Inbound Number is not working

[vijith]

hi,

whenever i dial to my inbound number its saying the party is not available please help me.we used vicidial manager manual to create inbound did and ingroup




i'm using GoAutoDial CE 2.1
Asterisk 1.4.39.1-vici
did 13054071579

sip.conf

register => 63233xxxx:xxxx@66.xx.xxx.xxx/siptrunking

[Switch2Voip]
username=63233xxxx
type=peer
secret=xxxxx
progressinband=never
port=5060
nat=auto
insecure=very
ignoresdpversion=yes
host=66.xx.xx.xx
dtmfmode=rfc2833
context=trunkinbound
canreinvite=no
allow=g729&g711&g723

extentions.conf

exten => _001XX.,1,Set(CALLERID(name)="ComX Design")
exten => _001XX.,2,AGI(agi://127.0.0.1:4577/call_log)
exten => _001XX.,3,Dial(SIP/${EXTEN}@Switch2Voip,,tTo)
exten => _001XX.,4,Hangup


whenever i dial to my DID number this what i'm getting message in cli

Connected to Asterisk 1.4.39.1-vici RPM by demian@goautodialcom currently runni ng on go (pid = 11094)
Verbosity is at least 3
-- Executing [13054071579@trunkinbound:1] AGI("SIP/Switch2Voip-0000000a", "a gi-DID_route.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
-- AGI Script agi-DID_route.agi completed, returning 0
== Auto fallthrough, channel 'SIP/Switch2Voip-0000000a' status is 'UNKNOWN'
-- Executing [h@trunkinbound:1] DeadAGI("SIP/Switch2Voip-0000000a", "agi://1 27.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----0 --------------- completed, returning 0









My DID setup:

DID Extension: 13054072579
DID Description: USA DID
Active: Y
DID Route: IN_GROUP
Record Call: Y
Extension: 9998811112
Extension Context: default
Voicemail Box:
Phone Extension:
Server IP: 192.168.XXX.XX
Call Menu:
User Agent:
User Unavailable Action: VOICEMAIL
User Route Settings In-Group: Design Usa
In-Group ID: Design Usa
In-Group Call Handle Method: CID
In-Group Agent Search Method: LB
In-Group List ID: 999
In-Group Campaign ID: InTest
In-Group Phone Code:1
Clean CID Number:
Filter Inbound Number: DISABLE
Filter Phone Group ID:
Filter URL:
Filter Action: IN_GROUP
Filter Extension: 9998811112
Extension Context: default
Filter Voicemail Box:
Filter Phone Extension:
Filter Server IP:
Filter Call Menu:
Filter User Agent:
Filter User Unavailable Action: VOICEMAIL
Filter User Route Settings In-Group: AGENTDIRECT-Single Agent Direct Queue
Filter In-Group ID: NONE
Filter In-Group Call Handle Method: CID
Filter In-Group Agent Search Method: LB
Filter In-Group List ID: 999
Filter In-Group Campaign ID:
Filter In-Group Phone Code: 1




go*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
Switch2Voip/6323xxxx 66.3xx.xx.xx 5060 OK (228 ms)
8020/8020 (Unspecified) D N 0 UNKNOWN
8019/8019 (Unspecified) D N 0 UNKNOWN
8018/8018 (Unspecified) D N 0 UNKNOWN
8017/8017 (Unspecified) D N 0 UNKNOWN
8016/8016 (Unspecified) D N 0 UNKNOWN
8015/8015 (Unspecified) D N 0 UNKNOWN
8014/8014 (Unspecified) D N 0 UNKNOWN
8013/8013 (Unspecified) D N 0 UNKNOWN
8012/8012 (Unspecified) D N 0 UNKNOWN
8011/8011 (Unspecified) D N 0 UNKNOWN
8010/8010 (Unspecified) D N 0 UNKNOWN
8009/8009 (Unspecified) D N 0 UNKNOWN
8008/8008 (Unspecified) D N 0 UNKNOWN
8007/8007 (Unspecified) D N 0 UNKNOWN
8006/8006 (Unspecified) D N 0 UNKNOWN
8005/8005 (Unspecified) D N 0 UNKNOWN
8004/8004 (Unspecified) D N 0 UNKNOWN
8003/8003 (Unspecified) D N 0 UNKNOWN
8002/8002 (Unspecified) D N 0 UNKNOWN
8001/8001 (Unspecified) D N 0 UNKNOWN
101/101 192.168.1.132 D N 32780 OK (118 ms)


here is sip set debug:



<--- SIP read from 66.xx.xx.x0:5060 --->
SIP/2.0 407 Unauthorized
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK263340c3;received=115.1xx.xx.xx;rport=11487
From: <sip:6323xxxxx@66.xx.xxx.xxx>;tag=as40a59a87
To: <sip:6323xxxxx@66.xx.xxx.xxx>
Call-ID: 28d24d1d23b8d6d27988cdad01399122@127.0.0.1
CSeq: 146 REGISTER
Contact: <sip:66.x.xxx.xx:5060>
Server: Net2Phone Carrier
Proxy-Authenticate: Digest realm="net2phone",nonce="BA3C458D4F28D78204E0942F51C043C0"
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Responding to challenge, registration to domain/host name 66.xx.xxx.xxx
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 66.xx.xxx.xxx:5060:
REGISTER sip:66.xx.xxx.xx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK6f6c6dca;rport
From: <sip:6323xxxxx@66.xx.xxx.xxx>;tag=as68853156
To: <6323xxxxx@66.xx.xxx.xxx>
Call-ID: 28d24d1d23b8d6d27988cdad01399122@127.0.0.1
CSeq: 147 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="6323xxxxxx", realm="net2phone", algorithm=MD5, uri="sip:66.xx.xxx.xxx", nonce="BA3C458D4F28D78204E0942F51C043C0", response="0fe27c0fca375d59c6cee3b5f89bb8e6"
Expires: 120
Contact: <sip:siptrunking@192.168.1.251>
Event: registration
Content-Length: 0


---

<--- SIP read from 66.xx.xx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK6f6c6dca;received=115.xxx.xxx.xx;rport=11487
From: <sip:6323xxxxx@66.xx.xxx.xxx>;tag=as68853156
To: <sip:6323xxxxx@66.xx.xxx.xxx>
Call-ID: 28d24d1d23b8d6d27988cdad01399122@127.0.0.1
Expires: 90
CSeq: 147 REGISTER
Contact: <sip:siptrunking@192.168.1.251>
Server: Net2Phone Carrier
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '28d24d1d23b8d6d27988cdad01399122@127.0.0.1' in 32000 ms (Method: REGISTER)
[Aug 3 18:04:22] NOTICE[11167]: chan_sip.c:13779 handle_response_register: Outbound Registration: Expiry for 66.xx.xxx.xxx is 90 sec (Scheduling reregistration in 75 s)

<--- SIP read from 192.168.1.132:32780 --->
please help me sir .no issues with outbound dialing

[williamconley]

do you have g729 installed?

you posted lots of DEBUG cli from variouis other activities, but none from the 13054071579@trunkinbound call. i'm expecting you to find that the g729 handshake fails because it's not installed or that your route to the ingroup is somehow not working. try letting the call go to 'extension' (use the default one already there). if you get the standard "this number is not in service" "goodbye" and that same CLI output to match, begin from that point trying to route the call. if you don't, then check that SIP DEBUG to see if the handshake is failing (if the two server cannot agree on a codec that's actually installed in both servers, the call will terminate at the moment audio begins).

[vijith]

thank you for your reply william
installed g729 free version


here is trunkinbound cli

-- Executing [13054071579@trunkinbound:1] AGI("SIP/Switch2Voip-00000000", "agi-DID_route.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
-- AGI Script agi-DID_route.agi completed, returning 0
== Auto fallthrough, channel 'SIP/Switch2Voip-00000000' status is 'UNKNOWN'
-- Executing [h@trunkinbound:1] DeadAGI("SIP/Switch2Voip-00000000", "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 ... ---------- completed, returning 0



if i add this setting in extentions.conf the sip phone will ring but it wont go to in group and the voice is not clear some echo is coming

;exten => 13054071579,1,Ringing
;exten => 13054071579,2,Wait(1)
;exten => 13054071579,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}-----${CALLERID(all)}-----${CALLERID(num)}-----${CALLERID(name)})
;exten => 13054071579,4,Answer
;exten => 13054071579,5,Dial(sip/8001&sip/101,,tTo)
;exten => 13054071579,6,Voicemail,u2000

[williamconley]

as i said before: you showed CLI, but your DEBUG cli was for unrelated calls. and if you installed g729 yourself, it is entirely possible you failed. but if the call ends during the handshake, this will be shown in the DEBUG for that call. and it could well be that your g729 is the flaw.

Code: Select all

show translation

is g729 listed with translation times?

[vijith]

thank you so much william

here is cli report

go*CLI> show translation
Translation times between formats (in milliseconds) for one second of d ata
Source Format (Rows) Destination Format (Columns)

g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
g723 - 3 2 2 2 2 1 3 6 - - 2 -
gsm 10 - 2 2 2 2 1 3 6 - - 2 -
ulaw 10 3 - 1 2 2 1 3 6 - - 2 -
alaw 10 3 1 - 2 2 1 3 6 - - 2 -
g726aal2 10 3 2 2 - 2 1 3 6 - - 2 -
adpcm 10 3 2 2 2 - 1 3 6 - - 2 -
slin 9 2 1 1 1 1 - 2 5 - - 1 -
lpc10 11 4 3 3 3 3 2 - 7 - - 3 -
g729 10 3 2 2 2 2 1 3 - - - 2 -
speex - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - -
g726 10 3 2 2 2 2 1 3 6 - - - -
g722 - - - - - - - - - - - - -

[williamconley]

cool. you have g729

now what about that sip debug for a failed call (instead of some random registrations)?

[vijith]

hi william,

thanks william
here is failed call debug


<------------>
-- Executing [13054071579@trunkinbound:1] AGI("SIP/Switch2Voip-00000001", "agi-DID_route.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
-- AGI Script agi-DID_route.agi completed, returning 0
== Auto fallthrough, channel 'SIP/Switch2Voip-00000001' status is 'UNKNOWN'
-- Executing [h@trunkinbound:1] DeadAGI("SIP/Switch2Voip-00000001", "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 ... ---------- completed, returning 0
Scheduling destruction of SIP dialog '20130807072412-787-143@66.33.xxx.xxx' in 24832 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 66.33.xxx.xxx:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 66.33.xxx.xxx:5060;branch=z9hG4bK-726499437-1;received=66.33.147.150
From: "COMX.DESIGN" <sip:918147635607@66.33.XXX.XXX:5060>;user=phone;tag=ccid-267700447-726499437
To: "632334xxx" <sip:13054071579@192.168.1.251>;tag=as1c32bf3d
Call-ID: 20130807072412-787-143@66.33.xxx.xxx
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>

<--- SIP read from 66.33.xxx.xxx:5060 --->
ACK sip:13054071579@192.168.1.251 SIP/2.0
Via: SIP/2.0/UDP 66.33.xxx.xxx:5060;branch=z9hG4bK-726499437-1
From: "COMX.DESIGN" <sip:918147635607@66.33.xxx.xxx:5060>;user=phone;tag=ccid-267700447-726499437
To: "632334xxxx" <sip:13054071579@192.168.1.251>;tag=as1c32bf3d
Call-ID: 20130807072412-787-143@66.33.xxx.xxx
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '5859fafe2fbdd3841ecd0adb3da1cfa1@127.0.0.1' Method: REGISTER
Really destroying SIP dialog '20130807072353-821-141@66.33.xxx.xxx' Method: ACK
Really destroying SIP dialog '20130807072412-787-143@66.33.xxx.xxx' Method: ACK
Reliably Transmitting (no NAT) to 66.33.xxx.xxx:5060:
OPTIONS sip:66.33.147.150;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK24d46fb4;rport
From: "asterisk" <sip:asterisk@192.168.1.251>;tag=as2c802a65
To: <sip:66.33.xxx.xxx;cpd=on>
Contact: <sip:asterisk@192.168.1.251>
Call-ID: 1a22ca400c4637c458954a105ba03d7f@192.168.1.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 07 Aug 2013 11:24:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---

<--- SIP read from 66.33.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK24d46fb4;received=115.115.xxx.xxx;rport=64004
From: "asterisk" <sip:asterisk@192.168.1.251>;tag=as2c802a65
To: <sip:66.33.xxx.xxx;cpd=on>
Call-ID: 1a22ca400c4637c458954a105ba03d7f@192.168.1.251
CSeq: 102 OPTIONS
Contact: <sip:66.33.xxx.xxx:5060>
Server: Net2Phone Carrier
Content-Length: 0

go*CLI>

[vijith]

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '1a22ca400c4637c458954a105ba03d7f@192.168.1.251' Method: OPTIONS
[Aug 7 16:54:59] NOTICE[3538]: chan_sip.c:8178 sip_reregister: -- Re-registration for 632334xxxx@66.33.xxx.xxx
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 66.33.xxx.xxx:5060:
REGISTER sip:66.33.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK3140ed26;rport
From: <sip:6323342xxx@66.33.xxx.xxx>;tag=as140bcb32
To: <sip:632334xxxx@66.33.xxx.xxx>
Call-ID: 5859fafe2fbdd3841ecd0adb3da1cfa1@127.0.0.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="63233xxxxx", realm="net2phone", algorithm=MD5, uri="sip:66.33.xxx.xxx", nonce="807D8CCD2EE19540453FED9E39570A08", response="19c9eac492f94f971bb6a553629c72ac"
Expires: 120
Contact: <sip:siptrunking@192.168.1.251>
Event: registration
Content-Length: 0


---

<--- SIP read from 66.33.xxx.xxx:5060 --->
SIP/2.0 407 Unauthorized
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK3140ed26;received=115.115.xxx.xxx;rport=64004
From: <sip:632334xxxx@66.33.xxx.xxx>;tag=as140bcb32
To: <sip:632334xxx@66.33.xxx.xxx>
Call-ID: 5859fafe2fbdd3841ecd0adb3da1cfa1@127.0.0.1
CSeq: 104 REGISTER
Contact: <sip:66.33.xxx.xxx:5060>
Server: Net2Phone Carrier
Proxy-Authenticate: Digest realm="net2phone",nonce="CCB876C10B463C68A957404BE6D00F0F"
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Responding to challenge, registration to domain/host name 66.33.xxx.xxx
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 66.33.xxx.xxx:5060:
REGISTER sip:66.33.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK67437095;rport
From: <sip:63233xxxx@66.33.xxx.xxx>;tag=as5a5ed300
To: <sip:63233xxxx@66.33.xxx.xxx>
Call-ID: 5859fafe2fbdd3841ecd0adb3da1cfa1@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="632334xxx", realm="net2phone", algorithm=MD5, uri="sip:66.33.xxx.xxx", nonce="CCB876C10B463C68A957404BE6D00F0F", response="f34b2931288a7330f609099ef146c6cb"
Expires: 120
Contact: <sip:siptrunking@192.168.1.251>
Event: registration
Content-Length: 0


---
<--- SIP read from 66.33.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK67437095;received=115.115.xxx.xx;rport=64004
From: <sip:632334xxxx@66.33.xxx.xxx>;tag=as5a5ed300
To: <sip:632334xxxx@66.33.xxx.xxx>
Call-ID: 5859fafe2fbdd3841ecd0adb3da1cfa1@127.0.0.1
Expires: 90
CSeq: 105 REGISTER
Contact: <sip:siptrunking@192.168.1.251>
Server: Net2Phone Carrier
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '5859fafe2fbdd3841ecd0adb3da1cfa1@127.0.0.1' in 32000 ms (Method: REGISTER)
[Aug 7 16:54:59] NOTICE[3538]: chan_sip.c:13779 handle_response_register: Outbound Registration: Expiry for 66.33.xxx.xxx is 90 sec (Scheduling reregistration in 75 s)
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

[williamconley]

you showed the end of one call that may or may not have been related and another registration failure.

perhaps you should turn off all other connections and note exactly what happens during a single call from beginning to end.

i'm mostly interested in getting a single call from INVITE to BYE.

if you are using putty to capture, you may need to use "shift page up" to get earlier output. if you set your putty "Window->Lines of scrollback" to a nice huge number (like 20000?) you'll be much more able to grab earlier output. And then there's the "Copy to clipboard" option which will allow you to drop it into notepad or word and extract the section from invite to bye!

[vijith]

hi william

sorry for the late reply .i fixed it .i hav one doubt in my goautodial there is two lan card eth0 and eth1

eth0 static ip
eht1 local ip (192.168.1.251)
eth0 i disabled in firewall. if i enable eht0 in firewall my inbound wont work
this how i did ths setting
goauto dial 2.1
centos
command mode typed setup and there is on option firewall settings inthat option enabled there is one more option customize in that trusted device eth1 selected which is my local ip (192.168.1.251).this ip only working now for agent login and soft phone settings.because we want to access our server from outside ..
thanks

[williamconley]

you should re-enable the firewall and expressly allow the IP of your VOIP carrier through the firewall along with any external agents or administrator IPs. if you leave it open for more than a few days you will be attacked.

i'm not sure, but i believe later versions of goautdial have a firewall control panel for just such an occasion.

[vijith]

thanks william,

everything works fine now.Ive one more doubt.this is my inbound extension.conf

exten => 13053962264,1,Ringing
exten => 13053962264,2,Wait(5)
exten => 13053962264,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}-----${CALLERID(all)}-----${CALLERID(num)}-----${CALLERID(name)})
exten => 13053962264,4,Answer
exten => 13053962264,5,Playback(welcome) (thank you for calling "company name" press 1 for support ,2 for latest offer)

here I'm confused ,help me on this. if customer press 1 it has to go to sip cc150 or cc159 ,if they press to 2 it has to go to offer audio


exten => 13053962264,6,Dial(sip/cc150&sip/cc149,,tTo)
exten => 13053962264,7,Voicemail,150


I'm waiting for a possible reply

thanks vijith

[williamconley]

you are building your own dialplan instead of using Vicidial.

change the "context" for the inbound carrier to "trunkinbound" and Vicidial will handle the calls for you and you can do it all in the web administration system in Vicidial.

Code: Select all

context=trunkinbound

This should be the setting for ALL carriers in the "Account Entry" field. There is never a reason for any other "context=" value. Trust me.

oh: And remove any customizations you made to extensions.conf and/or sip.conf or any other conf file manually with the sole exception of "externip" in sip.conf if you needed that to communicate with your carrier.

Download the Vicidial Manager's Manual from EFLO.net (which is the Vicidial Group's main site, the owners of this software!). Get the free version of the manual. Start at the first page and verify you have done everything on every page until it works the way you want it. Including inbound.

You would do much better to have the inbound call routed to an "On-Hook" remote agent (or two or three) to allow music on hold, failover to the next agent if they don't answer, and failover to VM if nobody ever answers.

This is what Vicidial is built for.

Otherwise you could just as well be using Asterisk and build it all yourself. Cool, but it takes months to learn all this stuff. But the Vicidial Manager's Manual will walk you through all the basic setup.

[williamconley]

I also moved your post from general to support where it belongs.

Consider looking at "Vicibox.com" for their .iso which also installs Vicidial (the actual software you are using, which has two installation methods from .iso: GoAutoDial and Vicibox). Vicibox is supported by The Vicidial Group and everyone else in the project. GoAutoDial is supported only by a select few (PoundTeam, where I work, is one of them). But in all cases, you will be installing Vicidial which is the Dialer software you want to use. Both have their good points. Both are free.

Happy hunting!

[vijith]

hi william

thanks for your reply

we dont get much call dialy maximum 2 or 3 call that is why we using own custom plan in extension.conf.


one more doubt ,in this plan
exten => 13053962264,1,Ringing
exten => 13053962264,2,Wait(5)
exten => 13053962264,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}-----${CALLERID(all)}-----${CALLERID(num)}-----${CALLERID(name)})
exten => 13053962264,4,Answer
exten => 13053962264,5,Playback(welcome) (thank you for calling "company name" press 1 for support ,2 for latest offer)
exten => 13053962264,6,Dial(sip/cc150&sip/cc149,,tTo)
exten => 13053962264,7,Voicemail,150
added two phone and configured it in xlite (cc150 and cc149) one issue is there voice are not recording.should i add any entry in extension.conf or its saving elsewhere in my system pls heplp me on this .

[vijith]

hi wlliams,

thanks i fixed all the issues and everything working fie now (firewall and recordings issues)


thanks
vijith