Re: [asterisk-users] asterisk-users Digest, Vol 81, Issue 21
Thank you Paul. I have downloaded the code. How out-of-call messaging can be configured in the Dialplan? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com Date: Thu, 07 Apr 2011 10:14:37 -0400 From: Paul Belanger pabelan...@digium.com Subject: Re: [asterisk-users] asterisk SIP MESSAGE method support To: asterisk-users@lists.digium.com Message-ID: 4d9dc6cd.1060...@digium.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 11-04-07 09:59 AM, Deka, Rajib IN MAA SL wrote: Is the following trunk has development version of out-of-call messaging capability, also what is the version of asterisk, http://svn.asterisk.org/svn/asterisk/trunk/ I don't believe the branches has been merged into trunk, you can use russellb's branch [1]. [1] http://svn.digium.com/svn/asterisk/team/russell/messaging/ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. Thank You. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk login to voicemail
Wow, thanks, that worked...
in case anyone is interested this is what i did
[voicemail]
exten = a,1,VoiceMailMain(${MAILBOXID}@${MAILBOXCONTEXT},p)
in AGI...
$AGI-set_variable(MAILBOXID, $options);
$AGI-set_variable(MAILBOXCONTEXT,4);
$AGI-set_context(voicemail);
$AGI-exec(VoiceMail, $options);
now the question is how to I get the VoiceMailMain to not ask for Mailbox
and already know which mailbox and just prompt for Password
On Thu, Apr 7, 2011 at 6:44 PM, Dan Journo d...@keshercommunications.comwrote:
Unfortunately, that solution will not work for me... The user must be
able to hit * during the greeting of any voicemail and be prompted for the
Password to that particular mailbox looks like i got a lot of
programming to do to create a work around for this... thanks for your
help...
Forgive me if i'm wrong, but you guys seem to be over complicating things.
Taken from: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
during the prompt if the caller presses:
'*' - the call jumps to extension 'a' in the current voicemail context.
*Example:*
Exten = a, 1, VoicemailMain(@default)
Exten = a, 2, Hangup
When using the star '*' it's important to note that the context you placed
the application voicemail in is irrelevant, it's the context for the
voicemail box that we're looking for in the dialplan for the jump to the 'a'
extension.
So this is what i do...
Before passing the call to the voicemail app, i set ${MAILBOXCONTEXT} to
the correct context, and i set ${MAILBOXID} to the mailbox name.
Then, in extensions.conf, I added this:-
[voicemail]
exten = a,1,Playback(astcc-please-enter-your)
exten = a,n,VoicemailMain(${MAILBOXID}@${MAILBOXCONTEXT})
When the user presses *, they are passed to the 'a' extension above and
into VoicemailMain.
I'm sure you can turn this into AGI easily enough if needed.
Dan Journo
Kesher Communications (UK)
Business Phone Systems http://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
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Re: [asterisk-users] asterisk login to voicemail
SIP/8.224.32.2-AGI Rx SET VARIABLE MAILBOXID 7167435000
SIP/8.224.32.2-AGI Tx 200 result=1
SIP/8.224.32.2-AGI Rx SET VARIABLE MAILBOXCONTEXT 4
SIP/8.224.32.2-AGI Tx 200 result=1
SIP/8.224.32.2-AGI Rx SET CONTEXT voicemail
And the mailbox 7167435000 does exist
On Fri, Apr 8, 2011 at 8:29 AM, Dan Journo d...@keshercommunications.comwrote:
now the question is how to I get the VoiceMailMain to not ask for
Mailbox and already know which mailbox and just prompt for Password
If its asking you for the mailbox, then the ${MAILBOXID}@${MAILBOXCONTEXT}
values aren't correct.
Use the CLI to debug and make sure ${MAILBOXID}@${MAILBOXCONTEXT} is
correct and the mailbox exists.
Dan Journo
Kesher Communications (UK)
Business Phone Systems http://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
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[asterisk-users] User registration failure bug ?
Hi list,
I have a user, referenced by his IMSI (IMSI208300618462231), who is
assigned to extension 2111 in /etc/asterisk/extensions.conf and
sip.conf (see below).
From time to time, registration of this user fails (see below), but I
do not know why. Anybody has a clue what could be wrong ? Is this a
bug ?
[I rebooted asterisk, and now it works.]
Regards
Axelle.
Logs of failed registration:
sip show users
Username Secret Accountcode
Def.Context ACL NAT
IMSI208011234567890 sip-local
No RFC3581
IMSI208302141472352
sip-external No RFC3581
IMSI208304424439206
sip-external No RFC3581
[Apr 8 15:01:01] NOTICE[20626]: chan_sip.c:15642
handle_request_register: Registration from 'IMSI208300618462231
sip:IMSI208300618462231@127.0.0.1' failed for '127.0.0.1' - No
matching peer found
sip show user IMSI208300618462231
User IMSI208300618462231 not found.
My configuration in extensions.conf:
[IMSI208300618462231]
callerid=2111
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
sip.conf:
exten = 2111,1,Macro(dialSIP,IMSI208300618462231)
where dialSIP is a macro:
[macro-dialSIP]
exten = s,1,Dial(SIP/${ARG1})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy(3000)
exten = s-CONGESTION,1,Congestion(3000)
exten = s-CHANUNAVAIL,1,playback(ss-noservice)
exten = s-CANCEL,2,Hangup
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[asterisk-users] Maniuplate callerID based off of callerID
Hey all!
I'm trying to figure out a way to manipulate a call's caller ID based off of
the caller's caller ID. Basically, I've got a situation where anything that
goes through an Nortel Opt11's IVR comes out with the caller ID 400 (the
Opt11's IVR's ext). When the call goes out the trunk that the call is
destined for, I'd like to grab the 400 caller ID and delete it so it comes
through as Unknown. The Unknown part doesn't have to be the literal
string, just a blank CallerID would be fine.
Would it be something like:
Exten = ExecIf($[${CALLERID(number)} = 400]?SetCallerID())
Thanks all!
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Re: [asterisk-users] MOH not working
Add exten = 6000,n,StartMusicOnHold() to the end of your current dialplan and
try again.
Thanks,
--Warren Selby, dCAP
On Apr 8, 2011, at 1:51 AM, virendra bhati virbh...@gmail.com wrote:
I am using Elastix. Asterisk is used for PBX application in Elastix. I want
to make customize MOH. But not able to use MOH. I make simple extension in
asterisk conf file but no success :(
Below are the details of configuration files.
Even default MOH is also not working
Asterisk Version 1.6.2.17.2
1) Extension.conf
[incoming]
exten = 6000,1,Answer
exten = 6000,n,Set(CHANNEL(musicclass)=BSNL)
exten = 6000,n,Set(foo=${CHANNEL(musicclass)})
exten = 6000,n,MusicOnHold(BSNL)
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Re: [asterisk-users] asterisk login to voicemail
Why are you using agi for this ? They are inbuild features of asterisk.
Or may be I am missing something
--
Sent from my iPhone
On Apr 8, 2011, at 8:26 AM, vip killa vipki...@gmail.com wrote:
Wow, thanks, that worked...
in case anyone is interested this is what i did
[voicemail]
exten = a,1,VoiceMailMain(${MAILBOXID}@${MAILBOXCONTEXT},p)
in AGI...
$AGI-set_variable(MAILBOXID, $options);
$AGI-set_variable(MAILBOXCONTEXT,4);
$AGI-set_context(voicemail);
$AGI-exec(VoiceMail, $options);
now the question is how to I get the VoiceMailMain to not ask for
Mailbox and already know which mailbox and just prompt for
Password
On Thu, Apr 7, 2011 at 6:44 PM, Dan Journo d...@keshercommunications.com
wrote:
Unfortunately, that solution will not work for me... The user must
be able to hit * during the greeting of any voicemail and be
prompted for the Password to that particular mailbox looks
like i got a lot of programming to do to create a work around for
this... thanks for your help...
Forgive me if i'm wrong, but you guys seem to be over complicating
things.
Taken from: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
during the prompt if the caller presses:
'*' - the call jumps to extension 'a' in the current voicemail
context.
Example:
Exten = a, 1, VoicemailMain(@default)
Exten = a, 2, Hangup
When using the star '*' it's important to note that the context you
placed the application voicemail in is irrelevant, it's the context
for the voicemail box that we're looking for in the dialplan for the
jump to the 'a' extension.
So this is what i do...
Before passing the call to the voicemail app, i set $
{MAILBOXCONTEXT} to the correct context, and i set ${MAILBOXID} to
the mailbox name.
Then, in extensions.conf, I added this:-
[voicemail]
exten = a,1,Playback(astcc-please-enter-your)
exten = a,n,VoicemailMain(${MAILBOXID}@${MAILBOXCONTEXT})
When the user presses *, they are passed to the 'a' extension above
and into VoicemailMain.
I'm sure you can turn this into AGI easily enough if needed.
Dan Journo
Kesher Communications (UK)
Business Phone Systems | Hosted PBX
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Re: [asterisk-users] Maniuplate callerID based off of callerID
Almost,
If you use Asterisk version 1.6 or higher use
Exten = ExecIf($[${CALLERID(number)} = 400]?Set(CALLERID(num)=)
Or
Exten = ExecIf($[${CALLERID(number)} = 400]?Set(CALLERID(all)=)
Michel Verbraak
**
http://www.intercommit.nl/
On 08-04-11 15:56, Louis Carreiro wrote:
Hey all!
I'm trying to figure out a way to manipulate a call's caller ID based
off of the caller's caller ID. Basically, I've got a situation where
anything that goes through an Nortel Opt11's IVR comes out with the
caller ID 400 (the Opt11's IVR's ext). When the call goes out the
trunk that the call is destined for, I'd like to grab the 400 caller
ID and delete it so it comes through as Unknown. The Unknown part
doesn't have to be the literal string, just a blank CallerID would be
fine.
Would it be something like:
Exten = ExecIf($[${CALLERID(number)} = 400]?SetCallerID())
Thanks all!
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[asterisk-users] IAX2/0.0.29.199
Where this revers IP comes from ? == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-006b, stdexten,7623,SIP/7623) in new stack -- Executing [s@macro-stdexten:1] ChanIsAvail(SIP/7527-006b, SIP/7623IAX2/7623,20,t) in new stack -- Hungup 'IAX2/0.0.29.199:4569-5255' -- Executing [s@macro-stdexten:2] NoOp(SIP/7527-006b, IAX2/0.0.29.199:4569-5255) in new stack -- Executing [s@macro-stdexten:3] NoOp(SIP/7527-006b, 00) in new stack -- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN' -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any PHP Ming + for Asterisk guides, tutorial, how-to anywhere?
Hi Everyone, Looking to create a flash status update page from Asterisk events using PHP MING but I can't seem to find much documentation other than those on PHP site. Has anyone ever tried or has came across PHP Ming usage for Asterisk? Any hints of much appreciated. ***I am aware of FOP using ming. Looking for something other than that. Specifically in PHP rather than PERL. Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/0.0.29.199
On 11-04-08 10:48 AM, satish patel wrote: Where this revers IP comes from ? == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-006b, stdexten,7623,SIP/7623) in new stack -- Executing [s@macro-stdexten:1] ChanIsAvail(SIP/7527-006b, SIP/7623IAX2/7623,20,t) in new stack -- Hungup 'IAX2/0.0.29.199:4569-5255' -- Executing [s@macro-stdexten:2] NoOp(SIP/7527-006b, IAX2/0.0.29.199:4569-5255) in new stack -- Executing [s@macro-stdexten:3] NoOp(SIP/7527-006b, 00) in new stack -- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN' Asterisk 1.8? Are you using realtime? Looks to be an issue with netsock2.c. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maniuplate callerID based off of callerID
Perfect! That was it! Thanks!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michel
Verbraak
Sent: Friday, April 08, 2011 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Maniuplate callerID based off of callerID
Almost,
If you use Asterisk version 1.6 or higher use
Exten = ExecIf($[${CALLERID(number)} = 400]?Set(CALLERID(num)=)
Or
Exten = ExecIf($[${CALLERID(number)} = 400]?Set(CALLERID(all)=)
Michel Verbraak
http://www.intercommit.nl/
On 08-04-11 15:56, Louis Carreiro wrote:
Hey all!
I'm trying to figure out a way to manipulate a call's caller ID
based off of the caller's caller ID. Basically, I've got a situation where
anything that goes through an Nortel Opt11's IVR comes out with the caller
ID 400 (the Opt11's IVR's ext). When the call goes out the trunk that the
call is destined for, I'd like to grab the 400 caller ID and delete it so it
comes through as Unknown. The Unknown part doesn't have to be the
literal string, just a blank CallerID would be fine.
Would it be something like:
Exten = ExecIf($[${CALLERID(number)} = 400]?SetCallerID())
Thanks all!
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Re: [asterisk-users] IAX2/0.0.29.199
No I am not using any realtime config. its text file.. shirley*CLI core show settings PBX Core settings - Version: 1.8.3.2 Build Options: LOADABLE_MODULES Maximum calls: 250 (Current 0) Maximum open file handles: Not set Verbosity: 3 Debug level: 0 Maximum load average:0.00 Minimum free memory: 0 MB Startup time:15:08:59 Last reload time:15:08:59 System: Linux/2.6.32-24-server built by root on x86_64 2011-03-22 18:38:19 UTC System name: Entity ID: 00:30:48:77:1c:3c Default language:en Language prefix: Enabled User name and group: asterisk/asterisk Executable includes: Disabled Transcode via SLIN: Enabled Internal timing: Enabled Transmit silence during rec: Disabled Generic PLC: Enabled * Subsystems - Manager (AMI): Enabled Web Manager (AMI/HTTP): Disabled Call data records: Enabled Realtime Architecture (ARA): Disabled * Directories - Configuration file: Configuration directory: /etc/asterisk Module directory:/usr/lib/asterisk/modules Spool directory: /var/spool/asterisk Log directory: /var/log/asterisk Run/Sockets directory: /var/run/asterisk PID file:/var/run/asterisk/asterisk.pid VarLib directory:/var/lib/asterisk Data directory: /var/lib/asterisk ASTDB: /var/lib/asterisk/astdb IAX2 Keys directory: /var/lib/asterisk/keys AGI Scripts directory: /var/lib/asterisk/agi-bin Date: Fri, 8 Apr 2011 11:12:59 -0400 From: pabelan...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IAX2/0.0.29.199 On 11-04-08 10:48 AM, satish patel wrote: Where this revers IP comes from ? == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-006b, stdexten,7623,SIP/7623) in new stack -- Executing [s@macro-stdexten:1] ChanIsAvail(SIP/7527-006b, SIP/7623IAX2/7623,20,t) in new stack -- Hungup 'IAX2/0.0.29.199:4569-5255' -- Executing [s@macro-stdexten:2] NoOp(SIP/7527-006b, IAX2/0.0.29.199:4569-5255) in new stack -- Executing [s@macro-stdexten:3] NoOp(SIP/7527-006b, 00) in new stack -- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN' Asterisk 1.8? Are you using realtime? Looks to be an issue with netsock2.c. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.3
From: Chris Owen ow...@hubris.net Sent: Thursday, April 07, 2011 9:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.3 Best I can tell, multi-tenant parking also hasn't worked in any of the 1.8.x releases. Chris Chris I have not been able to get multi-tenant parking stable there either. I gave up yesterday on 1.8.3.2 as I could not get it stable with any number of patches I could find. I fell back to 1.8.2.3 as that is the last version that I have been able to run production with. My customers have now been happy for the last 24 hours. I also tried 1.8.4 rc and the stability did not appear to be much better then 1.8.3.2 I hope they don't release 1.8.4 until the stability issues are addressed more rc version with fixes would be ideal. The longer these items drag out the worse it gets for users to know what to use. I would ask the developers to hold 1.8.4 until some of these items can be fixed and rolled in. Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable inheritance with dialplan command Originate
On 4/8/2011 4:57 AM, Naomi Rosenberg wrote:
Hi,
I would have thought that when spawning a channel using the Originate()
dialplan command, variables prefixed with two underscores would be preserved.
However this does not work in the following case.
Dialplan code:
[intern]
exten = 200,1,Set(__myvar=foo)
exten = 200,n,Originate(Local/123@test_orig,exten,dummy)
[test_orig]
exten = 123,1,NoOp(${myvar})
exten = 123,n,Hangup()
[dummy]
/end dialplan code.
Console output:
-- Executing [200@intern:1] Set(SIP/200-0018, __myvar=foo) in
new stack
-- Executing [200@intern:2] Originate(SIP/200-0018,
Local/123@test_orig,exten,dummy) in new stack
-- Executing [123@test_orig:1] NoOp(Local/123@test_orig-cbab;2, ) in
new stack
-- Executing [123@test_orig:2] Hangup(Local/123@test_orig-cbab;2, )
in new stack
/end console output.
This is in Asterisk 1.8.3.
Is this expected behaviour or a bug, or am I just confused? I would
appreciate your thoughts on the matter.
Thank you,
Naomi
I believe that it's expected behavior because you're not creating a
child channel, you're originating a different set. Try using Dial
instead of Originate, and you'll get the inheritance behavior you expected.
--
Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] IAX2/0.0.29.199
@Paul - many time i am gettting following SIP error when channel isn't available. I want to get rid on this revers thing. I tried all version 1.8.1,1.8.2,1.8.3 but not fix :( [Apr 8 11:52:22] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2f40580 (len 793) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 8 11:52:26] NOTICE[13912]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -Satish Date: Fri, 8 Apr 2011 11:12:59 -0400 From: pabelan...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IAX2/0.0.29.199 On 11-04-08 10:48 AM, satish patel wrote: Where this revers IP comes from ? == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-006b, stdexten,7623,SIP/7623) in new stack -- Executing [s@macro-stdexten:1] ChanIsAvail(SIP/7527-006b, SIP/7623IAX2/7623,20,t) in new stack -- Hungup 'IAX2/0.0.29.199:4569-5255' -- Executing [s@macro-stdexten:2] NoOp(SIP/7527-006b, IAX2/0.0.29.199:4569-5255) in new stack -- Executing [s@macro-stdexten:3] NoOp(SIP/7527-006b, 00) in new stack -- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN' Asterisk 1.8? Are you using realtime? Looks to be an issue with netsock2.c. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable inheritance with dialplan command Originate
Thanks. That's as I thought (feared). Dial is not an option in this case but I
have come up with a workaround involving using a reference number as the
extension and then doing a database call. Not pretty but it works!
Naomi
- Original Message -
From: Sherwood McGowan sherwood.mcgo...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Friday, 8 April, 2011 4:35:43 PM
Subject: Re: [asterisk-users] Variable inheritance with dialplan command
Originate
On 4/8/2011 4:57 AM, Naomi Rosenberg wrote:
Hi,
I would have thought that when spawning a channel using the
Originate() dialplan command, variables prefixed with two underscores
would be preserved.
However this does not work in the following case.
Dialplan code:
[intern]
exten = 200,1,Set(__myvar=foo)
exten = 200,n,Originate(Local/123@test_orig,exten,dummy)
[test_orig]
exten = 123,1,NoOp(${myvar})
exten = 123,n,Hangup()
[dummy]
/end dialplan code.
Console output:
-- Executing [200@intern:1] Set(SIP/200-0018,
__myvar=foo) in new stack
-- Executing [200@intern:2] Originate(SIP/200-0018,
Local/123@test_orig,exten,dummy) in new stack
-- Executing [123@test_orig:1] NoOp(Local/123@test_orig-cbab;2,
) in new stack
-- Executing [123@test_orig:2]
Hangup(Local/123@test_orig-cbab;2, ) in new stack
/end console output.
This is in Asterisk 1.8.3.
Is this expected behaviour or a bug, or am I just confused? I would
appreciate your thoughts on the matter.
Thank you,
Naomi
I believe that it's expected behavior because you're not creating a
child channel, you're originating a different set. Try using Dial
instead of Originate, and you'll get the inheritance behavior you
expected.
-- Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] CRC Zaptel.conf
On Fri, Apr 08, 2011 at 09:11:20AM +, salaheddine elharit wrote: i have a question related to CRC, yesterday i had an issue in our span when i verify with zttool i found recovering instead ok i verify the zaptel.conf and i found # Autogenerated by /usr/sbin/zapconf on Thu Apr 7 17:19:52 2011 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS span=1,1,0,ccs,hdb3,*crc4* # termtype: te bchan=1-15,17-31 dchan=16 ... in the normal zaptel there is no *crc4 * i delete the CRC4 from the 2 spans and all works perfectly. i want to know how the file was generated automatically with CRC4 and if there is any way to disable this option I am not familiar with how the /usr/sbin/zapconf tools works, but with dahdi_genconf in DAHDI, E1 spans default to use CRC4. This covers the majority of users. I did not see any way in /etc/dahdi/genconf_parameters to control the output of the CRC4 line. My opinion is that you did exactly what you needed: run zapconf to get sensible defaults then edit the output file for your specific provider. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable inheritance with dialplan command Originate
Another option is to pass the information in the extension. At times I have an
extension like
_[s][o][m][e]-[e][x][a][m][p][l][e].
And call it like some-example:info1:info2 and use cut to extract the info1 and
info2 values. Not real pretty but as this is computer generated calls it gets
the job done.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 8, 2011, at 8:57 AM, Naomi Rosenberg wrote:
Thanks. That's as I thought (feared). Dial is not an option in this case but
I have come up with a workaround involving using a reference number as the
extension and then doing a database call. Not pretty but it works!
Naomi
- Original Message -
From: Sherwood McGowan sherwood.mcgo...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Friday, 8 April, 2011 4:35:43 PM
Subject: Re: [asterisk-users] Variable inheritance with dialplan command
Originate
On 4/8/2011 4:57 AM, Naomi Rosenberg wrote:
Hi,
I would have thought that when spawning a channel using the
Originate() dialplan command, variables prefixed with two underscores
would be preserved.
However this does not work in the following case.
Dialplan code:
[intern]
exten = 200,1,Set(__myvar=foo)
exten = 200,n,Originate(Local/123@test_orig,exten,dummy)
[test_orig]
exten = 123,1,NoOp(${myvar})
exten = 123,n,Hangup()
[dummy]
/end dialplan code.
Console output:
-- Executing [200@intern:1] Set(SIP/200-0018,
__myvar=foo) in new stack
-- Executing [200@intern:2] Originate(SIP/200-0018,
Local/123@test_orig,exten,dummy) in new stack
-- Executing [123@test_orig:1] NoOp(Local/123@test_orig-cbab;2,
) in new stack
-- Executing [123@test_orig:2]
Hangup(Local/123@test_orig-cbab;2, ) in new stack
/end console output.
This is in Asterisk 1.8.3.
Is this expected behaviour or a bug, or am I just confused? I would
appreciate your thoughts on the matter.
Thank you,
Naomi
I believe that it's expected behavior because you're not creating a
child channel, you're originating a different set. Try using Dial
instead of Originate, and you'll get the inheritance behavior you
expected.
-- Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] Variable inheritance with dialplan command Originate
On 4/8/2011 10:57 AM, Naomi Rosenberg wrote: Thanks. That's as I thought (feared). Dial is not an option in this case but I have come up with a workaround involving using a reference number as the extension and then doing a database call. Not pretty but it works! Naomi I'm not sure why Dial wouldn't work...I use Dial all the time for triggering Local channels that perform database calls all the time -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable inheritance with dialplan command Originate
On 4/8/2011 11:05 AM, Jim Dickenson wrote:
Another option is to pass the information in the extension. At times I have
an extension like
_[s][o][m][e]-[e][x][a][m][p][l][e].
And call it like some-example:info1:info2 and use cut to extract the info1
and info2 values. Not real pretty but as this is computer generated calls it
gets the job done.
Still not sure why you guys need this...Here's my example
[firstleg]
exten = 200,1,Set(__myvar=foo) ; Don't forget you don't want quotes!)
exten = 200,n,Dial(Local/123@test_orig)
[test_orig]
exten = 123,1,Noop(${myvar})
same = n,Set(dbtest=${ODBC_TESTQUERY(myvar)})
--
Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] IAX2/0.0.29.199
Look at this sip debug its saying something related Retransmitting #1 (no NAT) to 0.0.29.200:5060: -- Executing [7624@from-sip:1] Macro(SIP/7527-00c2, stdexten,7624,SIP/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-00c2, SIP/7624IAX2/7624,20,t) in new stack == Using SIP RTP CoS mark 5 [Apr 8 12:20:53] WARNING[15194]: acl.c:698 ast_ouraddrfor: Cannot connect Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 0.0.29.200:5060: INVITE sip:7624 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3af914e2 Max-Forwards: 70 From: Cambridge Guest sip:7527@172.30.1.47;tag=as6f6822ba To: sip:7624 Contact: sip:7527@172.30.1.47:5060 Call-ID: 0ca7784d38d29be168f8f85711c43e4f@172.30.1.47:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Fri, 08 Apr 2011 19:20:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 257 v=0 o=root 1407056235 1407056235 IN IP4 172.30.1.47 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.30.1.47 t=0 0 m=audio 16720 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Apr 8 12:20:53] WARNING[15194]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2ef3f00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7624 Retransmitting #1 (no NAT) to 0.0.29.200:5060: INVITE sip:7624 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3af914e2 Max-Forwards: 70 From: Cambridge Guest sip:7527@172.30.1.47;tag=as6f6822ba To: sip:7624 Contact: sip:7527@172.30.1.47:5060 Call-ID: 0ca7784d38d29be168f8f85711c43e4f@172.30.1.47:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Fri, 08 Apr 2011 19:20:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 257 Date: Fri, 8 Apr 2011 11:12:59 -0400 From: pabelan...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IAX2/0.0.29.199 On 11-04-08 10:48 AM, satish patel wrote: Where this revers IP comes from ? == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-006b, stdexten,7623,SIP/7623) in new stack -- Executing [s@macro-stdexten:1] ChanIsAvail(SIP/7527-006b, SIP/7623IAX2/7623,20,t) in new stack -- Hungup 'IAX2/0.0.29.199:4569-5255' -- Executing [s@macro-stdexten:2] NoOp(SIP/7527-006b, IAX2/0.0.29.199:4569-5255) in new stack -- Executing [s@macro-stdexten:3] NoOp(SIP/7527-006b, 00) in new stack -- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN' Asterisk 1.8? Are you using realtime? Looks to be an issue with netsock2.c. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk login to voicemail
can you explain how this can be done simpler?
On Fri, Apr 8, 2011 at 10:10 AM, Satish Patel satish...@hotmail.com wrote:
Why are you using agi for this ? They are inbuild features of asterisk.
Or may be I am missing something
--
Sent from my iPhone
On Apr 8, 2011, at 8:26 AM, vip killa vipki...@gmail.com wrote:
Wow, thanks, that worked...
in case anyone is interested this is what i did
[voicemail]
exten = a,1,VoiceMailMain(${MAILBOXID}@${MAILBOXCONTEXT},p)
in AGI...
$AGI-set_variable(MAILBOXID, $options);
$AGI-set_variable(MAILBOXCONTEXT,4);
$AGI-set_context(voicemail);
$AGI-exec(VoiceMail, $options);
now the question is how to I get the VoiceMailMain to not ask for Mailbox
and already know which mailbox and just prompt for Password
On Thu, Apr 7, 2011 at 6:44 PM, Dan Journo d...@keshercommunications.com
d...@keshercommunications.com wrote:
Unfortunately, that solution will not work for me... The user must be
able to hit * during the greeting of any voicemail and be prompted for the
Password to that particular mailbox looks like i got a lot of
programming to do to create a work around for this... thanks for your
help...
Forgive me if i'm wrong, but you guys seem to be over complicating things.
Taken from: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
during the prompt if the caller presses:
'*' - the call jumps to extension 'a' in the current voicemail context.
*Example:*
Exten = a, 1, VoicemailMain(@default)
Exten = a, 2, Hangup
When using the star '*' it's important to note that the context you placed
the application voicemail in is irrelevant, it's the context for the
voicemail box that we're looking for in the dialplan for the jump to the 'a'
extension.
So this is what i do...
Before passing the call to the voicemail app, i set ${MAILBOXCONTEXT} to
the correct context, and i set ${MAILBOXID} to the mailbox name.
Then, in extensions.conf, I added this:-
[voicemail]
exten = a,1,Playback(astcc-please-enter-your)
exten = a,n,VoicemailMain(${MAILBOXID}@${MAILBOXCONTEXT})
When the user presses *, they are passed to the 'a' extension above and
into VoicemailMain.
I'm sure you can turn this into AGI easily enough if needed.
Dan Journo
Kesher Communications (UK)
Business Phone Systems http://www.keshercommunications.com/ | Hosted
PBX http://www.keshercommunications.com/hostedpbx.html
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Re: [asterisk-users] asterisk login to voicemail
I have this for same function.
[voice-mail]
;VM for external users calling from PSTN prompt for mailbox number and pin
exten = 8000,1,Answer()
exten = 8000,n,Wait(1)
exten = 8000,n,VoicemailMain(@default)
exten = 8000,n,Hangup()
;VM for internal users only pin
exten = 8500,1,Answer()
exten = 8500,n,Wait(1)
exten = 8500,n,VoicemailMain(${CALLERID(num):-4}@default)
exten = 8500,n,Hangup()
exten = i,1,playback(invalid)
exten = i,2,hangup
Date: Fri, 8 Apr 2011 12:26:27 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk login to voicemail
can you explain how this can be done simpler?
On Fri, Apr 8, 2011 at 10:10 AM, Satish Patel satish...@hotmail.com wrote:
Why are you using agi for this ? They are inbuild features of asterisk.
Or may be I am missing something
--Sent from my iPhone
On Apr 8, 2011, at 8:26 AM, vip killa vipki...@gmail.com wrote:
Wow, thanks, that worked...in case anyone is interested this is what i did
[voicemail]exten = a,1,VoiceMailMain(${MAILBOXID}@${MAILBOXCONTEXT},p)
in AGI...
$AGI-set_variable(MAILBOXID,
$options);$AGI-set_variable(MAILBOXCONTEXT,4);$AGI-set_context(voicemail);
$AGI-exec(VoiceMail, $options);
now the question is how to I get the VoiceMailMain to not ask for Mailbox and
already know which mailbox and just prompt for Password
On Thu, Apr 7, 2011 at 6:44 PM, Dan Journo d...@keshercommunications.com
wrote:
Unfortunately, that solution will not work for me... The user must be able to
hit * during the greeting of any voicemail and be prompted for the Password
to that particular mailbox looks like i got a lot of programming to do to
create a work around for this... thanks for your help...
Forgive me if i'm wrong, but you guys seem to be over complicating things.
Taken from: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
during the prompt if the caller presses:
'*' - the call jumps to extension 'a' in the current voicemail context.
Example:
Exten = a, 1, VoicemailMain(@default)
Exten = a, 2, Hangup
When using the star '*' it's important to note that the context you placed the
application voicemail in is irrelevant, it's the context for the voicemail box
that we're looking for in the dialplan for the jump to the 'a' extension.
So this is what i do...
Before passing the call to the voicemail app, i set ${MAILBOXCONTEXT} to the
correct context, and i set ${MAILBOXID} to the mailbox name.
Then, in extensions.conf, I added this:-[voicemail]
exten = a,1,Playback(astcc-please-enter-your)
exten = a,n,VoicemailMain(${MAILBOXID}@${MAILBOXCONTEXT})When the user presses
*, they are passed to the 'a' extension above and into VoicemailMain.
I'm sure you can turn this into AGI easily enough if needed.
Dan JournoKesher Communications (UK)Business Phone Systems | Hosted PBX
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Re: [asterisk-users] IAX2/0.0.29.199
On 11-04-08 11:55 AM, satish patel wrote: @Paul - many time i am gettting following SIP error when channel isn't available. I want to get rid on this revers thing. I tried all version 1.8.1,1.8.2,1.8.3 but not fix :( Best you can do is collect a full debug[1] log and see when the issue is introduced. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording - methodology
Dan et al;
This looks like a perfect solution.
However, I have one issue. If I initiate the macro manually (put it in
the proper context/dialplan) it works. I see the *.wav file being
created and growing in the /var/spool/asterisk/monitor directory.
If I try to implement it adding the MixMonApp =
*1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I
cannot get it to work.
Steps.
1. added the example macro to the dialplan in extensions.conf
2. added the line MixMonApp = *1,self/both,Macro,mixmon to the
features.conf file under [applicationmap]
3. sip reload / dialplan reload / reload res_features
4. see the message that 'Mapping Feature 'apps' to app 'Macro(callrec)'
5. make incoming call - answer with SIP phone
6. I press *1 on the keypad, I hear the tones, but it does not begin
recording
7. see nothing in the CLI and no new files get created in
/var/spool/asterisk/monitor directory.
What am I missing? Probably something simple.
Any words of wisdom?
Glen
On 4/6/2011 07:29, Dan Journo wrote:
I am looking for a solution to record calls that come into our Asterisk
server. I am hoping for something that is easy to use - however, if I
have to modify it to make it easier to use, I do not mind.
Does anyone know of any opensource or otherwise solutions out there
that
I can try out?
We give our clients to option of either recording all calls, or
allowing the operator to press *1 during a call to start recording
manually.
Using Asterisk 1.4, this is what we do:-
We created a Macro in extensions.conf like this:-
[macro-mixmon]
exten = s,1,GotoIf($[${XAD} = 0 | ${XAD} =
]?startrec:donothing)
exten = s,n(startrec),GotoIf($[${ARG1}=]?beep:nobeep)
exten = s,n(beep),Playback(/var/lib/asterisk/sounds/rec1)
exten = s,n(nobeep),Set(XAD=1)
exten = s,n,MixMonitor(FILENAME.wav,b)
exten = s,n(donothing),MacroExit
(please note, FILENAME.wav is usually ${UNIQUEID:0:10}, but I changed
it to make it easier for you to understand. You'll need to change it
back to something like ${UNIQUEID:0:10}.wav if you are recording
multiple calls because otherwise they'll be constantly saved to
FILENAME.wav and you'll lose all the previous calls.)
(please note, /var/lib/asterisk/sounds/rec1 is a beep tone so that the
operator knows that he's successfully started the recording.)
Then to recording every call, we add this before the
DIAL(SIP/extension) command in extensions.conf:-
exten = _9.,14,Macro(mixmon,nobeep)
If you don't want to record every call, you can give the operator the
option of press *1. We did this by adding the following to features.conf:-
MixMonApp = *1,self/both,Macro,mixmon
Hope that helps.
Dan Journo
Kesher Communications (UK)
Business Phone Systems http://www.keshercommunications.com/ | Hosted
PBX http://www.keshercommunications.com/hostedpbx.html
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Re: [asterisk-users] No ringback even though progressinband=yes is set
On 7 April 2011 23:04, Douglas Mortensen d...@impalanetworks.com wrote: Steve. Thanks for the insight. I won't pretend to know what early-audio is, but I guess I'm about to find out :-). Also, I believe that I have a nearly identical setup like this with the exact same SIP provider w/o any trouble. However, I think that system must be running asterisk 1.4 or 1.2 (my guess is 1.4, but I'll have to check to confirm). Is there a significant difference between 1.2/1.4 1.6 in this scenario? Thanks a million!! :-) - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . -Original Message- From: Steve Davies [mailto:davies...@gmail.com] Sent: Thursday, April 07, 2011 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No ringback even though progressinband=yes is set On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote: Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen In my personal experience with SIP and 1.6.x, that mostly depends on where you are sending the call to. It depends on whether the next or subsequent leg tries to use early-audio for the ring tone, or uses a Ringing event to signal that is what is happening. It then depends on whether the originating caller's equipment can understand early-audio ringing. We have a setup here where all our trunks support early-audio ringing except one (an ISDN30 circuit) and we have to juggle things a bit sometimes to ensure ringing occurs. Perhaps provide more details? Or you may find that tracing the SIP gives you the clue that you need. Hope that helps, Steve Early audio is audio that is sent before the call is answered, usually in the form of a custom ring-tone or perhaps a cannot connect, try later message. Some systems do not support it as it can be abused to communicate at least basic information for free. We had a problem with this when connecting Asterisk 1.2 to Asterisk 1.6 via IAX. A 1.2 SIP system will automatically switch into early audio if it sees an early audio frame. 1.6 defaults to not doing this, but there is a parameter to re-enable it. In this case we solved the problem by upgrading to 1.6 everywhere :) Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/0.0.29.199
On 11-04-08 12:56 PM, Paul Belanger wrote: On 11-04-08 11:55 AM, satish patel wrote: @Paul - many time i am gettting following SIP error when channel isn't available. I want to get rid on this revers thing. I tried all version 1.8.1,1.8.2,1.8.3 but not fix :( Best you can do is collect a full debug[1] log and see when the issue is introduced. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Do you mind trying the following branch[2]? Not sure if it will help, but I made some changes to chan_iax2 a few months ago. [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/0.0.29.199
I can try but i have same issue with chan_sip channel also. and next we have scheduled to put this box 1.8.3.2 in production :( -S Date: Fri, 8 Apr 2011 13:16:30 -0400 From: pabelan...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IAX2/0.0.29.199 On 11-04-08 12:56 PM, Paul Belanger wrote: On 11-04-08 11:55 AM, satish patel wrote: @Paul - many time i am gettting following SIP error when channel isn't available. I want to get rid on this revers thing. I tried all version 1.8.1,1.8.2,1.8.3 but not fix :( Best you can do is collect a full debug[1] log and see when the issue is introduced. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Do you mind trying the following branch[2]? Not sure if it will help, but I made some changes to chan_iax2 a few months ago. [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/0.0.29.199
I have opened case here: https://issues.asterisk.org/view.php?id=19087 Date: Fri, 8 Apr 2011 13:16:30 -0400 From: pabelan...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IAX2/0.0.29.199 On 11-04-08 12:56 PM, Paul Belanger wrote: On 11-04-08 11:55 AM, satish patel wrote: @Paul - many time i am gettting following SIP error when channel isn't available. I want to get rid on this revers thing. I tried all version 1.8.1,1.8.2,1.8.3 but not fix :( Best you can do is collect a full debug[1] log and see when the issue is introduced. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Do you mind trying the following branch[2]? Not sure if it will help, but I made some changes to chan_iax2 a few months ago. [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/0.0.29.199
I have just compiled asterisk 1.6.x and its working without any issue no error related revers lookup etc.. Look like there is some glitch in asterisk 1.8 :( -S Date: Fri, 8 Apr 2011 13:16:30 -0400 From: pabelan...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IAX2/0.0.29.199 On 11-04-08 12:56 PM, Paul Belanger wrote: On 11-04-08 11:55 AM, satish patel wrote: @Paul - many time i am gettting following SIP error when channel isn't available. I want to get rid on this revers thing. I tried all version 1.8.1,1.8.2,1.8.3 but not fix :( Best you can do is collect a full debug[1] log and see when the issue is introduced. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Do you mind trying the following branch[2]? Not sure if it will help, but I made some changes to chan_iax2 a few months ago. [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/0.0.29.199
I tried to compile your version and got bunch of error on make and it failed to compile. root@satish-desktop:/home/satish/issue18183# make [CC] chan_iax2.c - chan_iax2.o chan_iax2.c: In function âsocket_processâ: chan_iax2.c:11533: error: invalid storage class for function âiax2_process_thread_cleanupâ chan_iax2.c:11532: warning: no previous prototype for âiax2_process_thread_cleanupâ chan_iax2.c:11544: error: invalid storage class for function âiax2_process_threadâ chan_iax2.c:11543: warning: no previous prototype for âiax2_process_threadâ chan_iax2.c:11683: error: invalid storage class for function âiax2_do_registerâ chan_iax2.c:11682: warning: no previous prototype for âiax2_do_registerâ chan_iax2.c:11744: error: invalid storage class for function âiax2_provisionâ chan_iax2.c:11743: warning: no previous prototype for âiax2_provisionâ chan_iax2.c:11796: error: invalid storage class for function âiax2_prov_appâ chan_iax2.c:11795: warning: no previous prototype for âiax2_prov_appâ chan_iax2.c:11825: error: invalid storage class for function âhandle_cli_iax2_provisionâ chan_iax2.c:11824: warning: no previous prototype for âhandle_cli_iax2_provisionâ chan_iax2.c:11864: error: invalid storage class for function â__iax2_poke_noanswerâ chan_iax2.c:11863: warning: no previous prototype for â__iax2_poke_noanswerâ chan_iax2.c:11887: error: invalid storage class for function âiax2_poke_noanswerâ ... ... ... chan_iax2.c:14723: warning: no previous prototype for â__reg_moduleâ chan_iax2.c:14723: error: invalid storage class for function â__unreg_moduleâ chan_iax2.c:14723: warning: no previous prototype for â__unreg_moduleâ chan_iax2.c:14723: error: expected declaration or statement at end of input chan_iax2.c:14723: warning: unused variable âast_module_infoâ make[1]: *** [chan_iax2.o] Error 1 make: *** [channels] Error 2 root@satish-desktop:/home/satish/issue18183# Date: Fri, 8 Apr 2011 13:16:30 -0400 From: pabelan...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IAX2/0.0.29.199 On 11-04-08 12:56 PM, Paul Belanger wrote: On 11-04-08 11:55 AM, satish patel wrote: @Paul - many time i am gettting following SIP error when channel isn't available. I want to get rid on this revers thing. I tried all version 1.8.1,1.8.2,1.8.3 but not fix :( Best you can do is collect a full debug[1] log and see when the issue is introduced. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Do you mind trying the following branch[2]? Not sure if it will help, but I made some changes to chan_iax2 a few months ago. [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] send voicemail to multiple emails
Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] send voicemail to multiple emails
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Friday, April 08, 2011 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] send voicemail to multiple emails Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? [Danny Nicholas] If you set up a mail group that might do it. I personally do this with an AGI, but then I work too hard. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] send voicemail to multiple emails
On 4/8/2011 1:13 PM, vip killa wrote: Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The easiest way would be to set up an alias in your MTA configuration. That way, you could configure the mailbox for the alias email address and copies would be sent to all addresses in the alias definition. -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] send voicemail to multiple emails
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: Friday, April 08, 2011 1:16 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] send voicemail to multiple emails On 4/8/2011 1:13 PM, vip killa wrote: Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The easiest way would be to set up an alias in your MTA configuration. That way, you could configure the mailbox for the alias email address and copies would be sent to all addresses in the alias definition. -- Sherwood McGowan sherwood.mcgo...@gmail.com [Danny Nicholas] That is a grand suggestion - as much as I like Asterisk, it is always easier to let Linux do the grunt work when applicable. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] send voicemail to multiple emails
That does not sound easy... besides these email addresses would be taken from a MySQL database. The easiest way would be to set up an alias in your MTA configuration. That way, you could configure the mailbox for the alias email address and copies would be sent to all addresses in the alias definition. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] send voicemail to multiple emails
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Friday, April 08, 2011 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] send voicemail to multiple emails That does not sound easy... besides these email addresses would be taken from a MySQL database. The easiest way would be to set up an alias in your MTA configuration. That way, you could configure the mailbox for the alias email address and copies would be sent to all addresses in the alias definition. [Danny Nicholas] Actually, it is MUCH easier than it sounds. Instead of sending an email to b...@example.com, send it to account...@example.com (set this up in users.conf) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] send voicemail to multiple emails
On 4/8/2011 1:18 PM, vip killa wrote: That does not sound easy... besides these email addresses would be taken from a MySQL database. The easiest way would be to set up an alias in your MTA configuration. That way, you could configure the mailbox for the alias email address and copies would be sent to all addresses in the alias definition. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Easy for you and easiest to configure are two different things. Aliasing email addresses to multiple addresses is not a problem for me, so I shared what I know. You could also just replace the mailcmd (usually sendmail -t) with a script that you wrote (or even application, if you care to compile something) that would take care of the functionality you wish. If you don't think that's easy, just ignore my message, we're not here to compare skill levels. -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] send voicemail to multiple emails
On 4/8/2011 1:20 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: Friday, April 08, 2011 1:16 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] send voicemail to multiple emails On 4/8/2011 1:13 PM, vip killa wrote: Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The easiest way would be to set up an alias in your MTA configuration. That way, you could configure the mailbox for the alias email address and copies would be sent to all addresses in the alias definition. -- Sherwood McGowan sherwood.mcgo...@gmail.com [Danny Nicholas] That is a grand suggestion - as much as I like Asterisk, it is always easier to let Linux do the grunt work when applicable. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks Danny, it's the solution I've used many many times :) -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Documentation for Asterisk AMI Events?
Hi Everyone, I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is specific to 1.6. I am wondering if the developers cared to write about the new events that are spit out in Asterisk 1.8 somewhere on the web? I checked the tar ball for asterisk 1.8 and documentation doesn't include this event: *Event: Unlink* Privilege: call,all Channel1: SIP/-0029 Channel2: SIP/192.168.0.2-002a Uniqueid1: 1302288405.41 Uniqueid2: 1302288405.42 CallerID1: 101 CallerID2: 1212555 So, I am assuming that documentation is outdated or not for asterisk 1.8. Please guide if there is any other place I can grab these. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Documentation for Asterisk AMI Events?
On 11-04-08 02:56 PM, Bruce B wrote: Hi Everyone, I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is specific to 1.6. I am wondering if the developers cared to write about the new events that are spit out in Asterisk 1.8 somewhere on the web? It doesn't exist. The only method at the moment is to look at the code, not the best solution. I'd like to add them for Asterisk 1.10, on my TODO list. Maybe I'll do some work on them this weekend; depends on the weather :) -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Recording using MixMonitor - close, but would like some more words of wisdom.
Dan et al; This looks like a perfect solution. However, I have one issue. If I initiate the macro manually (put it in the proper context/dialplan) it works. I see the *.wav file being created and growing in the /var/spool/asterisk/monitor directory. If I try to implement it adding the MixMonApp = *1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I cannot get it to work. Steps. 1. added the example macro to the dialplan in extensions.conf 2. added the line MixMonApp = *1,self/both,Macro,mixmon to the features.conf file under [applicationmap] 3. sip reload / dialplan reload / reload res_features 4. see the message that 'Mapping Feature 'apps' to app 'Macro(callrec)' 5. make incoming call - answer with SIP phone 6. I press *1 on the keypad, I hear the tones, but it does not begin recording 7. see nothing in the CLI and no new files get created in /var/spool/asterisk/monitor directory. What am I missing? Probably something simple. Any words of wisdom? Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording using MixMonitor - close, but would like some more words of wisdom.
On Fri, Apr 8, 2011 at 3:35 PM, Silver Thorne szilvertho...@gmail.comwrote: Dan et al; This looks like a perfect solution. snip It pretty much is. I've used it in similar situations. I was just about to respond to your original post, but I see you reposted here, so I'll respond here. Steps. 1. added the example macro to the dialplan in extensions.conf 2. added the line MixMonApp = *1,self/both,Macro,mixmon to the features.conf file under [applicationmap] 3. sip reload / dialplan reload / reload res_features 4. see the message that 'Mapping Feature 'apps' to app 'Macro(callrec)' 5. make incoming call - answer with SIP phone 6. I press *1 on the keypad, I hear the tones, but it does not begin recording 7. see nothing in the CLI and no new files get created in /var/spool/asterisk/monitor directory. What am I missing? Probably something simple. DYNAMIC_FEATURES=MixMonApp, either declared in your globals section of extensions.conf, or used in a Set(DYNAMIC_FEATURES=MixMonApp) fashion on a per channel basis in extensions.conf. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] send voicemail to multiple emails
On Fri, Apr 8, 2011 at 1:18 PM, vip killa vipki...@gmail.com wrote: That does not sound easy... besides these email addresses would be taken from a MySQL database. It's actually what you're going to end up doing, whether you do it on the MTA level or your code it into your script that you execute instead of sendmail -f. Currently, there is no way to natively have asterisk send one voicemail to multiple email addresses. What's probably going to work best for you since you seem to like program your own scripts (and I'm not talking an AGI here, I'm talking either pure bash, php, perl, or whichever you prefer), is to change the mailcmd= option inside voicemail.conf and replace it with a script of your own design. I'm not sure off the top of my head which variables are passed to the command, but you could always write a simple script that just outputs all arguments to see and go from there. My guess is you're going to at the least get the preconfigured email address and the contents of your emailsubject and emailbody options (both of which have the option of passing multiple useful variables). -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH not working
Hi Selby,
First of all thanks for reply, I added that line at the end of dialplan but
still the same.
Actually *asterisk default MOH is also not playing*. after 12 sec I get this
message...
* -- Stopped music on hold on SIP/1001-000f*
Strange when I did the same thing with asterisk 1.4.39 then all things are
woking. But with asterisk 1.6.2.18-rc1 not working
On Fri, Apr 8, 2011 at 7:29 PM, Warren Selby wcse...@selbytech.com wrote:
Add exten = 6000,n,StartMusicOnHold() to the end of your current dialplan
and try again.
Thanks,
--Warren Selby, dCAP
On Apr 8, 2011, at 1:51 AM, virendra bhati virbh...@gmail.com wrote:
I am using Elastix. Asterisk is used for PBX application in Elastix. I want
to make customize MOH. But not able to use MOH. I make simple extension in
asterisk conf file but no success :(
Below are the details of configuration files.
Even default MOH is also not working
*Asterisk Version 1.6.2.17.2
*
*1) Extension.conf*
[incoming]
exten = 6000,1,Answer
exten = 6000,n,Set(CHANNEL(musicclass)=BSNL)
exten = 6000,n,Set(foo=${CHANNEL(musicclass)})
exten = 6000,n,MusicOnHold(BSNL)
--
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Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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