Incoming Call By DID Numbers

[faisal_ashraf]

I have a little problem when i dial In-Group Number: with number specified it does not work.
My extension.conf
[default]

exten => 6467231856,1,Ringing
exten => 6467231856,2,Wait(1)
exten => 6467231856,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}-----${CALLERID}-----${CALLERIDNUM}-----${CALLERIDNAME})
exten => 6467231856,4,Answer
exten => 6467231856,5,AGI(agi-VDADcloser_inboundCIDlookup.agi,SALESLINE-----6467231856-----Closer---------------999-----1)
exten =>6467231856,6,Voicemail,u2000

on CLI nothing happends

but when i make it like this it works perfectly, I can get call on my softphone and it shows on vicidial front as well.

exten => _.,1,Ringing
exten => _.,2,Wait(1)
exten => _.,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}-----${CALLERID}-----${CALLERIDNUM}-----${CALLERIDNAME})
exten => _.,4,Answer
exten => _.,5,AGI(agi-VDADcloser_inboundCIDlookup.agi,SALESLINE-----6467231856-----Closer---------------999-----1)
exten => _.,6,Voicemail,u2000

I am confused because i will be configuring more then one in-groups with different DID's.

Question 2
Transfer settings for call :: As per manager manual says:

Transfer-Conf DTMF 1: CXFER
Transfer-Conf Number 1: CXFER4444 (My Extension)

so what will agent need to click on to transfer the call Closer, Internal, D1,D2

I configured in-group from Manager Manual Full Version

[gardo]

Check with your provider on what DID numbers they are passing to your vicidial/asterisk box. Sometimes it's not exactly the same as your DID number. Happened to me twice. Got this DID number for example 9876543210 and they were passing 9876543210XX.

[faisal_ashraf]

Well DID numbers are exactly the same as i have configured.

I have one Asterisks Box Running perfectly since last 3 years.

i am running PD Box with my local Asterisk.

[mflorell]

Post some Asterisk CLI output of a call coming in on that number.

[faisal_ashraf]

-- Executing Ringing("SIP/SIPtrunk-09463fe8", "") in new stack
-- Executing Wait("SIP/SIPtrunk-09463fe8", "1") in new stack
-- Executing AGI("SIP/SIPtrunk-09463fe8", "agi://127.0.0.1:4577/call_log--fullCID--s-----"unknown" <8667575878>-----8667575878-----unknown") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--fullCID--s-----"unknown" <8667575878>-----8667575878-----unknown completed, returning 0
-- Executing Answer("SIP/SIPtrunk-09463fe8", "") in new stack
-- Executing AGI("SIP/SIPtrunk-09463fe8", "agi-VDADcloser_inboundCIDlookup.agi|SALESLINE-----6467231856-----Closer---------------999-----1") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADcloser_inboundCIDlookup.agi
== Spawn extension (default, s, 5) exited non-zero on 'SIP/SIPtrunk-09463fe8'
-- Executing DeadAGI("SIP/SIPtrunk-09463fe8", "agi://192.168.100.179:4577/call_log") in new stack
-- AGI Script agi://192.168.100.179:4577/call_log completed, returning 0
-- Executing DeadAGI("SIP/SIPtrunk-09463fe8", "agi://192.168.100.179:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----0---------------)") in new stack
-- AGI Script agi://192.168.100.179:4577/VD_hangup--H ... ----------) completed, returning 0
-- Executing AGI("SIP/SIPtrunk-09463fe8", "agi://127.0.0.1:4577/call_log--fullCID--h-----Y100263240000000859----------Y100263240000000859") in new stack
Oct 2 06:33:18 WARNING[29100]: res_agi.c:2063 agi_exec: If you want to run AGI on hungup channels you should use DeadAGI!
== Spawn extension (default, h, 3) exited non-zero on 'SIP/SIPtrunk-09463fe8'
== 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
== Refreshing DNS lookups.

this result comes with option two config on my 1st post
I received call on my Agent Phone perfect.

but when i change it to mynumber then Asterisk refuse the call no activity on CLI at all like didn't received any call.

[mflorell]

Do you see anything before this(what you posted) on the Asterisk CLI when a call comes in?

[faisal_ashraf]

Well I see nothing else this is from start i logged in as agent then called from another phone and got call on my agent phone and on vicidial.

== Refreshing DNS lookups.
-- Registered SIP '4444' at 192.168.100.241 port 38830 expires 180
-- Saved useragent "eyeBeam release 1011w stamp 41323" for peer 4444
== 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
== 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
> Channel SIP/4444-08a115b0 was answered.
-- Executing MeetMe("SIP/4444-08a115b0", "8600051") 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
== 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
-- Executing Ringing("SIP/SIPtrunk-08a1e880", "") in new stack
-- Executing Wait("SIP/SIPtrunk-08a1e880", "1") in new stack
-- Executing AGI("SIP/SIPtrunk-08a1e880", "agi://127.0.0.1:4577/call_log--fullCID--s-----"unknown" <8667575878>-----8667575878-----unknown") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--fullCID--s-----"unknown" <8667575878>-----8667575878-----unknown completed, returning 0
-- Executing Answer("SIP/SIPtrunk-08a1e880", "") in new stack
-- Executing AGI("SIP/SIPtrunk-08a1e880", "agi-VDADcloser_inboundCIDlookup.agi|SALESLINE-----6467231856-----Closer---------------999-----1") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADcloser_inboundCIDlookup.agi
-- AGI Script agi-VDADcloser_inboundCIDlookup.agi completed, returning 0
-- Executing MeetMe("SIP/SIPtrunk-08a1e880", "8600051") in new stack
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/SIPtrunk-08a1e880'
-- Executing DeadAGI("SIP/SIPtrunk-08a1e880", "agi://192.168.100.179:4577/call_log") in new stack
-- AGI Script agi://192.168.100.179:4577/call_log completed, returning 0
-- Executing DeadAGI("SIP/SIPtrunk-08a1e880", "agi://192.168.100.179:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----0---------------)") in new stack
-- AGI Script agi://192.168.100.179:4577/VD_hangup--H ... ----------) completed, returning 0
-- Executing AGI("SIP/SIPtrunk-08a1e880", "agi://127.0.0.1:4577/call_log--fullCID--h-----Y100303705000000859----------Y100303705000000859") in new stack
Oct 3 00:37:29 WARNING[3245]: res_agi.c:2063 agi_exec: If you want to run AGI on hungup channels you should use DeadAGI!
== Spawn extension (default, h, 3) exited non-zero on 'SIP/SIPtrunk-08a1e880'
== 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

[faisal_ashraf]

Finally worked I used different methods and this one worked.

I got call.

Now i am going to configure another DID and i hope it will work.

this is how i change the configs and worked perfectly.

exten => _.6467231856,1,Ringing
exten => _.6467231856,2,Wait(1)
exten => _.6467231856,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}----
-${CALLERID}-----${CALLERIDNUM}-----${CALLERIDNAME})
exten => _.6467231856,4,Answer
exten => _.6467231856,5,AGI(agi-VDADcloser_inboundCIDlookup.agi,SALESLINE-----64
67231856-----Closer---------------999-----1)

[faisal_ashraf]

Well i thought it worked but did not.

do you guys have any solution.

I am stuck with incoming calls by DID #s

[faisal_ashraf]

Well i made IT

there was some changes i made on my local asterisk. then it worked fine.