, 3) exited non-zero on 'Local/8600051@default-0849,1'
    -- Executing DeadAGI("Local/8600051@default-0849,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----38-----CONGESTION----------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----38-----CONGESTION---------- completed, returning 0
  == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-0849,2'
    -- Executing DeadAGI("Local/8600051@default-0849,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1
  == 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
    -- Executing MeetMeAdmin("Local/55558600051@default-43c1,2", "8600051|K") in new stack
    -- Hungup 'Zap/pseudo-1967012198'
  == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-081a5410'
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Executing DeadAGI("SIP/cc100-081a5410", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
Nov 27 18:02:51 NOTICE[14374]: app_meetme.c:2210 admin_exec: Conference Number not found
    -- Executing Hangup("Local/55558600051@default-43c1,2", "") in new stack
  == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-43c1,2'
    -- Executing DeadAGI("Local/55558600051@default-43c1,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/


Still the same problem. It calls and disconnects immediately, I can hear call getting disconnected (low pitched beep on ekiga).

Anyfile you would like to view?
techster
 
Posts: 124
Joined: Tue Nov 24, 2009 7:55 am

Postby mflorell » Fri Nov 27, 2009 8:28 am

"44441314492368" that's a lot of 4's there. Are you dialing with the phone_code(there is an option to disable this in the Campaign Detail screen)
mflorell
Site Admin
 
Posts: 18406
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Postby techster » Fri Nov 27, 2009 8:35 am

Hi florell

While uploading list it gave option for country code; I added 44 back then. Should try removing it?

Or should I just put 91 instead of 9144 in dial code?
mflorell wrote:"44441314492368" that's a lot of 4's there. Are you dialing with the phone_code(there is an option to disable this in the Campaign Detail screen)
techster
 
Posts: 124
Joined: Tue Nov 24, 2009 7:55 am

Postby mflorell » Fri Nov 27, 2009 8:39 am

No, do not remove it from the database. You just want to use the option in the campaign to not send the phone_code with the dial string.
mflorell
Site Admin
 
Posts: 18406
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Postby techster » Sat Nov 28, 2009 12:23 am

:cry:
It is still same.. call disconnects...CLI:
Code: Select all
  -- Executing MeetMe("SIP/cc100-081ca7c0", "8600051|F") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '8600051'
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Playing 'conf-onlyperson' (language 'en')
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing MeetMe("Local/8600051@default-d9ff,2", "8600051|F") in new stack
       > Channel Local/8600051@default-d9ff,1 was answered.
    -- Executing AGI("Local/8600051@default-d9ff,1", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing Dial("Local/8600051@default-d9ff,1", "SIP/terrasip/441501742711||tTor") in new stack
    -- Called terrasip/441501742711
    -- SIP/terrasip-0817a2b0 is making progress passing it to Local/8600051@default-d9ff,1
  == Manager 'sendcron' logged off from 127.0.0.1
Nov 28 10:47:55 NOTICE[7728]: rtp.c:587 ast_rtp_read: Unknown RTP codec 126 received
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-d9ff,2'
    -- Executing DeadAGI("Local/8600051@default-d9ff,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
  == Spawn extension (default, 91441501742711, 2) exited non-zero on 'Local/8600051@default-d9ff,1'
    -- Executing DeadAGI("Local/8600051@default-d9ff,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL---------- completed, returning 0
  == 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
Nov 28 10:48:05 NOTICE[7728]: rtp.c:587 ast_rtp_read: Unknown RTP codec 126 received
  == 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
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing MeetMeAdmin("Local/55558600051@default-fed8,2", "8600051|K") in new stack
    -- Executing Hangup("Local/55558600051@default-fed8,2", "") in new stack
  == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-fed8,2'
    -- Executing DeadAGI("Local/55558600051@default-fed8,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
    -- Playing 'conf-kicked' (language 'en')
Nov 28 10:48:08 WARNING[5885]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x81c4a90', 9 retries!
Nov 28 10:48:08 WARNING[5885]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x81c4680', 9 retries!
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Hungup 'Zap/pseudo-1192496435'
    -- Executing DeadAGI("SIP/cc100-081ca7c0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
    -- Unregistered SIP 'cc100'
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off 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
  == 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
  == Manager 'sendcron' logged off from 127.0.0.1
exserve*CLI>



CLI says -- Playing 'conf-kicked' (language 'en')....why??? What have I done wrong?
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Postby mflorell » Sat Nov 28, 2009 1:30 am

You might need to change the Dial string that you are using, Did you configure this in the Carriers section of the web interface?
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Postby techster » Sat Nov 28, 2009 1:37 am

I did as was told in this post. I don't have to dial any prefix while dialing through xlite. So if I was to dial 1234568890 I'll just add 44(country code) to it.

mflorell wrote:You might need to change the Dial string that you are using, Did you configure this in the Carriers section of the web interface?


::edit::
Yes, I did change carriers through vicidial admin>carriers interface. Do I need to manually update it as well?
Last edited by techster on Sat Nov 28, 2009 3:38 am, edited 1 time in total.
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Postby techster » Sat Nov 28, 2009 3:29 am

I do not have any dial prefix requirement from my service provider - Terrasip. So for example if I have to dial a number I just prefix the country code and off I go...

For this reason I have defined dialplan as (Searched through forums):

Code: Select all
exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,2,Dial(${SIPtrunk}/${EXTEN:2},,tTor)
exten => _9X.,3,Hangup


I have also added 91 as a prefix to my campaign dial code.
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Postby techster » Sat Nov 28, 2009 4:40 am

I have tried using almost all the combinations; what can I do?

Interestingly, I figured out that my provider gives me an option which says "Dialplan" where I can input what my dialplan looks like.

it says:

The VoIP dialling rules are: international prefix + national prefix + destination number. But if you want to dial as you are used to it from landline, you can setup your own dialling rules. Then you don’t need to dial a national or international prefix within our own city or to another city in your country.

Example: for Germany, Berlin = +49 (0)30 12345678 please enter these settings:
International Prefix from Germany to other countries .... = 00
Country Code ................................................................. = 49
National Prefix for every city inside Germany ................ = 0
City / Area Code for Berlin ............................................. = 30

Note: If DIAL PLAN is activated, you must dial the international prefix for internal numbers (e.g. mailbox, phone services, internal calls)
[/img]
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Few Developments

Postby techster » Sat Nov 28, 2009 6:24 am

:?

CLI
Code: Select all
    -- Registered SIP 'cc100' at 70.168.1.5 port 5060 expires 3600
    -- Saved useragent "Ekiga/3.2.5" for peer cc100
Nov 28 16:46:12 NOTICE[5827]: chan_sip.c:11518 handle_request: Unknown SIP command 'PUBLISH' from '70.168.1.5'
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 70.168.1.5
  == 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
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off 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
       > Channel SIP/cc100-081e0490 was answered.
    -- Executing MeetMe("SIP/cc100-081e0490", "8600051|F") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '8600051'
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Playing 'conf-onlyperson' (language 'en')
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing MeetMe("Local/8600051@default-fb39,2", "8600051|F") in new stack
       > Channel Local/8600051@default-fb39,1 was answered.
    -- Executing AGI("Local/8600051@default-fb39,1", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing Dial("Local/8600051@default-fb39,1", "SIP/terrasip/441506418955") in new stack
    -- Called terrasip/441506418955
  == 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
    -- SIP/terrasip-081f1db0 is making progress passing it to Local/8600051@default-fb39,1
  == 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
    -- SIP/terrasip-081f1db0 answered Local/8600051@default-fb39,1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Spawn extension (default, 441506418955, 2) exited non-zero on 'Local/8600051@default-fb39,1'
    -- Executing DeadAGI("Local/8600051@default-fb39,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----60-----48") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----60-----48 completed, returning 0
  == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-fb39,2'
    -- Executing DeadAGI("Local/8600051@default-fb39,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
  == 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
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Hungup 'Zap/pseudo-112038335'
  == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-081e0490'
    -- Executing DeadAGI("SIP/cc100-081e0490", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing MeetMeAdmin("Local/55558600051@default-1716,2", "8600051|K") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
Nov 28 16:49:05 NOTICE[4950]: app_meetme.c:2210 admin_exec: Conference Number not found
    -- Executing Hangup("Local/55558600051@default-1716,2", "") in new stack
  == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-1716,2'
    -- Executing DeadAGI("Local/55558600051@default-1716,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found


What I updated, interesting and confusing:
in vicidial admin
SIP setting
Code: Select all
[terrasip]
username=21100XXXXXX
fromuser=21100XXXXXX
type=peer
secret=XXXXX
host=193.47.84.4
nat=yes
disallaw=all
allow=g729
allow=alaw
allow=ulaw


Global string: SIPtrunk=SIP/terrasip

extensions
Code: Select all
exten => _XXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXX.,2,Dial(SIP/terrasip/${EXTEN})
exten => _XXXX.,3,Hangup


Campaign dial prefix: X
Problem: I can not hear anything in ekiga..

I read about the parameters, TtoR.. could figure out t, T, R,r but not o. Why are these required?
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Postby Michael_N » Sat Nov 28, 2009 6:54 am

make a call with your softphone conneted to vicibox.

and see if the calls get tru.

and why are you using ekiga now?
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Postby techster » Sat Nov 28, 2009 6:58 am

Michael_N wrote:make a call with your softphone conneted to vicibox.

and see if the calls get tru.

and why are you using ekiga now?


:-(

It went even worse... now I cant even hear the conf message the moment I login. I used ekiga in linux and xlite in windows.

Code: Select all
exserve*CLI> console
active console is [dsp]
  == 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
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Registered SIP 'cc100' at 70.168.1.5 port 34334 expires 3600
    -- Saved useragent "X-Lite Beta release 4.0 Beta 2 stamp 55091" for peer cc100
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Got SIP response 488 "Not Acceptable Here" back from 70.168.1.5
       > Channel SIP/cc100-081c7cc8 was never answered.
Nov 28 17:27:30 WARNING[6700]: cdr.c:566 ast_cdr_disposition: Cause not handled
  == Manager 'sendcron' logged off from 127.0.0.1
exserve*CLI>
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Postby Michael_N » Sat Nov 28, 2009 7:00 am

did you install your vicibox your self?
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Postby techster » Sat Nov 28, 2009 7:02 am

Michael_N wrote:did you install your vicibox your self?


Yup.. it was working fine as in I could hear the welcome message.

I had installed vicidial from scratch install document, then I installed the server (I just had to put the CD) rest happened :-)


::EDIT::

Few packages had the problem, I went to the server room and it had few messages on the screen. Probably from the previous restart, I had done it through ssh and not being physically present.

Lesson learnt: Always restart/shutdown/start server physically :-)
Last edited by techster on Sat Nov 28, 2009 7:39 am, edited 1 time in total.
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Postby Michael_N » Sat Nov 28, 2009 7:08 am

have you considerd hireing an consultat to help you?
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Postby techster » Sat Nov 28, 2009 7:10 am

I would want to learn it as well.. that is why I am reading things day & night.. consultant would obviously do the job but what interests me is the fact that I can gain some knowledge out of it. :-)
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try this

Postby Michael_N » Sat Nov 28, 2009 7:21 am

enter this in account entry in carriers

[terrasip]
username=SIPKEY
fromuser=SIPKEY
type=peer
secret=PASSWORD
host=terrasip.net
nat=yes
disallaw=all
allow=g729
allow=alaw


and this in dialplan entry

exten => _X!,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X!,2,Dial(sip/${EXTEN}@terrasip,55,o)
exten => _X!,3,Hangup
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Solved

Postby techster » Sat Nov 28, 2009 7:37 am

Yippeee!

My first set of calls through vicidial. :lol: :D

This is awesome!

@Michael: I did not change the exten as yet, should I change it to what you said?

Also, how to I make it throw calls automatically? I used tapered with dial level of 1.2, it traferred answering machine, I disposed it and then I had to click dial next number. Can't it be automated? By automated I mean, the moment agents dispose the call, they should get next number waiting?

Second query is with regards to answering machine, will vicidial pass all answering machines to agents? Is there anyway to control it?

::Lessons learnt::
1. Check out with service/voip provider for dialplans; they might have a set pattern. If not, you can just request them to tell you how to setup their server in asterisk.
2. Check what all codecs are supported both by your provider and software. Your sip.conf/vicidial admin carrier should have entries of all the codecs which are allowed; if possible just select as few as possible.
3. Read as much as you can :-)
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Re: Solved

Postby Michael_N » Sat Nov 28, 2009 7:50 am

techster wrote:Yippeee!

My first set of calls through vicidial. :lol: :D

This is awesome!

@Michael: I did not change the exten as yet, should I change it to what you said?

Also, how to I make it throw calls automatically? I used tapered with dial level of 1.2, it traferred answering machine, I disposed it and then I had to click dial next number. Can't it be automated? By automated I mean, the moment agents dispose the call, they should get next number waiting?

Second query is with regards to answering machine, will vicidial pass all answering machines to agents? Is there anyway to control it?

::Lessons learnt::
1. Check out with service/voip provider for dialplans; they might have a set pattern. If not, you can just request them to tell you how to setup their server in asterisk.
2. Check what all codecs are supported both by your provider and software. Your sip.conf/vicidial admin carrier should have entries of all the codecs which are allowed; if possible just select as few as possible.
3. Read as much as you can :-)


you mean you press resume after you did your dispotion?
vicidial will pass answering mashines to your agents.
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Re: Solved

Postby techster » Sat Nov 28, 2009 7:53 am

It says, dial next number not resume.

::EDIT::
Removed: Figured out the answer through posts.


Michael_N wrote:you mean you press resume after you did your dispotion?
vicidial will pass answering mashines to your agents.
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Re: Solved

Postby Michael_N » Sat Nov 28, 2009 7:59 am

techster wrote:It says, dial next number not resume.

::EDIT::
Removed: Figured out the answer through posts.


Michael_N wrote:you mean you press resume after you did your dispotion?
vicidial will pass answering mashines to your agents.


post a screenshot
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updates required

Postby techster » Sat Nov 28, 2009 8:01 am

Hi Michael

Since you had been using vici for a longtime, I would like to know what to be done when it shows updates are available.. my system is asking me to update for lib, php5, etc.. should I do it?
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Re: updates required

Postby Michael_N » Sat Nov 28, 2009 8:07 am

techster wrote:Hi Michael

Since you had been using vici for a longtime, I would like to know what to be done when it shows updates are available.. my system is asking me to update for lib, php5, etc.. should I do it?


ubuntu is asking you to do that?
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Re: updates required

Postby techster » Sat Nov 28, 2009 8:10 am

Michael_N wrote:
techster wrote:Hi Michael

Since you had been using vici for a longtime, I would like to know what to be done when it shows updates are available.. my system is asking me to update for lib, php5, etc.. should I do it?


ubuntu is asking you to do that?


Yes, ubuntu it is.
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Re: updates required

Postby Michael_N » Sat Nov 28, 2009 8:12 am

techster wrote:
Michael_N wrote:
techster wrote:Hi Michael

Since you had been using vici for a longtime, I would like to know what to be done when it shows updates are available.. my system is asking me to update for lib, php5, etc.. should I do it?


ubuntu is asking you to do that?


Yes, ubuntu it is.


ok do it, you could allways reinstall if it fails
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Location: sweden

Postby techster » Sat Nov 28, 2009 8:18 am

Alright, I will backup and then do it.

Xlite is a pain, it disconnects when it is waiting for the call, then web interface says "Call again again"... is there any alternative to xlite which is more reliable?
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Postby Michael_N » Sat Nov 28, 2009 8:31 am

techster wrote:Alright, I will backup and then do it.

Xlite is a pain, it disconnects when it is waiting for the call, then web interface says "Call again again"... is there any alternative to xlite which is more reliable?


what kind of network are you using?
between client and server?
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Location: sweden

Postby techster » Sat Nov 28, 2009 8:36 am

Michael_N wrote:
techster wrote:Alright, I will backup and then do it.

Xlite is a pain, it disconnects when it is waiting for the call, then web interface says "Call again again"... is there any alternative to xlite which is more reliable?


what kind of network are you using?
between client and server?


wireless
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Postby Michael_N » Sat Nov 28, 2009 8:38 am

techster wrote:
Michael_N wrote:
techster wrote:Alright, I will backup and then do it.

Xlite is a pain, it disconnects when it is waiting for the call, then web interface says "Call again again"... is there any alternative to xlite which is more reliable?


what kind of network are you using?
between client and server?


wireless


thats why it disconnects.

change to wire.
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Finally, last 2 issues

Postby techster » Sat Nov 28, 2009 9:17 am

Finally last 2 problems

1. Ploticus; I installed it in /usr/local. Dont know what to do to get it working. What pl executable are needed to be copied?
2. AMD: Files are already present in the server installation, why is it still not detecting AM.
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Re: Finally, last 2 issues

Postby Michael_N » Sat Nov 28, 2009 9:33 am

techster wrote:Finally last 2 problems

1. Ploticus; I installed it in /usr/local. Dont know what to do to get it working. What pl executable are needed to be copied?
2. AMD: Files are already present in the server installation, why is it still not detecting AM.


1 Start a new tread about ploticus.

2 If you need good AM detection. you should look at sangoma CPA

http://www.vicidial.org/VICIDIALforum/v ... php?t=7203

http://sangoma.com/products/software_bu ... r_cpa.html
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Location: sweden

Postby techster » Sat Nov 28, 2009 9:39 am

That looks nice :-)

Alright, just to clarify, our vicidial DOES filter AM. Just that this card will increase the rate/effeciency.. is it?
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