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gw@adcomcorp.com

2006-Jan-01 12:42 UTC

[Asterisk-Users] Having major issues with TDM2400

Hello Kerry,

Maybe it's me, but why are you using hint in this fashion?  Shouldn't
you be doing exten => 100,1,Dial(SIP/900&zap/g0/w5551212) or is there
something new that I have missed?

Regards,
Greg

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry
Garrison
Sent: Saturday, December 31, 2005 11:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Having major issues with TDM2400

To summarize, I spent 6 hours yesterday on the phone with Digium trying
to fix a problem with the TDM2400 ad we still don't have it working
right. The lastest version of everything are installed and confirmed by
Digium. So here is the issue:

Zapata.conf
; Disable call progress
; callprogress=yes

Outbound calls to PSTN phone numbers work properly

But using this:

exten => 100,hint,SIP/900&&zap/g0/w5551212

The extension will ring once, but as soon as the PSTN line is picked up,
the sip phone stops ringing because * thinks the phone has been
answered.

Zapata.conf
; Enable call progress
callprogress=yes

Outbound calls to PSTN phone numbers will dial out but there is no
answer detection from the far side. The far side may answer the phone
but * keeps ringing until the timeout expires.

And using this:

exten => 100,hint,SIP/900&&zap/g0/w5551212

Both the sip phone and zap line both ring at the same time until the
time.
Picking up the sip phone bridges the call and disconnects the zap line
as it should.

Any ideas? We are stuck until after the holidays at this point.
-Kerry



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Kerry Garrison

2006-Jan-01 13:24 UTC

head link

[Asterisk-Users] Having major issues with TDM2400

> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com 
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of 
> gw@adcomcorp.com
> Sent: Sunday, January 01, 2006 11:42 AM
> To: asterisk-users@lists.digium.com
> Subject: RE: [Asterisk-Users] Having major issues with TDM2400
> 
> Hello Kerry,
> 
> Maybe it's me, but why are you using hint in this fashion?  
> Shouldn't you be doing exten => 
> 100,1,Dial(SIP/900&zap/g0/w5551212) or is there something new 
> that I have missed?
> 
> Regards,
> Greg
I apologize for not being a config-file pureist, but I have this working
just fine on my office machine (using IAX trunks). 

Below is the extensions_additional.conf as created by AMP. If there is any
more information I can provide, please ask. -Kerry

[ext-local]
include => ext-local-custom
exten => 100,1,Macro(exten-vm,100,100)
exten => ${VM_PREFIX}100,1,Macro(vm,100)
exten => 100,hint,SIP/900&SIP/901&zap/g0/w2831212
exten => 101,1,Macro(exten-vm,novm,101)
exten => 101,hint,
exten => 300,1,Macro(exten-vm,novm,300)
exten => 300,hint,SIP/1000&SIP/1200&zap/g0/w8421212
exten => 301,1,Macro(exten-vm,301,301)
exten => ${VM_PREFIX}301,1,Macro(vm,301)
exten => 301,hint,zap/g0/w9331212&SIP/1001&SIP/1201
exten => 302,1,Macro(exten-vm,302,302)
exten => ${VM_PREFIX}302,1,Macro(vm,302)
exten => 302,1,Macro(exten-vm,302,302)
exten => ${VM_PREFIX}302,1,Macro(vm,302)
exten => 302,hint,SIP/1002&SIP/1202&zap/g0/w17149261212
exten => 303,1,Macro(exten-vm,303,303)
exten => ${VM_PREFIX}303,1,Macro(vm,303)
exten => 303,hint,SIP/1003&SIP/1203&zap/g0/w17143691212
exten => 304,1,Macro(exten-vm,304,304)
exten => ${VM_PREFIX}304,1,Macro(vm,304)
exten => 304,hint,SIP/1004&zap/g0/w17144760731
exten => 305,1,Macro(exten-vm,305,305)
exten => ${VM_PREFIX}305,1,Macro(vm,305)
exten => 305,hint,SIP/1005&SIP/1205&zap/g0/w4331212
exten => 306,1,Macro(exten-vm,306,306)
exten => ${VM_PREFIX}306,1,Macro(vm,306)
exten => 306,hint,SIP/1006&SIP/1206&zap/g0/6361212
exten => 307,1,Macro(exten-vm,307,307)
exten => ${VM_PREFIX}307,1,Macro(vm,307)
exten => 307,hint,SIP/1007&SIP/1207&zap/g0/w15627151212
exten => 308,1,Macro(exten-vm,308,308)
exten => 307,hint,SIP/1007&SIP/1207&zap/g0/w15627151212
exten => 308,1,Macro(exten-vm,308,308)
exten => ${VM_PREFIX}308,1,Macro(vm,308)
exten => 308,hint,zap/g0/w2941212&SIP/1008&SIP/1208
exten => 309,1,Macro(exten-vm,309,309)
exten => ${VM_PREFIX}309,1,Macro(vm,309)
exten => 309,hint,SIP/1009
exten => 310,1,Macro(exten-vm,310,310)
exten => ${VM_PREFIX}310,1,Macro(vm,310)
exten => 310,hint,SIP/1204&SIP/1010
exten => none,hint,

gw@adcomcorp.com

2006-Jan-01 18:05 UTC

head link

[Asterisk-Users] Having major issues with TDM2400

Hello Kerry, I do it exactly as such, however in steps.  My
understanding of the hint system is just for notification of status, not
for execution of dialing. 

I regularly use this same setup you are looking for, rings in, then
rings 2-5 devices (some zap, some iax) and the first one that answers
gets the call.

Make sure you use the Dial( command I replied with previously. (avoid
hint for testing).

Looking at your emails, it looks like you need to review the dialplan
setup, for example the hint and && do not look right to me.

One example for me: exten =>
s,8,Dial(IAX2/ArdsleySomers/314&IAX2/ArdsleySomers/331,,)

But it is the same as SIP/220&Zap/5, etc.

I cannot say anything specific to amp however.

Greg

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry
Garrison
Sent: Sunday, January 01, 2006 7:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Having major issues with TDM2400

The goal is to create a user that has a SIP device and a custom ZAP
channel device, have them both ring until one is answered, basically a
ring group.
But I am using AMP's users and device mode rather than the extensions
mode.
I have this working properly on my office system. However, with the
TDM2400 I cannot have both the zap channel and sip channel ringing at
the same time and only handing the call to the end device that answers
the call. I don't understand why this is so difficult for everyone to
grasp. Send a call to both a custom ZAP device and a sip phone and
whoever answers it gets the call.
-Kerry


 
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F
> Sent: Sunday, January 01, 2006 4:14 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Having major issues with TDM2400
> 
> On 12/31/05, Kerry Garrison <support@techdatapros.com> wrote:
> > To summarize, I spent 6 hours yesterday on the phone with Digium 
> > trying to fix a problem with the TDM2400 ad we still don't have it
> > working right. The lastest version of everything are installed and 
> > confirmed by Digium. So here is the issue:
> >
> > Zapata.conf
> > ; Disable call progress
> > ; callprogress=yes
> >
> > Outbound calls to PSTN phone numbers work properly
> >
> > But using this:
> >
> > exten => 100,hint,SIP/900&&zap/g0/w5551212
> 
> What are you trying to do here? You trying to hint to a zip channel 
> and dial a number using the hint priority?
> 
> >
> > The extension will ring once, but as soon as the PSTN line
> is picked
> > up, the sip phone stops ringing because * thinks the phone
> has been answered.
> 
> Which makes sense to me, since as soon as you start dialing you *are* 
> off hook, which in analog means the phone *is* answered. Since all the
> singalling is done in band, it is not difference than picking up the 
> Zap channel for incoming call, at which point you also understand it's
> considered answered.
> 
> >
> > Zapata.conf
> > ; Enable call progress
> > callprogress=yes
> >
> > Outbound calls to PSTN phone numbers will dial out but there is no 
> > answer detection from the far side. The far side may answer
> the phone
> > but * keeps ringing until the timeout expires.
> >
> 
> So don't use callprogress if it doesn't work for you, in no way do
I
> see this related to the subject line of this post.
> 
> > And using this:
> >
> > exten => 100,hint,SIP/900&&zap/g0/w5551212
> >
> 
> Again what is this suppose to do?
> 
> > Both the sip phone and zap line both ring at the same time
> until the time.
> > Picking up the sip phone bridges the call and disconnects
> the zap line
> > as it should.
> >
> > Any ideas? We are stuck until after the holidays at this point.
> > -Kerry
> >
> >
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 

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   http://lists.digium.com/mailman/listinfo/asterisk-users

gw@adcomcorp.com

2006-Jan-01 18:08 UTC

head link

[Asterisk-Users] Having major issues with TDM2400

Oh just a followup, if you are trying to do an outbound dialout over
analog, what others are saying is correct.  You could consider however
using a voip provider to make the outbound call, then you should have
status.

Greg
 

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gregory
Wiktor - ADCom Corp.
Sent: Sunday, January 01, 2006 8:05 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Having major issues with TDM2400

Hello Kerry, I do it exactly as such, however in steps.  My
understanding of the hint system is just for notification of status, not
for execution of dialing. 

I regularly use this same setup you are looking for, rings in, then
rings 2-5 devices (some zap, some iax) and the first one that answers
gets the call.

Make sure you use the Dial( command I replied with previously. (avoid
hint for testing).

Looking at your emails, it looks like you need to review the dialplan
setup, for example the hint and && do not look right to me.

One example for me: exten =>
s,8,Dial(IAX2/ArdsleySomers/314&IAX2/ArdsleySomers/331,,)

But it is the same as SIP/220&Zap/5, etc.

I cannot say anything specific to amp however.

Greg

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry
Garrison
Sent: Sunday, January 01, 2006 7:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Having major issues with TDM2400

The goal is to create a user that has a SIP device and a custom ZAP
channel device, have them both ring until one is answered, basically a
ring group.
But I am using AMP's users and device mode rather than the extensions
mode.
I have this working properly on my office system. However, with the
TDM2400 I cannot have both the zap channel and sip channel ringing at
the same time and only handing the call to the end device that answers
the call. I don't understand why this is so difficult for everyone to
grasp. Send a call to both a custom ZAP device and a sip phone and
whoever answers it gets the call.
-Kerry


 
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F
> Sent: Sunday, January 01, 2006 4:14 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Having major issues with TDM2400
> 
> On 12/31/05, Kerry Garrison <support@techdatapros.com> wrote:
> > To summarize, I spent 6 hours yesterday on the phone with Digium 
> > trying to fix a problem with the TDM2400 ad we still don't have it
> > working right. The lastest version of everything are installed and 
> > confirmed by Digium. So here is the issue:
> >
> > Zapata.conf
> > ; Disable call progress
> > ; callprogress=yes
> >
> > Outbound calls to PSTN phone numbers work properly
> >
> > But using this:
> >
> > exten => 100,hint,SIP/900&&zap/g0/w5551212
> 
> What are you trying to do here? You trying to hint to a zip channel 
> and dial a number using the hint priority?
> 
> >
> > The extension will ring once, but as soon as the PSTN line
> is picked
> > up, the sip phone stops ringing because * thinks the phone
> has been answered.
> 
> Which makes sense to me, since as soon as you start dialing you *are* 
> off hook, which in analog means the phone *is* answered. Since all the
> singalling is done in band, it is not difference than picking up the 
> Zap channel for incoming call, at which point you also understand it's
> considered answered.
> 
> >
> > Zapata.conf
> > ; Enable call progress
> > callprogress=yes
> >
> > Outbound calls to PSTN phone numbers will dial out but there is no 
> > answer detection from the far side. The far side may answer
> the phone
> > but * keeps ringing until the timeout expires.
> >
> 
> So don't use callprogress if it doesn't work for you, in no way do
I
> see this related to the subject line of this post.
> 
> > And using this:
> >
> > exten => 100,hint,SIP/900&&zap/g0/w5551212
> >
> 
> Again what is this suppose to do?
> 
> > Both the sip phone and zap line both ring at the same time
> until the time.
> > Picking up the sip phone bridges the call and disconnects
> the zap line
> > as it should.
> >
> > Any ideas? We are stuck until after the holidays at this point.
> > -Kerry
> >
> >
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 

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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
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gw@adcomcorp.com

2006-Jan-01 20:53 UTC

head link

[Asterisk-Users] Having major issues with TDM2400

Perhaps a Sipura-3000 could be of use here? Anyone have any ideas about
that?

Greg

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry
Garrison
Sent: Sunday, January 01, 2006 10:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Having major issues with TDM2400

As much as I like the option of implementing a follow-me type of script,
the second problem is that the client wants to use AMP to manage the
extensions.
Just doesn't seem like I have a solution that fits all of the client's
requirements. The easiest solution seems to be to use a SIP trunk for
the outbound call. 
-Kerry



> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F
> Sent: Sunday, January 01, 2006 6:24 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Having major issues with TDM2400
> 
> On 1/1/06, Kerry Garrison <support@techdatapros.com> wrote:
> > Thanks everyone, the reason I posted here was because a
> Digium support
> > tech said "it should work" and he couldn't figure it
out.
> So while I
> > appreciate everyone's comments that it "wont work", a
> technician from
> > Digium said it should, hence I turned to the list for
> clarification. 
> > This is not really a good answer for me to go back to my
> client with
> > as this is one primary feature he liked which pushed him into an 
> > Asterisk solution. For right now,
> 
> It will still work using the M option in the dial command, as I wrote 
> before, also look up the follwoing:
> http://www.voip-info.org/wiki-asterisk+cmd+dial
> http://bugs.digium.com/view.php?id=5574
> Using some creativity you can give your client what you promised plus.
> 
> > their bandwidth is insuffecient for using a SIP provider,
> although a
> > T1 line is on order.
> >
> > -Kerry
> >
> >
> >
> >
> > > -----Original Message-----
> > > From: asterisk-users-bounces@lists.digium.com
> > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of 
> > > gw@adcomcorp.com
> > > Sent: Sunday, January 01, 2006 5:08 PM
> > > To: asterisk-users@lists.digium.com
> > > Subject: RE: [Asterisk-Users] Having major issues with TDM2400
> > >
> > > Oh just a followup, if you are trying to do an outbound
> dialout over
> > > analog, what others are saying is correct.  You could consider 
> > > however using a voip provider to make the outbound call, then you
> > > should have status.
> > >
> > > Greg
> > >
> > >
> > > -----Original Message-----
> > > From: asterisk-users-bounces@lists.digium.com
> > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of 
> > > Gregory Wiktor - ADCom Corp.
> > > Sent: Sunday, January 01, 2006 8:05 PM
> > > To: asterisk-users@lists.digium.com
> > > Subject: RE: [Asterisk-Users] Having major issues with TDM2400
> > >
> > > Hello Kerry, I do it exactly as such, however in steps.  My 
> > > understanding of the hint system is just for notification
> of status,
> > > not for execution of dialing.
> > >
> > > I regularly use this same setup you are looking for,
> rings in, then
> > > rings 2-5 devices (some zap, some iax) and the first one that 
> > > answers gets the call.
> > >
> > > Make sure you use the Dial( command I replied with previously. 
> > > (avoid hint for testing).
> > >
> > > Looking at your emails, it looks like you need to review the 
> > > dialplan setup, for example the hint and && do not look
> right to me.
> > >
> > > One example for me: exten =>
> > > s,8,Dial(IAX2/ArdsleySomers/314&IAX2/ArdsleySomers/331,,)
> > >
> > > But it is the same as SIP/220&Zap/5, etc.
> > >
> > > I cannot say anything specific to amp however.
> > >
> > > Greg
> > >
> > > -----Original Message-----
> > > From: asterisk-users-bounces@lists.digium.com
> > > [mailto:asterisk-users-bounces@lists.digium.com] On
> Behalf Of Kerry
> > > Garrison
> > > Sent: Sunday, January 01, 2006 7:34 PM
> > > To: 'Asterisk Users Mailing List - Non-Commercial
Discussion'
> > > Subject: RE: [Asterisk-Users] Having major issues with TDM2400
> > >
> > > The goal is to create a user that has a SIP device and a
> custom ZAP
> > > channel device, have them both ring until one is
> answered, basically
> > > a ring group.
> > > But I am using AMP's users and device mode rather than the 
> > > extensions mode.
> > > I have this working properly on my office system. 
> However, with the
> > > TDM2400 I cannot have both the zap channel and sip
> channel ringing
> > > at the same time and only handing the call to the end device that
> > > answers the call. I don't understand why this is so difficult
for
> > > everyone to grasp. Send a call to both a custom ZAP
> device and a sip
> > > phone and whoever answers it gets the call.
> > > -Kerry
> > >
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: asterisk-users-bounces@lists.digium.com
> > > > [mailto:asterisk-users-bounces@lists.digium.com] On
> Behalf Of C F
> > > > Sent: Sunday, January 01, 2006 4:14 PM
> > > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > > Subject: Re: [Asterisk-Users] Having major issues with
TDM2400
> > > >
> > > > On 12/31/05, Kerry Garrison <support@techdatapros.com>
wrote:
> > > > > To summarize, I spent 6 hours yesterday on the phone
> with Digium
> > > > > trying to fix a problem with the TDM2400 ad we still
> > > don't have it
> > > > > working right. The lastest version of everything are
> > > installed and
> > > > > confirmed by Digium. So here is the issue:
> > > > >
> > > > > Zapata.conf
> > > > > ; Disable call progress
> > > > > ; callprogress=yes
> > > > >
> > > > > Outbound calls to PSTN phone numbers work properly
> > > > >
> > > > > But using this:
> > > > >
> > > > > exten => 100,hint,SIP/900&&zap/g0/w5551212
> > > >
> > > > What are you trying to do here? You trying to hint to a zip 
> > > > channel and dial a number using the hint priority?
> > > >
> > > > >
> > > > > The extension will ring once, but as soon as the PSTN
line
> > > > is picked
> > > > > up, the sip phone stops ringing because * thinks the
phone
> > > > has been answered.
> > > >
> > > > Which makes sense to me, since as soon as you start dialing
> > > you *are*
> > > > off hook, which in analog means the phone *is* answered.
> > > Since all the
> > >
> > > > singalling is done in band, it is not difference than
> > > picking up the
> > > > Zap channel for incoming call, at which point you also
> > > understand it's
> > >
> > > > considered answered.
> > > >
> > > > >
> > > > > Zapata.conf
> > > > > ; Enable call progress
> > > > > callprogress=yes
> > > > >
> > > > > Outbound calls to PSTN phone numbers will dial out but
> > > there is no
> > > > > answer detection from the far side. The far side may
answer
> > > > the phone
> > > > > but * keeps ringing until the timeout expires.
> > > > >
> > > >
> > > > So don't use callprogress if it doesn't work for
you, in no
> > > way do I
> > > > see this related to the subject line of this post.
> > > >
> > > > > And using this:
> > > > >
> > > > > exten => 100,hint,SIP/900&&zap/g0/w5551212
> > > > >
> > > >
> > > > Again what is this suppose to do?
> > > >
> > > > > Both the sip phone and zap line both ring at the same
time
> > > > until the time.
> > > > > Picking up the sip phone bridges the call and
disconnects
> > > > the zap line
> > > > > as it should.
> > > > >
> > > > > Any ideas? We are stuck until after the holidays at
> this point.
> > > > > -Kerry
> > > > >
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > --Bandwidth and Colocation provided by Easynews.com --
> > > > >
> > > > > Asterisk-Users mailing list
> > > > > To UNSUBSCRIBE or update options visit:
> > > > >   
http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > >
> > > > _______________________________________________
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> > > >
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C F

2006-Jan-02 18:41 UTC

head link

Fwd: [Asterisk-Users] Having major issues with TDM2400

---------- Forwarded message ----------
From: C F <shmaltz@gmail.com>
Date: Jan 2, 2006 8:17 PM
Subject: Re: [Asterisk-Users] Having major issues with TDM2400
To: BJ Weschke <bweschke@gmail.com>


Here are my results from the test, first the dialplan used:

================begin dialplan example=================[default]
exten => 698,1,Noop()
exten => 698,2,Dial(Zap/g1/1732xxxxxxx&Zap/g2/1908xxxxxxx,,M(mtest))
;numbers where changed to xxxxxxxx for privacy reasons nothing to do
with pattern matching

[macro-mtest]
exten => s,1,Noop()
exten => s,2,Read(KEY|goodbye|1) ;read one digit while playing the goodbye
file
exten => s,3,GotoIf($[${KEY} = 1]?50) ;if it's one then goto pri 50
exten => s,4,Noop()
exten => s,5,Goto(1000) ; else just end this macro (I know that it
would end here without this pri as well)

exten => s,50,Noop()
exten => s,51,Set(MACRO_RESULT=CONGESTION) ;one was pressed so just
play congestion app dial
exten => s,52,Goto(1000)
exten => s,1000,Noop()
===========end dialplan example=================
When I called extension 698, both phones (the 732 and the 908) rang as
soon as I picked up one (i.e. the line was answered), ringing to the
other stopped (ouch). So you were right, I have to post this back to
the list to clarify that it will only work consecutive and not in
parallel.
BTW, I used 2 PRIs in this case not POTS, but it shouldn't make any
difference, since answer is what we are looking for, and we got it.


On 1/2/06, BJ Weschke <bweschke@gmail.com> wrote:>  Correct. state==AST_STATE_UP != the two call legs (caller and callee)
> being bridged together.
>
>  wait_for_answer is waiting for the first channel to switch to state
> AST_STATE_UP and then it progresses with Macro which then may or may
> not result in the channel actually becoming bridged with the original
> caller.
>
> On 1/2/06, C F <shmaltz@gmail.com> wrote:
> > And as soon as one answers it is NOT bridged (because Macro is first
> > executing), and this fact (that it is not bridged), dosn't change
> > anything?
> >
> > On 1/2/06, BJ Weschke <bweschke@gmail.com> wrote:
> > >  I believe all the other channels get dumped as soon as THAT
channel
> > > answers. There's a method inside of app_dial called
wait_for_answer
> > > for which the return value is only one channel. It's
thunderdome. Many
> > > can enter, only one leaves.  :)
> > >
> > >  Anyway, wait_for_answer must return that one channel pointer
before
> > > Maco execution begins.
> > >
> > > On 1/2/06, C F <shmaltz@gmail.com> wrote:
> > > > I can't remember if I actualy tested it, but my
understanding is that
> > > > the other phones in an extension that implements the M
option should
> > > > still ring even though one answered already, if it is not
yet bridged.
> > > > I might be wrong though, please let me know. The only way I
realy
> > > > implemented it is with consecutive dialing (pri 1 dials
phone1, and
> > > > pri 2 dials phone2 if in the M macro the right key
wasn't pressed).
> > > > Now writing all this I can tell you I actualy NEVER tested
it on
> > > > multiple channels that are dialed in the same Dial command.
So I don't
> > > > really know.
> > > > I guess you are right that the Macro execution doens't
start until
> > > > THAT channel answered for THAT channel, but what about the
other
> > > > channels? do they get hung up on with the start of the Macro
on ANY of
> > > > the channels? or only after bridging?
> > > >
> > > >
> > > >
> > > > On 1/1/06, BJ Weschke <bweschke@gmail.com> wrote:
> > > > >  Does this work? Looking at app_dial.c, it looks like
Macro execution
> > > > > on a channel doesn't begin until it has found THE
channel that
> > > > > answered first. I could be wrong, and if I am I'd
be interested to
> > > > > know it, because then I know where to look to borrow
that piece of
> > > > > functionality for forked dialing/ivr in app_followme.c
:)
> > > > >
> > > > > ---------- Forwarded message ----------
> > > > > From: C F <shmaltz@gmail.com>
> > > > > Date: Jan 1, 2006 9:23 PM
> > > > > Subject: Re: [Asterisk-Users] Having major issues with
TDM2400
> > > > > To: Asterisk Users Mailing List - Non-Commercial
Discussion
> > > > > <asterisk-users@lists.digium.com>
> > > > >
> > > > >
> > > > > It will still work using the M option in the dial
command, as I wrote
> > > > > before, also look up the follwoing:
> > > > > http://www.voip-info.org/wiki-asterisk+cmd+dial
> > > > > http://bugs.digium.com/view.php?id=5574
> > > > > Using some creativity you can give your client what you
promised plus.
> > > > >
> > > > >
> > > > > --
> > > > > Bird's The Word Technologies, Inc.
> > > > > http://www.btwtech.com/
> > > > >
> > > >
> > >
> > >
> > > --
> > > Bird's The Word Technologies, Inc.
> > > http://www.btwtech.com/
> > >
> >
>
>
> --
> Bird's The Word Technologies, Inc.
> http://www.btwtech.com/
>

asterisk users - Jan 2006 - Having major issues with TDM2400

[Asterisk-Users] Having major issues with TDM2400

[Asterisk-Users] Having major issues with TDM2400

[Asterisk-Users] Having major issues with TDM2400

[Asterisk-Users] Having major issues with TDM2400

[Asterisk-Users] Having major issues with TDM2400

Fwd: [Asterisk-Users] Having major issues with TDM2400