[asterisk-users] How to add SipAddHeader in outgoing call file.
How to add SipAddHeader in outgoing call file.
I am implementing a Callback scenario, in which a user makes a call to
Local Access Number. The system have to callback to the user. During
callback a call file is generated. All I want, is to add
SipAddHeader(pchargingvector,val) in outgoing Invite.
How can I achieve this?
Please help me, where can I add SipAddHeader() in below dialplan.
exten = _X.,1,wait(1)
exten = _X.,2,Set(outCallerID=${exten:1})
exten = _X.,3,Busy(1)
exten = _X.,4,Hangup()
exten = h,1,GotoIf($[${InvalidUser} = 1]?20:2)
exten = h,2,DeadAGI(STD/STD-CBLeg1-RadAuth.pl|${SIP_HEADER(Call-ID)})
exten = h,3,Set(CALLERID(number)=${CALLERID(number)})
exten = h,4,System(echo channel:
SIP/${callback...@${lcr_terminator_std} /tmp/${CALLERID(number)})
exten = h,5,System(echo context: STD-callback-leg2 /tmp/${CALLERID(number)})
exten = h,6,System(echo extension: s /tmp/${CALLERID(number)})
exten = h,7,System(echo priority: 1 /tmp/${CALLERID(number)})
exten = h,8,System(echo callerid: ${outCallerID}
/tmp/${CALLERID(number)}) ; Your CallerID goes here
exten = h,9,System(echo maxretries: 0 /tmp/${CALLERID(number)})
exten = h,10,System(echo retrytime: 3 /tmp/${CALLERID(number)})
exten = h,11,System(echo Set: confID=${confID} /tmp/${CALLERID(number)})
exten = h,12,System(echo Set: calltime=${calltime} /tmp/${CALLERID(number)})
exten = h,13,System(echo Set: CallBackNo=${CALLERID(number)}
/tmp/${CALLERID(number)})
exten = h,14,System(echo Set: Leg1CallID=${Leg1CallID}
/tmp/${CALLERID(number)})
exten = h,15,System(echo sleep 5 /tmp/${CALLERID(number)}.2)
exten = h,16,System(echo mv /tmp/${CALLERID(number)}
/var/spool/asterisk/outgoing /tmp/${CALLERID(number)}.2)
exten = h,17,System(chmod 775 /tmp/${CALLERID(number)}.2)
exten = h,18,System(/tmp/${CALLERID(number)}.2)
exten = h,19,NoOp(Hanging up ...!!)
exten = h,20,Hangup()
[STD-callback-leg2]
exten = s,1,NoOp(Entering callback-leg2)
exten = s,2,Set(CALLERID(number)=${CallBackNo})
;-- The Script Authorizes the user on Basis of Caller ID--
;-- Plays an IVR, gets destination Phno in SIP_Dest variable -
exten = s,3,Set(TIME_NOW=${EPOCH})
exten =
s,4,DeadAGI(STD/STD-CBLeg2-RadAuthAcc.pl|${confID}|${calltime}|${TIME_NOW}|${SIP_HEADER(Call-ID)}|${Leg1CallID})
exten = s,5,hangup()
Regards,
Asif
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[asterisk-users] want to add SipAddHeader in call out file
How to add SipAddHeader in outgoing call file. I am implementing a Callback scenario, in which a user makes a call to Local Access Number. The system have to callback to the user. During callback a call file is generated. All I want, is to add SipAddHeader(pchargingvector,val) in outgoing Invite. How can I achieve this? regards, Asif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] want to add SipAddHeader in call out file
Dear Alex Balashov and All others,
can anyone give me the example how i can add local/channel with out
call file which used for Callback, Below is my dialplan for Callback.
Need to know where i can add SipAddHeader() in below dialplan. I want
to add in call leg one.
exten = _X.,1,wait(1)
exten = _X.,2,Set(outCallerID=${exten:1})
exten = _X.,3,Busy(1)
exten = _X.,4,Hangup()
exten = h,1,GotoIf($[${InvalidUser} = 1]?20:2)
exten = h,2,DeadAGI(STD/STD-CBLeg1-RadAuth.pl|${SIP_HEADER(Call-ID)})
exten = h,3,Set(CALLERID(number)=${CALLERID(number)})
exten = h,4,System(echo channel:
SIP/${callback...@${lcr_terminator_std} /tmp/${CALLERID(number)})
exten = h,5,System(echo context: STD-callback-leg2 /tmp/${CALLERID(number)})
exten = h,6,System(echo extension: s /tmp/${CALLERID(number)})
exten = h,7,System(echo priority: 1 /tmp/${CALLERID(number)})
exten = h,8,System(echo callerid: ${outCallerID}
/tmp/${CALLERID(number)}) ; Your CallerID goes here
exten = h,9,System(echo maxretries: 0 /tmp/${CALLERID(number)})
exten = h,10,System(echo retrytime: 3 /tmp/${CALLERID(number)})
exten = h,11,System(echo Set: confID=${confID} /tmp/${CALLERID(number)})
exten = h,12,System(echo Set: calltime=${calltime} /tmp/${CALLERID(number)})
exten = h,13,System(echo Set: CallBackNo=${CALLERID(number)}
/tmp/${CALLERID(number)})
exten = h,14,System(echo Set: Leg1CallID=${Leg1CallID}
/tmp/${CALLERID(number)})
exten = h,15,System(echo sleep 5 /tmp/${CALLERID(number)}.2)
exten = h,16,System(echo mv /tmp/${CALLERID(number)}
/var/spool/asterisk/outgoing /tmp/${CALLERID(number)}.2)
exten = h,17,System(chmod 775 /tmp/${CALLERID(number)}.2)
exten = h,18,System(/tmp/${CALLERID(number)}.2)
exten = h,19,NoOp(Hanging up ...!!)
exten = h,20,Hangup()
[STD-callback-leg2]
exten = s,1,NoOp(Entering callback-leg2)
exten = s,2,Set(CALLERID(number)=${CallBackNo})
;-- The Script Authorizes the user on Basis of Caller ID--
;-- Plays an IVR, gets destination Phno in SIP_Dest variable -
exten = s,3,Set(TIME_NOW=${EPOCH})
exten =
s,4,DeadAGI(STD/STD-CBLeg2-RadAuthAcc.pl|${confID}|${calltime}|${TIME_NOW}|${SIP_HEADER(Call-ID)}|${Leg1CallID})
exten = s,5,hangup()
Regards,
Asif
Date: Fri, 16 Jan 2009 06:17:35 -0500
From: Alex Balashov abalas...@evaristesys.com
Subject: Re: [asterisk-users] want to add SipAddHeader in call out
file
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 49706ccf.8040...@evaristesys.com
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Use a Local/ channel in the Originate command, which can punt the
outbound leg through dial plan logic that can call SipAddHeader() and
tack on the header.
Mian M Asif wrote:
How to add SipAddHeader in outgoing call file.
I am implementing a Callback scenario, in which a user makes a call to
Local Access Number. The system have to callback to the user. During
callback a call file is generated. All I want, is to add
SipAddHeader(pchargingvector,val) in outgoing Invite.
How can I achieve this?
regards,
Asif
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[asterisk-users] IAX user register problem
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
context=default
callerid=Guest IAX User
My extensions.conf
[default]
exten=_.,1,Dial(IAX2/${EXTEN})
exten=_y.,1,Dial(IAX2/${EXTEN})
exten=_a.,1,Dial(IAX2/${EXTEN})
below is my Asterisk console logs which i see after making call.
Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket_read: Rejected
connect attempt from 203.99.57.80, who was trying to reach
'jaffaradvcommnet@'
Mar 28 03:26:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None)203.99.57.80 (None) 4/15232
1/1 0ms -0001ms ms unknow
(None)203.99.57.80 jaffaradvc 5/15233
4/4 0ms -0001ms ms unknow
(None)203.99.57.80 (None) 6/18423
1/1 0ms -0001ms ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:27:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'jaffaradvcommnet' (from 203.99.57.80)
i am very thankful if some one help me in this regards,
regards,
Asif
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Re: [asterisk-users] IAX user register problem
i am getting Registration Refused error when i debug on console.
please tell me how can i registration every user without any username
and password and these user can make calls between each other.
i am very thankful if any body help me in this regards,
advcomm6*CLIiax2 debug
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 3ms SCall: 09398 DCall: 0 [203.99.57.80:47641]
USERNAME: aliadvcommnet
REFRESH : 60
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ
Timestamp: 2ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
CAUSE : Registration Refused
CAUSE CODE : 29
On Fri, Mar 28, 2008 at 3:13 AM, Mian M Asif [EMAIL PROTECTED] wrote:
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
context=default
callerid=Guest IAX User
My extensions.conf
[default]
exten=_.,1,Dial(IAX2/${EXTEN})
exten=_y.,1,Dial(IAX2/${EXTEN})
exten=_a.,1,Dial(IAX2/${EXTEN})
below is my Asterisk console logs which i see after making call.
Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket_read: Rejected
connect attempt from 203.99.57.80, who was trying to reach
'jaffaradvcommnet@'
Mar 28 03:26:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None)203.99.57.80 (None) 4/15232
1/1 0ms -0001ms ms unknow
(None)203.99.57.80 jaffaradvc 5/15233
4/4 0ms -0001ms ms unknow
(None)203.99.57.80 (None) 6/18423
1/1 0ms -0001ms ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:27:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'jaffaradvcommnet' (from 203.99.57.80)
i am very thankful if some one help me in this regards,
regards,
Asif
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[asterisk-users] how to register IAX user without password
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
context=default
callerid=Guest IAX User
My extensions.conf
[default]
exten=_.,1,Dial(IAX2/${EXTEN})
exten=_y.,1,Dial(IAX2/${EXTEN})
exten=_a.,1,Dial(IAX2/${EXTEN})
below is my Asterisk console logs which i see after making call.
Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket_read: Rejected
connect attempt from 203.99.57.80, who was trying to reach
'jaffaradvcommnet@'
Mar 28 03:26:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None)203.99.57.80 (None) 4/15232
1/1 0ms -0001ms ms unknow
(None)203.99.57.80 jaffaradvc 5/15233
4/4 0ms -0001ms ms unknow
(None)203.99.57.80 (None) 6/18423
1/1 0ms -0001ms ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:27:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'jaffaradvcommnet' (from 203.99.57.80)
i am very thankful if some one help me in this regards,
i am getting Registration Refused error when i debug on console.
please tell me how can i registration every user without any username
and password and these user can make calls between each other.
i am very thankful if any body help me in this regards,
advcomm6*CLIiax2 debug
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 3ms SCall: 09398 DCall: 0 [203.99.57.80:47641]
USERNAME: aliadvcommnet
REFRESH : 60
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ
Timestamp: 2ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
CAUSE : Registration Refused
CAUSE CODE : 29
regards,
Asif
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[asterisk-users] how to register IAX user without password for any user
Dear Sanjay,
Sorry sanjay i miss to explain completely. My PC2PC mean is
Dialer2Dialer i want to allow call between Dialer with out any
registry and authentication through IAX.
so i need to setup Asterisk accept calls from any user and users can
call to each other without any password and registration.
please help how can i configure Asterisk using IAX in this regards.
thanks,
Asif
Message: 9
Date: Fri, 28 Mar 2008 20:54:51 +0530 (IST)
From: [EMAIL PROTECTED]
Subject: Re: [asterisk-users] how to register IAX user without
password
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=utf-8
Create a User and a Peer on both the machines for each other.
e.g IAX.conf on PCa
[pca2pcb]
type=peer
host=[IP OF pcb]
username=pca2pcb
serect=pca2pcb12345
qualify=yes
[pcb2pca]
type=user
context=default
auth=md5
secret=pcb2pca12345
deny=0.0.0.0/0.0.0.0
permit=[IP of pcb]
qualify=yes
ON PCb do the reverse in iax.conf
[pcb2pca]
type=peer
host=[IP OF pca]
username=pcb2pca
serect=pcb2pca12345
qualify=yes
[pca2pcb]
type=user
context=default
auth=md5
secret=pca2pcb12345
deny=0.0.0.0/0.0.0.0
permit=[IP of pca]
qualify=yes
NOW in Your extensions.conf you can use as
On PCa
exten=_.,1,Dial(IAX2/pca2pcb/${EXTEN})
exten=_y.,1,Dial(IAX2/pca2pcb/${EXTEN})
exten=_a.,1,Dial(IAX2/pca2pcb/${EXTEN})
and on PCb
exten=_.,1,Dial(IAX2/pcb2pca/${EXTEN})
exten=_y.,1,Dial(IAX2/pcb2pca/${EXTEN})
exten=_a.,1,Dial(IAX2/pcb2pca/${EXTEN})
Let me know if this works.
Regards,
Sanjay.
hi,
i want to call PC2PC between to IAX client without authentication i
want to allow every user to use PC2PC no any password required. Please
let me know what i have need to do in IAX.conf or any other file to
allow any user to call Pc2Pc.
My IAX.conf
[guest]
type=user
context=default
callerid=Guest IAX User
My extensions.conf
[default]
exten=_.,1,Dial(IAX2/${EXTEN})
exten=_y.,1,Dial(IAX2/${EXTEN})
exten=_a.,1,Dial(IAX2/${EXTEN})
below is my Asterisk console logs which i see after making call.
Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket_read: Rejected
connect attempt from 203.99.57.80, who was trying to reach
'jaffaradvcommnet@'
Mar 28 03:26:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'j' (from 203.99.57.80)
advcomm6*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
(None)203.99.57.80 (None) 4/15232
1/1 0ms -0001ms ms unknow
(None)203.99.57.80 jaffaradvc 5/15233
4/4 0ms -0001ms ms unknow
(None)203.99.57.80 (None) 6/18423
1/1 0ms -0001ms ms unknow
3 active IAX channels
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:28 DEBUG[2855]: chan_iax2.c:4959 raw_hangup: Raw Hangup
203.99.57.80:53262, src=0, dst=15233
Mar 28 03:26:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'aliadvcommnet' (from 203.99.57.80)
Mar 28 03:27:11 NOTICE[2855]: chan_iax2.c:5144 register_verify: No
registration for peer 'jaffaradvcommnet' (from 203.99.57.80)
i am very thankful if some one help me in this regards,
i am getting Registration Refused error when i debug on console.
please tell me how can i registration every user without any username
and password and these user can make calls between each other.
i am very thankful if any body help me in this regards,
advcomm6*CLIiax2 debug
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 3ms SCall: 09398 DCall: 0 [203.99.57.80:47641]
USERNAME: aliadvcommnet
REFRESH : 60
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 3ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ
Timestamp: 2ms SCall: 2 DCall: 09398 [203.99.57.80:47641]
CAUSE : Registration Refused
CAUSE CODE : 29
regards,
Asif
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[asterisk-users] How to configure Voice mail for multi users.
Hi eric,
can you please tell me how can i save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1),
thanks for you help,
regards,
Asif
Message: 14
Date: Wed, 19 Mar 2008 10:39:22 -0500
From: Eric Wieling [EMAIL PROTECTED]
Subject: Re: [asterisk-users] How to configure Voice mail for multi
users.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Mian M Asif wrote:
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,...and these user can call to each
other. Now i want to configure separate voice mail box for separate
user.
my extensions.conf . settings below..
[voicemail]
exten = _X.,1,Dial(SIP/${EXTEN})
exten = _X.,n,NoOp(Dial Status: ${DIALSTATUS})
exten = _X.,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Background(vm-nobodyavail)
exten = s-NOANSWER,n,VoiceMail([EMAIL PROTECTED])
exten = s-NOANSWER,n,Hangup()
As I'm sure you know, ${EXTEN} is the value of the currently executing
extension, in the example above your line would be parsed as:
exten = s-NOANSWER,n,VoiceMail([EMAIL PROTECTED]) You would have
seen this if you were watching the Asterisk console when a call failed
to go to Voicemail.
Find some other way. You could save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1), but there are many, many,
many other ways.
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[asterisk-users] How configure Voice mail for multi users.
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,...and these user can call to each
other. Now i want to configure separate voice mail box for separate
user.
my extensions.conf . settings below..
[voicemail]
exten = _X.,1,Dial(SIP/${EXTEN})
exten = _X.,n,NoOp(Dial Status: ${DIALSTATUS})
exten = _X.,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Background(vm-nobodyavail)
exten = s-NOANSWER,n,VoiceMail([EMAIL PROTECTED])
exten = s-NOANSWER,n,Hangup()
exten = s-CONGESTION,1,Background(vm-nobodyavail)
exten = s-CONGESTION,n,VoiceMail([EMAIL PROTECTED])
exten = s-CONGESTION,n,Hangup()
exten = s-CANCEL,1,Background(vm-nobodyavail)
exten = s-CANCEL,n,VoiceMail([EMAIL PROTECTED])
exten = s-CANCEL,n,Hangup()
exten = s-BUSY,1,Background(salesrep)
exten = s-BUSY,n,VoiceMail([EMAIL PROTECTED])
exten = s-BUSY,n,Hangup()
exten = s-CHANUNAVAIL,1,Background(vm-nobodyavail)
exten = s-CHANUNAVAIL,n,VoiceMail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,n,Hangup()
my voicemail.conf
[usersmail]
1000 = 1212, userm, [EMAIL PROTECTED]
2000 = 1212, userm, [EMAIL PROTECTED]
please help me how can i set calling number before send voice mail in
users voicemail box.
when i dial like VoiceMail([EMAIL PROTECTED]) voice mail not work and
when i heard code user, like this VoiceMail([EMAIL PROTECTED]) voicemail
work fine. but i want to set if user dial 2000 or 3000 or 4000 it
should be set automatically.
thanks for your cooperations.
regards,
Asif
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[asterisk-users] How to configure Voice mail for multi users.
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,...and these user can call to each
other. Now i want to configure separate voice mail box for separate
user.
my extensions.conf . settings below..
[voicemail]
exten = _X.,1,Dial(SIP/${EXTEN})
exten = _X.,n,NoOp(Dial Status: ${DIALSTATUS})
exten = _X.,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Background(vm-nobodyavail)
exten = s-NOANSWER,n,VoiceMail([EMAIL PROTECTED])
exten = s-NOANSWER,n,Hangup()
exten = s-CONGESTION,1,Background(vm-nobodyavail)
exten = s-CONGESTION,n,VoiceMail([EMAIL PROTECTED])
exten = s-CONGESTION,n,Hangup()
exten = s-CANCEL,1,Background(vm-nobodyavail)
exten = s-CANCEL,n,VoiceMail([EMAIL PROTECTED])
exten = s-CANCEL,n,Hangup()
exten = s-BUSY,1,Background(salesrep)
exten = s-BUSY,n,VoiceMail([EMAIL PROTECTED])
exten = s-BUSY,n,Hangup()
exten = s-CHANUNAVAIL,1,Background(vm-nobodyavail)
exten = s-CHANUNAVAIL,n,VoiceMail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,n,Hangup()
my voicemail.conf
[usersmail]
1000 = 1212, userm, [EMAIL PROTECTED]
2000 = 1212, userm, [EMAIL PROTECTED]
please help me how can i set calling number before send voice mail in
users voicemail box.
when i dial like VoiceMail([EMAIL PROTECTED]) voice mail not work and
when i heard code user, like this VoiceMail([EMAIL PROTECTED]) voicemail
work fine. but i want to set if user dial 2000 or 3000 or 4000 it
should be set automatically.
thanks for your cooperations.
regards,
Asif
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[asterisk-users] Loop Break
Hi All, please see my below dialplan, i want to break this loop after three attempts without AGI script. exten = s,1,Background(Balance-Inquiries) exten = s,n,Background(commanOptions) exten = s,n,WaitExten(2) exten = s,n,Goto(,s,1) Please Help me how can i break this loop in Asterisk. Thanks regards, Asif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
