[asterisk-users] deadagi on v1.4.xx
Hello all,
I have a perl script that updates a M$ SQL DB based on an ivr that is
run on asterisk.
When it runs as a normal agi, it works great.
when run as a DeadAGI it does not work.
When i execute the script from h channel withDeadAGI and agi debug on i get:
[2010-12-20 01:08:54] -- Launched AGI Script
/var/lib/asterisk/agi-bin/insert_10day_var.pl
[2010-12-20 01:08:54] AGI Tx >> agi_request: insert_10day_var.pl
[2010-12-20 01:08:54] AGI Tx >> agi_channel: SIP/
[2010-12-20 01:08:54] AGI Tx >> agi_language: en
[2010-12-20 01:08:54] AGI Tx >> agi_type: SIP
[2010-12-20 01:08:54] AGI Tx >> agi_uniqueid: 1292825277.243
[2010-12-20 01:08:54] AGI Tx >> agi_callerid:
[2010-12-20 01:08:54] AGI Tx >> agi_calleridname:
[2010-12-20 01:08:54] AGI Tx >> agi_callingpres: 0
[2010-12-20 01:08:54] AGI Tx >> agi_callingani2: 0
[2010-12-20 01:08:54] AGI Tx >> agi_callington: 0
[2010-12-20 01:08:54] AGI Tx >> agi_callingtns: 0
[2010-12-20 01:08:54] AGI Tx >> agi_dnid: unknown
[2010-12-20 01:08:54] AGI Tx >> agi_rdnis: unknown
[2010-12-20 01:08:54] AGI Tx >> agi_context:
[2010-12-20 01:08:54] AGI Tx >> agi_extension: h
[2010-12-20 01:08:54] AGI Tx >> agi_priority: 3
[2010-12-20 01:08:54] AGI Tx >> agi_enhanced: 0.0
[2010-12-20 01:08:54] AGI Tx >> agi_accountcode: in call file>
[2010-12-20 01:08:54] AGI Tx >>
[2010-12-20 01:08:54] -- AGI Script insert_10day_var.pl completed,
returning 0
Which is identical to the debug output when it work from the live
channel AGI
but I do not get the data in the db as it does when run by hand.
I am very tired, frustrated and have been googleing my butt of, no luck.
I know the script is getting the vars in as I had mistakenly left a
print statement in, which of course caused the script to bail but it
showed the correct info before it failed.
Could it be that even though it is running DeadAGI that there is a
sighup killing the script? if any suggestions on what to do about it?
AT: http://www.voip-info.org/wiki/view/Asterisk+cmd+DeadAGI
I did see that in 1.2 even on a deadagi I might have to catch the sighup
and it said
Your script will have to block SIGHUP signals, which you can do like so:
Perl:
$SIG{HUP} = "IGNORE"
I tried this and now at least I get a status after the DeadAGI returns,
which it did not get with out it. Although the status is FAILURE.
Any suggestions?
Thanks,
JohnM
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Re: [asterisk-users] Contradiction between 2 AMI actions QueueSummary and Queuestatus
Let me explain this with some more details. I have 2 members logged into Queue 'retailBanking' using AddQueueMember application on 2 different softphones. The softphones from which these 2 members were added, later unregistered from Asterisk. I then fired below mentioned AMI actions and observed the output. 'QueueSummary' didn't show any member logged into 'retailBanking', where as 'Queuestatus' did show members with 'QueueMember' event. Is this bug or intended behavior? Should I submit a bug report? On Fri, Dec 17, 2010 at 4:59 PM, Asterisk Man wrote: > Asterisk Version: 1.8.0 > Members are added through AddQueueMember in realtime Queues > > > On Fri, Dec 17, 2010 at 4:52 PM, Asterisk Man wrote: > >> Guys, >> Why is such contradiction between 2 AMI actions QueueSummary and >> Queuestatus? >> Look at LoggedIn of QueueSummary and Event: QueueMember. >> Also LongestHoldTime of QueueSummary does not give correct value. >> >> - >> >> Action: QueueSummary >> Queue: retailBanking >> >> Response: Success >> Message: Queue summary will follow >> >> Event: QueueSummary >> Queue: retailBanking >> LoggedIn: 0 >> Available: 0 >> Callers: 0 >> HoldTime: 22 >> TalkTime: 231 >> LongestHoldTime: 0 >> >> Event: QueueSummaryComplete >> - >> Action: Queuestatus >> Queue: retailBanking >> >> Response: Success >> Message: Queue status will follow >> >> Event: QueueParams >> Queue: retailBanking >> Max: 0 >> Strategy: rrmemory >> Calls: 0 >> Holdtime: 22 >> TalkTime: 231 >> Completed: 5 >> Abandoned: 4 >> ServiceLevel: 0 >> ServicelevelPerf: 0.0 >> Weight: 0 >> >> Event: QueueMember >> Queue: retailBanking >> Name: agent2 >> Location: SIP/1110 >> Membership: dynamic >> Penalty: 0 >> CallsTaken: 1 >> LastCall: 1292570332 >> Status: 5 >> Paused: 0 >> >> Event: QueueMember >> Queue: retailBanking >> Name: agent1 >> Location: SIP/ >> Membership: dynamic >> Penalty: 0 >> CallsTaken: 3 >> LastCall: 1292581231 >> Status: 5 >> Paused: 0 >> >> Event: QueueStatusComplete >> - >> >> --AsteriskMan >> > > -- _ -- 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] Ported Asterisk in Android
http://groups.google.com/group/village-telco-dev/files?hl=en&pli=1 On Sun, Dec 19, 2010 at 10:50 PM, Outback Dingo wrote: > check the serval phone project > > > On Fri, Dec 17, 2010 at 5:21 AM, Nikhil wrote: > >> Hi >>Does anyone ported Asterisk to Android OS .please give details >> >> Thanks >> Nikhil >> >> -- >> _ >> -- 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] Ported Asterisk in Android
check the serval phone project On Fri, Dec 17, 2010 at 5:21 AM, Nikhil wrote: > Hi >Does anyone ported Asterisk to Android OS .please give details > > Thanks > Nikhil > > -- > _ > -- 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] Ported Asterisk in Android
reply please.. On 12/17/2010 03:51 PM, Nikhil wrote: Hi Does anyone ported Asterisk to Android OS .please give details Thanks Nikhil -- _ -- 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] (no subject)
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Re: [asterisk-users] Specifying DID for outbound calls
On Sun, Dec 19, 2010 at 2:57 PM, Stephen Reese wrote:
> I believe I have made a little headway. I have two outgoing DID
> contexts and have changed the GotoIf statement to the extension name.
> User One acts as expected and User two now displays unknown when
> calling so I believe it is trying to to goto 20 but it's not quite
> making it. Any tips? Thanks
>
> [outgoing]
> exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
> exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
> exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "201"]?20:10)
> exten => _1NXXNXX,10,Set(CALLERID(all)="User One" <3012323434>)
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
> exten => _1NXXNXX,20,Set(CALLERID(num)="User Two" <3013232322>)
This should either be CALLERID(all) or just set the number on the line
above. As a side note, I prefer to use labels an not line numbers.
Less to change later...
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
> exten => _1NXXNXX,n,Goto(h,1)
I'll also give a +1 to using setvar. It allows you to abstract the
dial plan much more. I use this feature a lot in both static and
Realtime configurations. For example (not tested, but based on live
production code):
sip.conf:
[101]
...
setvar=EXTERNAL_CALLERID="User One <3012323434>"
[201]
...
setvar=EXTERNAL_CALLERID="User Two <3013232322>"
extensions.conf:
[outgoing]
exten => _1NXXNXX,1,Verbose(1, Someone is making a call out)
exten =>
_1NXXNXX,n,ExecIf($[${EXISTS(${EXTERNAL_CALLERID})}]?Set(CALLERID(all)=${EXTERNAL_CALLERID}))
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
But then I am sure there are 100 other ways to do this same thing.
-Jonathan
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Re: [asterisk-users] Specifying DID for outbound calls
> I believe I have made a little headway. I have two outgoing DID
> contexts and have changed the GotoIf statement to the extension name.
> User One acts as expected and User two now displays unknown when
> calling so I believe it is trying to to goto 20 but it's not quite
> making it. Any tips? Thanks
>
> [outgoing]
> exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
> exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
> exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "201"]?20:10)
> exten => _1NXXNXX,10,Set(CALLERID(all)="User One" <3012323434>)
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
> exten => _1NXXNXX,20,Set(CALLERID(num)="User Two" <3013232322>)
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
> exten => _1NXXNXX,n,Goto(h,1)
>
Disregard, I had num instead of all for the CALLERID statement.
Thanks for all of the help!
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Re: [asterisk-users] Specifying DID for outbound calls
On Sun, Dec 19, 2010 at 4:36 AM, Jeroen Eeuwes wrote:
> Hi Stephen,
>
>> Thanks for the heads up, I have been setting the caller-ID but the
>> trouble I'm running into is specifying the which number to call out
>> as. How can an extension specify a different number? See below for my
>> current extension.conf, thanks.
>
> You can check the channel-name to see which extension is making the
> call and set the CallerID accordingly. The channel-name will be
> something like "SIP/201-abc23ef34" or "SIP/User1-def34abc51". The 201
> or User1 part depends on how you put the username in sip.conf You can
> use the CUT function to get the calling extension and then jump to the
> correct CallerID. I've used something like this:
>
> [outgoing]
> exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
> exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
> exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "User2"]?20:10)
> exten => _1NXXNXX,10,Set(CALLERID(num)=3012323434)
> exten => _1NXXNXX,n,Set(CALLERID(name)="User1")
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
> exten => _1NXXNXX,20,Set(CALLERID(num)=3013232322)
> exten => _1NXXNXX,n,Set(CALLERID(name)="User2")
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
>
> But in my case I had two different domains. E.g.
> Dial(SIP/${ext...@provider-id1) and Dial(SIP/${ext...@provider-id2)
> instead of setting the CallerID.
>
> Not that the Cut doesn't work correctly if you use a minus-sign in the
> username.
>
> Best regards,
> Jeroen Eeuwes
I believe I have made a little headway. I have two outgoing DID
contexts and have changed the GotoIf statement to the extension name.
User One acts as expected and User two now displays unknown when
calling so I believe it is trying to to goto 20 but it's not quite
making it. Any tips? Thanks
[outgoing]
exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "201"]?20:10)
exten => _1NXXNXX,10,Set(CALLERID(all)="User One" <3012323434>)
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _1NXXNXX,n,Goto(h,1)
exten => _1NXXNXX,20,Set(CALLERID(num)="User Two" <3013232322>)
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten => _1NXXNXX,n,Goto(h,1)
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Re: [asterisk-users] Specifying DID for outbound calls
>> So I should have multiple incoming and outgoing contexts? Vitelity >> will allow me to use IP routing or user/pass auth, the latter would >> allow me to specify the outgoing context, this would also guarantee >> the correct account is billed and not alone rely on caller-ID. > > Let me clarify further. For calls FROM vitelity you are pretty much > limited to a single context in sip.conf doing IP based matching. Most > equipment will not authenticate to you, and chan_sip currently has no > additional method for separating the accounts into separate contexts. > > For calls TO vitelity you should probably have separate contexts. > >> Thanks for being responsive, I do not work with Asterisk much, >> actually I do not touch it unless I need to add more functionality >> outside of regular patching so my fu is not strong in this area ;-) > > > -- > Joshua Colp > Digium, Inc. | Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > Great, I'll get it changed and see if it helps, 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] Specifying DID for outbound calls
- Original Message - > >> First, when using multiple accounts from the same DID provider, is > >> it > >> ideal to use IP based routing using one context as I currently am > >> or > >> have a separate contexts for each account in the sip.conf? > > > > That's really the only way to do it presently. > > So I should have multiple incoming and outgoing contexts? Vitelity > will allow me to use IP routing or user/pass auth, the latter would > allow me to specify the outgoing context, this would also guarantee > the correct account is billed and not alone rely on caller-ID. Let me clarify further. For calls FROM vitelity you are pretty much limited to a single context in sip.conf doing IP based matching. Most equipment will not authenticate to you, and chan_sip currently has no additional method for separating the accounts into separate contexts. For calls TO vitelity you should probably have separate contexts. > Thanks for being responsive, I do not work with Asterisk much, > actually I do not touch it unless I need to add more functionality > outside of regular patching so my fu is not strong in this area ;-) -- Joshua Colp Digium, Inc. | Software 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] Specifying DID for outbound calls
>> First, when using multiple accounts from the same DID provider, is it >> ideal to use IP based routing using one context as I currently am or >> have a separate contexts for each account in the sip.conf? > > That's really the only way to do it presently. So I should have multiple incoming and outgoing contexts? Vitelity will allow me to use IP routing or user/pass auth, the latter would allow me to specify the outgoing context, this would also guarantee the correct account is billed and not alone rely on caller-ID. Thanks for being responsive, I do not work with Asterisk much, actually I do not touch it unless I need to add more functionality outside of regular patching so my fu is not strong in this area ;-) -- _ -- 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] Specifying DID for outbound calls
- Original Message -
> On Sun, Dec 19, 2010 at 2:40 PM, Joshua Colp wrote:
> > I'm surprised nobody has suggested using the setvar functionality.
> > It's extremely
> > useful for stuff like this and would allow you to keep all CallerID
> > information
> > with the actual configuration of the device.
> >
> > Using a configuration entry for sip.conf in another response as an
> > example:
> >
> > [101]
> > type=friend
> > username=101
> > secret=
> > mailbox=101
> > callerid="User One" <101>
> > host=dynamic
> > nat=yes
> > dtmfmode=rfc2833
> > canreinvite=no
> > reinvite=no
> > qualify=yes
> > setvar=EXTERNAL_CALLERID="User One" <3012323434>
> >
> > And then in extensions.conf:
> >
> > exten => _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
> > exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> >
> > Of course you could add some sanity checking there to make sure that
> > ${EXTERNAL_CALLERID} contains a value and if not default to your
> > main DID.
> >
> > --
> > Joshua Colp
> > Digium, Inc. | Software Developer
> > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> > Check us out at: www.digium.com & www.asterisk.org
>
> Joshua, that seems reasonable. I have two "best practice" questions
> before moving further if anyone would like to chime in.
>
> First, when using multiple accounts from the same DID provider, is it
> ideal to use IP based routing using one context as I currently am or
> have a separate contexts for each account in the sip.conf?
That's really the only way to do it presently.
> Secondly, it never crossed my mind that the caller-ID was being set in
> the sip.conf and extensions.conf. I guess the extension.conf takes
> precedence. At this point is was not my intention to use the sip.conf
> and I can easily remove it and set the variable in the extension.conf.
> I am just not familiar with what is ideal.
The callerid option configures the CallerID that is set on the channel when a
call
comes in from the device. Since there is no logic at that time you can't specify
multiple CallerID values.
The CALLERID dialplan function allows you to change the CallerID on the channel
through
logic you have constructed. It does not care about the previous CallerID, it
simply changes
it.
The suggestion I previously mentioned is a sort of mix of both, it allows you
to set
a separate CallerID when dialing externally by utilizing the setvar option to
make it
available in the dialplan and then the CALLERID dialplan function to actually
change it.
Since the above logic would, presumably, only be executed when dialing
externally you still
need the callerid option set for non-external calls.
--
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Specifying DID for outbound calls
On Sun, Dec 19, 2010 at 2:40 PM, Joshua Colp wrote:
> I'm surprised nobody has suggested using the setvar functionality. It's
> extremely
> useful for stuff like this and would allow you to keep all CallerID
> information
> with the actual configuration of the device.
>
> Using a configuration entry for sip.conf in another response as an example:
>
> [101]
> type=friend
> username=101
> secret=
> mailbox=101
> callerid="User One" <101>
> host=dynamic
> nat=yes
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> qualify=yes
> setvar=EXTERNAL_CALLERID="User One" <3012323434>
>
> And then in extensions.conf:
>
> exten => _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
>
> Of course you could add some sanity checking there to make sure that
> ${EXTERNAL_CALLERID} contains a value and if not default to your
> main DID.
>
> --
> Joshua Colp
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org
Joshua, that seems reasonable. I have two "best practice" questions
before moving further if anyone would like to chime in.
First, when using multiple accounts from the same DID provider, is it
ideal to use IP based routing using one context as I currently am or
have a separate contexts for each account in the sip.conf?
Secondly, it never crossed my mind that the caller-ID was being set in
the sip.conf and extensions.conf. I guess the extension.conf takes
precedence. At this point is was not my intention to use the sip.conf
and I can easily remove it and set the variable in the extension.conf.
I am just not familiar with what is ideal.
Most of my configuration has come from snippets I've found per example.
Thanks
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Re: [asterisk-users] Specifying DID for outbound calls
>On Sun, Dec 19, 2010 at 1:52 PM, William Stillwell
>wrote:
> You can also just use an agi script to look up their current caller-id in a
> database, and set it to the correct caller-id needed.
>
> exten => _NXXNXX,1,AGI(getcid.pl,${CALLERID(NUM)},1)
> exten => _NXXNXX,n,Dial(SIP/+1${ext...@providerx,60)
> exten => _NXXNXX,n,congestion()
>
> my getcid.pl expects two values, extension callerid, and a type.
>
> 911 gets 0, inhouse gets 1, outside 2 etc. (as I ust the getcid for
> different Dial() options.
>
> The script then looks up there "station" callerid, and set it to an
> apporiate value, 911 always gets local in house direct number, regular stuff
> gets a toll number, inhouse gets there extension number, and if there
> callerid is not found in the database it returns a 'default' value.
>
> This way every user can have multiple caller id's .
William, I'm not familiar with AGI scripts, it might be a little
overkill for what I am trying to accomplish.
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Re: [asterisk-users] Specifying DID for outbound calls
- Original Message -
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua
> Colp
> Sent: Sunday, December 19, 2010 2:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Specifying DID for outbound calls
>
> - Original Message -
> > The host I am working with has two accounts from the same DID
> > provider. Incoming calls work correctly and dial the appropriate
> > extensions. This also allows incoming calls to be billed
> > appropriately
> > to the
> > individual DID accounts.
> >
> > Outgoing calls from either extension default to the first DID, i.e.
> > calls from either extension have the same callerID. How can an
> > extension specify separate outgoing contexts so the correct number
> > is
> > associated with it, also allowing the SIP provider to recognize the
> > difference for billing purposes, or is there a better way?
> >
> > In short I'm looking to associate an outgoing call from an extension
> > with a specific number.
> >
> > Here's the sip.conf for both accounts as they are using IP routing,
> > I'm assuming I do not have to perform auth based to separate the two
> > accounts for outgoing calls:
>
> I'm surprised nobody has suggested using the setvar functionality.
> It's
> extremely
> useful for stuff like this and would allow you to keep all CallerID
> information
> with the actual configuration of the device.
>
> Using a configuration entry for sip.conf in another response as an
> example:
>
> [101]
> type=friend
> username=101
> secret=
> mailbox=101
> callerid="User One" <101>
> host=dynamic
> nat=yes
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> qualify=yes
> setvar=EXTERNAL_CALLERID="User One" <3012323434>
>
> And then in extensions.conf:
>
> exten => _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
>
> Of course you could add some sanity checking there to make sure that
> ${EXTERNAL_CALLERID} contains a value and if not default to your
> main DID.
>
>
> How would that work if a user has 3 different callerids, and the use
> of
> realtime?
If there are 3 different CallerIDs then you would have three differently
named dialplan variables containing the appropriate CallerID information
for each. The logic would have to know which dialplan variable to use depending
on the situation, but it would already have to know that regardless.
The only difference is having the CallerID stored with the device configuration
versus in dialplan logic itself.
As for realtime I have not utilized setvar with it but it may be possible to
separate each variable and value using &, just a theory though.
--
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Specifying DID for outbound calls
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Sunday, December 19, 2010 2:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Specifying DID for outbound calls
- Original Message -
> The host I am working with has two accounts from the same DID
> provider. Incoming calls work correctly and dial the appropriate
> extensions. This also allows incoming calls to be billed appropriately
> to the
> individual DID accounts.
>
> Outgoing calls from either extension default to the first DID, i.e.
> calls from either extension have the same callerID. How can an
> extension specify separate outgoing contexts so the correct number is
> associated with it, also allowing the SIP provider to recognize the
> difference for billing purposes, or is there a better way?
>
> In short I'm looking to associate an outgoing call from an extension
> with a specific number.
>
> Here's the sip.conf for both accounts as they are using IP routing,
> I'm assuming I do not have to perform auth based to separate the two
> accounts for outgoing calls:
I'm surprised nobody has suggested using the setvar functionality. It's
extremely
useful for stuff like this and would allow you to keep all CallerID
information
with the actual configuration of the device.
Using a configuration entry for sip.conf in another response as an example:
[101]
type=friend
username=101
secret=
mailbox=101
callerid="User One" <101>
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
setvar=EXTERNAL_CALLERID="User One" <3012323434>
And then in extensions.conf:
exten => _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
Of course you could add some sanity checking there to make sure that
${EXTERNAL_CALLERID} contains a value and if not default to your
main DID.
How would that work if a user has 3 different callerids, and the use of
realtime?
William Stillwell
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Re: [asterisk-users] Specifying DID for outbound calls
- Original Message -
> The host I am working with has two accounts from the same DID
> provider. Incoming calls work correctly and dial the appropriate
> extensions. This also allows incoming calls to be billed appropriately
> to the
> individual DID accounts.
>
> Outgoing calls from either extension default to the first DID, i.e.
> calls from either extension have the same callerID. How can an
> extension specify separate outgoing contexts so the correct number is
> associated with it, also allowing the SIP provider to recognize the
> difference for billing purposes, or is there a better way?
>
> In short I'm looking to associate an outgoing call from an extension
> with a specific number.
>
> Here's the sip.conf for both accounts as they are using IP routing,
> I'm assuming I do not have to perform auth based to separate the two
> accounts for outgoing calls:
I'm surprised nobody has suggested using the setvar functionality. It's
extremely
useful for stuff like this and would allow you to keep all CallerID information
with the actual configuration of the device.
Using a configuration entry for sip.conf in another response as an example:
[101]
type=friend
username=101
secret=
mailbox=101
callerid="User One" <101>
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
setvar=EXTERNAL_CALLERID="User One" <3012323434>
And then in extensions.conf:
exten => _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
Of course you could add some sanity checking there to make sure that
${EXTERNAL_CALLERID} contains a value and if not default to your
main DID.
--
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Specifying DID for outbound calls
On Sun, Dec 19, 2010 at 1:52 PM, William Stillwell
wrote:
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Reese
> Sent: Sunday, December 19, 2010 12:49 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Specifying DID for outbound calls
>
>> You can check the channel-name to see which extension is making the
>> call and set the CallerID accordingly. The channel-name will be
>> something like "SIP/201-abc23ef34" or "SIP/User1-def34abc51". The 201
>> or User1 part depends on how you put the username in sip.conf You can
>> use the CUT function to get the calling extension and then jump to the
>> correct CallerID. I've used something like this:
>>
>> [outgoing]
>> exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
>> exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
>> exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "User2"]?20:10)
>> exten => _1NXXNXX,10,Set(CALLERID(num)=3012323434)
>> exten => _1NXXNXX,n,Set(CALLERID(name)="User1")
>> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
>> exten => _1NXXNXX,n,Goto(h,1)
>> exten => _1NXXNXX,20,Set(CALLERID(num)=3013232322)
>> exten => _1NXXNXX,n,Set(CALLERID(name)="User2")
>> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
>> exten => _1NXXNXX,n,Goto(h,1)
>>
>> But in my case I had two different domains. E.g.
>> Dial(SIP/${ext...@provider-id1) and Dial(SIP/${ext...@provider-id2)
>> instead of setting the CallerID.
>>
>> Not that the Cut doesn't work correctly if you use a minus-sign in the
> username.
>>
>> Best regards,
>> Jeroen Eeuwes
>
> Thanks Jeroen, though it is still not firing correct, I have provided
> a little more information.
>
> Here are the channel-names:
>
> SIP/201-000a
>
> SIP/101-0012
>
> Here is the extension information from the sip.conf:
>
> [101]
> type=friend
> username=101
> secret=
> mailbox=101
> callerid="User One" <101>
> host=dynamic
> nat=yes
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> qualify=yes
>
> [201]
> type=friend
> username=201
> secret=
> mailbox=201
> callerid="User Two" <201>
> host=dynamic
> nat=yes
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> qualify=yes
>
> Here is the updated outgoing context that you provided with a few updates.
>
> [outgoing]
> exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
> exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
> exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "User Two"]?20:10)
> exten => _1NXXNXX,10,Set(CALLERID(num)=3012323434)
> exten => _1NXXNXX,n,Set(CALLERID(name)="User One")
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
> exten => _1NXXNXX,20,Set(CALLERID(num)=3013232322)
> exten => _1NXXNXX,n,Set(CALLERID(name)="User Two")
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
>
> Based on the information above, what should be altered to correctly
> associated the number with the relevant extension?
>
> Thanks
>
>
> You can also just use an agi script to look up their current caller-id in a
> database, and set it to the correct caller-id needed.
>
> exten => _NXXNXX,1,AGI(getcid.pl,${CALLERID(NUM)},1)
> exten => _NXXNXX,n,Dial(SIP/+1${ext...@providerx,60)
> exten => _NXXNXX,n,congestion()
>
> my getcid.pl expects two values, extension callerid, and a type.
>
> 911 gets 0, inhouse gets 1, outside 2 etc. (as I ust the getcid for
> different Dial() options.
>
> The script then looks up there "station" callerid, and set it to an
> apporiate value, 911 always gets local in house direct number, regular stuff
> gets a toll number, inhouse gets there extension number, and if there
> callerid is not found in the database it returns a 'default' value.
>
> This way every user can have multiple caller id's .
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> http://www.asterisk.org/hello
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> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
You're setting a callerid in sip.conf, so in extensions.conf why not:
if callerid(num) = 201, set callerid(num) = 3012323434 (or whatever)?
sean
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Re: [asterisk-users] Specifying DID for outbound calls
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Reese
Sent: Sunday, December 19, 2010 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Specifying DID for outbound calls
> You can check the channel-name to see which extension is making the
> call and set the CallerID accordingly. The channel-name will be
> something like "SIP/201-abc23ef34" or "SIP/User1-def34abc51". The 201
> or User1 part depends on how you put the username in sip.conf You can
> use the CUT function to get the calling extension and then jump to the
> correct CallerID. I've used something like this:
>
> [outgoing]
> exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
> exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
> exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "User2"]?20:10)
> exten => _1NXXNXX,10,Set(CALLERID(num)=3012323434)
> exten => _1NXXNXX,n,Set(CALLERID(name)="User1")
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
> exten => _1NXXNXX,20,Set(CALLERID(num)=3013232322)
> exten => _1NXXNXX,n,Set(CALLERID(name)="User2")
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
>
> But in my case I had two different domains. E.g.
> Dial(SIP/${ext...@provider-id1) and Dial(SIP/${ext...@provider-id2)
> instead of setting the CallerID.
>
> Not that the Cut doesn't work correctly if you use a minus-sign in the
username.
>
> Best regards,
> Jeroen Eeuwes
Thanks Jeroen, though it is still not firing correct, I have provided
a little more information.
Here are the channel-names:
SIP/201-000a
SIP/101-0012
Here is the extension information from the sip.conf:
[101]
type=friend
username=101
secret=
mailbox=101
callerid="User One" <101>
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
[201]
type=friend
username=201
secret=
mailbox=201
callerid="User Two" <201>
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
Here is the updated outgoing context that you provided with a few updates.
[outgoing]
exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "User Two"]?20:10)
exten => _1NXXNXX,10,Set(CALLERID(num)=3012323434)
exten => _1NXXNXX,n,Set(CALLERID(name)="User One")
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _1NXXNXX,n,Goto(h,1)
exten => _1NXXNXX,20,Set(CALLERID(num)=3013232322)
exten => _1NXXNXX,n,Set(CALLERID(name)="User Two")
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _1NXXNXX,n,Goto(h,1)
Based on the information above, what should be altered to correctly
associated the number with the relevant extension?
Thanks
You can also just use an agi script to look up their current caller-id in a
database, and set it to the correct caller-id needed.
exten => _NXXNXX,1,AGI(getcid.pl,${CALLERID(NUM)},1)
exten => _NXXNXX,n,Dial(SIP/+1${ext...@providerx,60)
exten => _NXXNXX,n,congestion()
my getcid.pl expects two values, extension callerid, and a type.
911 gets 0, inhouse gets 1, outside 2 etc. (as I ust the getcid for
different Dial() options.
The script then looks up there "station" callerid, and set it to an
apporiate value, 911 always gets local in house direct number, regular stuff
gets a toll number, inhouse gets there extension number, and if there
callerid is not found in the database it returns a 'default' value.
This way every user can have multiple caller id's .
--
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Re: [asterisk-users] Specifying DID for outbound calls
> You can check the channel-name to see which extension is making the
> call and set the CallerID accordingly. The channel-name will be
> something like "SIP/201-abc23ef34" or "SIP/User1-def34abc51". The 201
> or User1 part depends on how you put the username in sip.conf You can
> use the CUT function to get the calling extension and then jump to the
> correct CallerID. I've used something like this:
>
> [outgoing]
> exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
> exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
> exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "User2"]?20:10)
> exten => _1NXXNXX,10,Set(CALLERID(num)=3012323434)
> exten => _1NXXNXX,n,Set(CALLERID(name)="User1")
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
> exten => _1NXXNXX,20,Set(CALLERID(num)=3013232322)
> exten => _1NXXNXX,n,Set(CALLERID(name)="User2")
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
>
> But in my case I had two different domains. E.g.
> Dial(SIP/${ext...@provider-id1) and Dial(SIP/${ext...@provider-id2)
> instead of setting the CallerID.
>
> Not that the Cut doesn't work correctly if you use a minus-sign in the
> username.
>
> Best regards,
> Jeroen Eeuwes
Thanks Jeroen, though it is still not firing correct, I have provided
a little more information.
Here are the channel-names:
SIP/201-000a
SIP/101-0012
Here is the extension information from the sip.conf:
[101]
type=friend
username=101
secret=
mailbox=101
callerid="User One" <101>
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
[201]
type=friend
username=201
secret=
mailbox=201
callerid="User Two" <201>
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
Here is the updated outgoing context that you provided with a few updates.
[outgoing]
exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "User Two"]?20:10)
exten => _1NXXNXX,10,Set(CALLERID(num)=3012323434)
exten => _1NXXNXX,n,Set(CALLERID(name)="User One")
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _1NXXNXX,n,Goto(h,1)
exten => _1NXXNXX,20,Set(CALLERID(num)=3013232322)
exten => _1NXXNXX,n,Set(CALLERID(name)="User Two")
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _1NXXNXX,n,Goto(h,1)
Based on the information above, what should be altered to correctly
associated the number with the relevant extension?
Thanks
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Re: [asterisk-users] Asterisk and Dahdi ON Amazon EC2
On Wed, Dec 15, 2010 at 12:37:05PM +0530, DHAVAL INDRODIYA wrote: > Guys, > > I have rebooted system, and also same issue i have found that DAHDI module > is not found > i am stuck in what to do for loading DAHDI onto EC2 > > > */etc/init.d/dahdi restart > Unloading DAHDI hardware modules: done > Loading DAHDI hardware modules: > FATAL: Module dahdi not found.* What's the output of: uname -r find /lib/modules -name dahdi.ko -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Problem with AASTRA phone of NO SERVICE
antse...@tiscali.it wrote: When the internet connection for some reason fall down the 2 phones go to "NO SERVICE", searching on internet i found that this is due to DNS service. Has someone solve this problem? or suggestions? Yes, Install Bind either on the Asterisk server or someplace on your network and tell your phones to use it. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- 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] Problem with AASTRA phone of NO SERVICE
Hi, I have 2 phones AASTRA 57i with Asterisk 1.6. When the internet connection for some reason fall down the 2 phones go to "NO SERVICE", searching on internet i found that this is due to DNS service. Has someone solve this problem? or suggestions? Thanks in advance Antonio Supera i limiti: raddoppia la velocità da 10 a 20 Mega! Risparmia con Tutto Incluso: telefono + adsl 20 mega a soli 29,95 € al mese per due anni! SCONTO DI 240 EURO! http://abbonati.tiscali.it/telefono-adsl/prodotti/tc/tuttoincluso/?WT.mc_id=01fw -- _ -- 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] setting up callerid
When I call from a mobile to mobile (both registered on OPENBTS) the correct
caller ID is passed. That is the callerid that I set in the callerid=
field.
When calling from openbts to the PSTN the config header is passed.
Thanks,
Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Friday, December 17, 2010 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] setting up callerid
Hi Dave,
On Thu, Dec 16, 2010 at 1:52 PM, dave george
wrote:
> Tried the following but no luck:
>
> exten => _53.,1,Set(CALLERID(num)=473520)
>
> exten => _53.,n,Dial(SIP/${ext...@ss74)
>
> I am still passing IMSI310410381554227 as the CALLERID.
>
> My peer is setup as follows:
>
> [IMSI310410381554227]
>
> canreinvite=no
>
> type=peer
>
> context=openbts
>
> callerid=473520
I see you are using OpenBTS. To my understanding, OpenBTS does not
support caller ID, so I don't think it can work.
But as I have the same issue as you, I'd be glad to be wrong ! :D Let me
know.
Regards
Axelle
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[asterisk-users] In which version is eventfilter working?
Hey Guys, In which Version of Asterisk is "EventFilter:" in manager.conf working? Higher than 1.6.2.10 or from the 1.8.0 Version? Thank for your answer Daniel -- _ -- 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] Specifying DID for outbound calls
Hi Stephen,
> Thanks for the heads up, I have been setting the caller-ID but the
> trouble I'm running into is specifying the which number to call out
> as. How can an extension specify a different number? See below for my
> current extension.conf, thanks.
You can check the channel-name to see which extension is making the
call and set the CallerID accordingly. The channel-name will be
something like "SIP/201-abc23ef34" or "SIP/User1-def34abc51". The 201
or User1 part depends on how you put the username in sip.conf You can
use the CUT function to get the calling extension and then jump to the
correct CallerID. I've used something like this:
[outgoing]
exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "User2"]?20:10)
exten => _1NXXNXX,10,Set(CALLERID(num)=3012323434)
exten => _1NXXNXX,n,Set(CALLERID(name)="User1")
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _1NXXNXX,n,Goto(h,1)
exten => _1NXXNXX,20,Set(CALLERID(num)=3013232322)
exten => _1NXXNXX,n,Set(CALLERID(name)="User2")
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _1NXXNXX,n,Goto(h,1)
But in my case I had two different domains. E.g.
Dial(SIP/${ext...@provider-id1) and Dial(SIP/${ext...@provider-id2)
instead of setting the CallerID.
Not that the Cut doesn't work correctly if you use a minus-sign in the username.
Best regards,
Jeroen Eeuwes
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Re: [asterisk-users] Specifying DID for outbound calls
On Sun, Dec 19, 2010 at 12:14:11AM -0500, Stephen Reese wrote: >Thanks for the heads up, I have been setting the caller-ID but the >trouble I'm running into is specifying the which number to call out >as. How can an extension specify a different number? See below for my >current extension.conf, thanks. I think I'd probably replace the two outgoing contexts with one, using a GotoIf to distinguish between the two phones (branching into your current code). Alternatively you could give them each a custom context (say phone1 and phone2); phone1 would include incoming and outgoing1, phone2 would include incoming and outgoing2. -- _ -- 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
