i have installed g729 in go auto dial

[neeraj_tiwary]

I have install g729 in goauto dialEC 2.1 And it is showing like this which i have send below , is it install properly or not ????? want to know

g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
g723 - - - - - - - - - - - - -
gsm - - 2 2 2 2 1 2 5 - - 2 -
ulaw - 2 - 1 2 2 1 2 5 - - 2 -
alaw - 2 1 - 2 2 1 2 5 - - 2 -
g726aal2 - 2 2 2 - 2 1 2 5 - - 2 -
adpcm - 2 2 2 2 - 1 2 5 - - 2 -
slin - 1 1 1 1 1 - 1 4 - - 1 -
lpc10 - 2 2 2 2 2 1 - 5 - - 2 -
g729 - 2 2 2 2 2 1 2 - - - 2 -
speex - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - -
g726 - 2 2 2 2 2 1 2 5 - - - -
g722 - - - - - - - - - - - - -

[neeraj_tiwary]

Also when i am dialing number for Uk show

call_log") in new stack
[Mar 23 19:49:37] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Mar 23 19:49:37] -- Executing [9442070180189@default:2] Dial("Local/9442070180189@default-136d,2", "SIP/442070180189@77.72.174.129||tTo") in new stack
[Mar 23 19:49:37] -- Called 442070180189@77.72.174.129
[Mar 23 19:49:38] NOTICE[2927]: chan_sip.c:13470 handle_response_invite: Failed to authenticate on INVITE to '"V3231949370000000060" <sip:0000000000@192.168.1.2>;tag=as0d6b6c3d'
[Mar 23 19:49:38] -- SIP/77.72.174.129-0000002f is circuit-busy
[Mar 23 19:49:38] == Everyone is busy/congested at this time (1:0/1/0)
[Mar 23 19:49:38] -- Executing [9442070180189@default:3] Hangup("Local/9442070180189@default-136d,2", "") in new stack
[Mar 23 19:49:38] == Spawn extension (default, 9442070180189, 3) exited non-zero on 'Local/9442070180189@default-136d,2'
[Mar 23 19:49:38] -- Executing [h@default:1] DeadAGI("Local/9442070180189@default-136d,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
[Mar 23 19:49:39] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------

[ambiorixg12]

This is not a codec issue, it is Authencation issue on your SIP trunk.

[Mar 23 19:49:38] NOTICE[2927]: chan_sip.c:13470 handle_response_invite: Failed to authenticate on INVITE to '"V3231949370000000060" <sip:0000000000@192.168.1.2>;tag=as0d6b6c3d'

related to the g729 codec run the core show translation and check if g729 is listed - But the best way to know if is working fine is

disallow=all
allow=g729

[neeraj_tiwary]

Thank you so much for reply ,,

i am using eyebeam

[b][b]Registration String: register =>neeraj_tiwary*mevoip:tiwaryxxx@77.72.174.129:5060

Account Entry: [SIPtrunk]
disallow=all
allow=g729
type=friend
username=neeraj_tiwary*mevoip
fromuser=neeraj_tiwary*mevoip
secret=tiwaryxxx
host=77.72.174.129
dtmfmode=rfc2833
callerid=test
nat=yes
Dialplan Entry
exten => _944.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _944.,2,Dial(SIP/${EXTEN:1}@77.72.174.129,,tTo)
exten => _944.,3,Hangup[/b][/b]


this viop provide me option to call all over the globe so can you check that i have entered correctly or not and can you guide me plz .

180268@default-fabf,2", "SIP/442070180268@77.72.174.129||tTo") in new stack
[Mar 24 12:17:46] -- Called 442070180268@77.72.174.129
[Mar 24 12:17:46] NOTICE[2848]: chan_sip.c:13470 handle_response_invite: Failed to authenticate on INVITE to '"V3241217460000000139" <sip:0000000000@192.168.1.2>;tag=as4a3dcf90'
[Mar 24 12:17:46] -- SIP/77.72.174.129-00000008 is circuit-busy
[Mar 24 12:17:46] == Everyone is busy/congested at this time (1:0/1/0)
[Mar 24 12:17:46] -- Executing [9442070180268@default:3] Hangup("Local/9442070180268@default-fabf,2", "") in new stack
[Mar 24 12:17:46] == Spawn extension (default, 9442070180268, 3) exited non-zero on 'Local/9442070180268@default-fabf,2'
[Mar 24 12:17:46] -- Executing [h@default:1] DeadAGI("Local/9442070180268@default-fabf,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
[Mar 24 12:17:47] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Mar 24 12:17:49] == Manager 'sendcron' logged off from 127.0.0.1

[neeraj_tiwary]

i have did all thing but not receiving call ,

[neeraj_tiwary]

[Mar 24 12:16:21] == Parsing '/etc/asterisk/manager.conf': [Mar 24 12:16:21] F ound
[Mar 24 12:16:21] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 24 12:16:21] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/8002-00000000'
[Mar 24 12:16:21] -- Executing [h@default:1] DeadAGI("SIP/8002-00000000", "a gi://127.0.0.1:4577/call_log--HVcauses- ... ----------") in new stack
[Mar 24 12:16:21] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----16--------------- completed, returning 0
[Mar 24 12:16:21] == Parsing '/etc/asterisk/manager.conf': [Mar 24 12:16:21] F ound
[Mar 24 12:16:21] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 24 12:16:21] -- Executing [55558600051@default:1] MeetMeAdmin("Local/55 558600051@default-150c,2", "8600051|K") in new stack
[Mar 24 12:16:21] WARNING[15958]: app_meetme.c:3134 admin_exec: Conference numbe r '8600051' not found!
[Mar 24 12:16:21] -- Executing [55558600051@default:2] Hangup("Local/5555860 0051@default-150c,2", "") in new stack
[Mar 24 12:16:21] == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-150c,2'
[Mar 24 12:16:21] -- Executing [h@default:1] DeadAGI("Local/55558600051@defa ult-150c,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--- ------------") in new stack
[Mar 24 12:16:21] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----16--------------- completed, returning 0
[Mar 24 12:16:23] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 12:16:23] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 12:16:23] == Parsing '/etc/asterisk/manager.conf': [Mar 24 12:16:23] F ound
[Mar 24 12:16:23] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 24 12:16:27] > Channel SIP/8002-00000006 was answered.
[Mar 24 12:16:27] -- Executing [8600051@default:1] MeetMe("SIP/8002-00000006 ", "8600051|F") in new stack
[Mar 24 12:16:27] == Parsing '/etc/asterisk/meetme.conf': [Mar 24 12:16:27] Fo und
[Mar 24 12:16:27] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Mar 24 12: 16:27] Found
[Mar 24 12:16:27] -- Created MeetMe conference 1023 for conference '8600051'
[Mar 24 12:16:27] -- <SIP/8002-00000006> Playing 'conf-onlyperson' (language 'en')
[Mar 24 12:16:29] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 12:16:30] == Parsing '/etc/asterisk/manager.conf': [Mar 24 12:16:30] F ound
[Mar 24 12:16:30] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 24 12:16:30] -- Executing [9442070180267@default:1] AGI("Local/94420701 80267@default-3eef,2", "agi://127.0.0.1:4577/call_log") in new stack
[Mar 24 12:16:30] -- AGI Script agi://127.0.0.1:4577/call_log completed, ret urning 0
[Mar 24 12:16:30] -- Executing [9442070180267@default:2] Dial("Local/9442070 180267@default-3eef,2", "SIP/442070180267@77.72.174.129||tTo") in new stack
[Mar 24 12:16:34] -- Called 442070180267@77.72.174.129
[Mar 24 12:16:36] NOTICE[2848]: chan_sip.c:13470 handle_response_invite: Failed to authenticate on INVITE to '"V3241216300000000138" <sip:0000000000@192.168.1.2 >;tag=as694e777d'
[Mar 24 12:16:36] -- SIP/77.72.174.129-00000007 is circuit-busy
[Mar 24 12:16:36] == Everyone is busy/congested at this time (1:0/1/0)
[Mar 24 12:16:36] -- Executing [9442070180267@default:3] Hangup("Local/94420 70180267@default-3eef,2", "") in new stack
[Mar 24 12:16:36] == Spawn extension (default, 9442070180267, 3) exited non-ze ro on 'Local/9442070180267@default-3eef,2'
[Mar 24 12:16:36] -- Executing [h@default:1] DeadAGI("Local/9442070180267@de fault-3eef,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21- ----CONGESTION----------") in new stack
[Mar 24 12:16:37] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----21-----CONGESTION---------- completed, returning 0
[Mar 24 12:16:39] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 12:16:41] NOTICE[2848]: chan_sip.c:13822 handle_response_peerpoke: Peer 'SIPtrunk' is now Reachable. (476ms / 2000ms)
[Mar 24 12:17:01] == Parsing '/etc/asterisk/manager.conf': [Mar 24 12:17:01] F ound

[neeraj_tiwary]

VoIP Details:

Username: neeraj
Password: neerajxxx
Domain: sip.gw.digipigeon


Portal Details:
Username: nktsolution
Password: nktsolution
Portal url: control.digipigeon

[DIGIPIGEON]
type=friend
disallow=all
allow=ulaw
username=
secret=
fromuser=
host=sip.gw.digipigeon
fromdomain=sip.gw.digipigeon
dtmfmode=rfc2833
qualify=yes


[Mar 24 12:53:07] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 12:53:23] NOTICE[2848]: chan_sip.c:13822 handle_response_peerpoke: Peer '8001' is now Lagged. (2455ms / 2000ms)
[Mar 24 12:53:23] NOTICE[2848]: chan_sip.c:13822 handle_response_peerpoke: Peer '8001' is now Reachable. (4ms / 2000ms)
[Mar 24 12:54:02] == Parsing '/etc/asterisk/manager.conf': [Mar 24 12:54:02] Found
[Mar 24 12:54:02] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 24 12:54:02] == Parsing '/etc/asterisk/manager.conf': [Mar 24 12:54:02] Found
[Mar 24 12:54:02] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 24 12:54:02] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 12:54:04] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 12:54:07] == Parsing '/etc/asterisk/manager.conf': [Mar 24 12:54:07] Found
[Mar 24 12:54:07] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 24 12:54:07] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 12:54:36] == Parsing '/etc/asterisk/manager.conf': [Mar 24 12:54:36] Found
[Mar 24 12:54:36] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 24 12:54:36] NOTICE[9132]: chan_local.c:599 local_call: No such extension/context 9442070180293@default while calling Local channel
[Mar 24 12:54:36] NOTICE[9132]: channel.c:3612 __ast_request_and_dial: Unable to call channel Local/9442070180293@default
[Mar 24 12:54:38] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 12:54:56] == Refreshing DNS lookups.
[Mar 24 12:55:01] == Parsing '/etc/asterisk/manager.conf': [Mar 24 12:55:01] Found
[Mar 24 12:55:01] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 24 12:55:01] == Parsing '/etc/asterisk/manager.conf': [Mar 24 12:55:01] Found
[Mar 24 12:55:01] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 24 12:55:01] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 12:55:03] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 12:55:06] == Parsing '/etc/asterisk/manager.conf': [Mar 24 12:55:06] Found
[Mar 24 12:55:06] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 24 12:55:06] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 12:55:27] NOTICE[2848]: chan_sip.c:13822 handle_response_peerpoke: Peer '8001' is now Lagged. (2267ms / 2000ms)
[Mar 24 12:55:27] NOTICE[2848]: chan_sip.c:13822 handle_response_peerpoke: Peer '8001' is now Reachable. (14ms / 2000ms)
go*CLI>

nat=yes
insecure=invite,port
context=trunkinbound

[ambiorixg12]

Your issue is due to a wrong authentication, run this command sip show registry

[neeraj_tiwary]

i am getting Unregistered what it mean ?????????

[Mar 26 16:57:07] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 26 16:57:18] == Refreshing DNS lookups.
go*CLI> sip show registry
Host Username Refresh State Reg.Time
sip.gw.digipigeon.com:5060 neeraj 120 Unregistered
go*CLI>

[neeraj_tiwary]

Thank you for replying all the time , I want to ask few question but first i want to give some information :
I have center two call center and both of them are in different location and i want to create host dialer so that i can have server at my main center and can be manage hear only . i have 5 agent per center ,And we are looking for increase seats too.

number or total agent is 8-10 initially.
So i know that i need static ip ,(We have out bound campaign) But at what speed my internet connection must be for dial in good way , And i heard that number of agent is equal to batter dial by go auto dial .Is that correct ?????? ....how to make my server secure . And what configuration i need for calling 20 agent. And also want to know which dialer is good goauto dial or vici dial .

Hoping for brief reply .

[ambiorixg12]

I have 8 years working with Asterisk but I'm relative a newbie with vicidial. So my suggestion is you should hire a vicidial consultant, related to the Internet speed , the amount of bandwidth used will be subjected to many factors like the codecs your communication rotocol SIP or IAX. And make your server secure involve many things, like the use of tools like fail2ban, change the SIP and IAX port for a no standard port. Use secured password. etc

[neeraj_tiwary]

I like your answer in past what you send and really happy i am speaking with person who have all answer of mine .
my next question is :::::::::::::::::::::

i know that for calling data is important , I run Uk campaign but issue is that i to run data/ number which i have to dial in like :

Number Fname Lname City
2070180132
2070180132
2070180133
2070180134
2070180135
2070180136
2070180137
2070180138
2070180139
2070180140
2070180141
2070180142
2070180143
2070180144
2070180145
2070180146
2070180147
2070180148
2070180149
2070180150
2070180151
2070180152
2070180153
2070180154
2070180155
2070180156
2070180157
2070180158
2070180159
2070180160
2070180161
2070180162
2070180163
2070180164
2070180165
2070180166
2070180167
2070180168 ******************** and so on , Actually it is process requirement which i have to full fill in that case what happen , ANSWERING machine and duration of call increase and which count lots of problem , is any software which can check my row data apart from that how i can increase my call rate and decrease answering machine . I activated my answering machine deduction but it increase the length of time , but many company run same process have no issue in that that means dialer have some thing to do. And how to check my VIOP is good or not , is any method,