Re: [Asterisk-Users] ringing

[Lance Arbuckle]

Tim Sailer wrote:
> 
> When an extention (either ZAP or SIP) is dialed, the calling person
> hears just silence. Is it possible to get Ringing to work with this?
> It seems to cause people to hang up when there is silence.
> 
> Tim



http://www.voip-info.org/wiki-Asterisk+cmd+Dial

-Lance
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ringing

[Tim Sailer]

On Sun, Feb 08, 2004 at 12:04:09AM -0500, Lance Arbuckle wrote:
> 
> 
> Tim Sailer wrote:
> > 
> > When an extention (either ZAP or SIP) is dialed, the calling person
> > hears just silence. Is it possible to get Ringing to work with this?
> > It seems to cause people to hang up when there is silence.
> > 
> > Tim
> 
> 
> 
> http://www.voip-info.org/wiki-Asterisk+cmd+Dial

Jeez. I'm tired. Thank you. I read that 3 times, and, only on the
fourth time did I pick up on the 'r' option. What would be nice would
be to have a different ring (like our big seimens), regular ring for
an outside line, short-short for an inside line...

Tim

-- 
><
>> Tim Sailer   ><  Coastal Internet, Inc.  <<
>> Network and Systems Operations   ><  PO Box 726  <<
>> http://www.buoy.com  ><  Moriches, NY 11955  <<
>> [EMAIL PROTECTED] ><  (631) 399-2910  (888) 924-3728  <<
><
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ringing

[Andrew Thompson]

- Original Message -
From: "Todd Wallace" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, December 04, 2003 11:36 AM
Subject: [Asterisk-Users] ringing


> I get 2 ringing sounds when placing a SIP call through my carrier.  the
> first sounds European for 1 ring then, it goes to a US ring.
>
> Any thoughts?

It sounds like you're receiving ringback from your local asterisk first.
Then, somewhere along the progress, your asterisk receives an open channel
and connects you to the sip carrier. At this point, the carrier's channel is
not complely established, so you are getting ringback from them.

(Just a theory, but it makes sense in my head.)

-
Andrew Thompson
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The opinions
stated above are yours. You cannot imagine why you ever felt otherwise.



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ringing

[Todd Wallace]

Is there a wait or a setting that I can set so that * does not do this?


> It sounds like you're receiving ringback from your local asterisk first.
> Then, somewhere along the progress, your asterisk receives an open channel
> and connects you to the sip carrier. At this point, the carrier's channel
is
> not complely established, so you are getting ringback from them.
>
> (Just a theory, but it makes sense in my head.)
>
> -
> Andrew Thompson
> Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
> restful it is to watch the cursor blink. Close your eyes. The opinions
> stated above are yours. You cannot imagine why you ever felt otherwise.
>
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ringing

[Andrew Thompson]

- Original Message -
From: "Todd Wallace" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, December 04, 2003 1:04 PM
Subject: Re: [Asterisk-Users] ringing


> > It sounds like you're receiving ringback from your local asterisk first.
> > Then, somewhere along the progress, your asterisk receives an open
channel
> > and connects you to the sip carrier. At this point, the carrier's
channel
> is
> > not complely established, so you are getting ringback from them.
> >
> > (Just a theory, but it makes sense in my head.)
> >
> > -
> > Andrew Thompson
> > Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
> > restful it is to watch the cursor blink. Close your eyes. The opinions
> > stated above are yours. You cannot imagine why you ever felt otherwise.
> >

>
> Is there a wait or a setting that I can set so that * does not do this?
>

You could take the 'r' out of your Dial statement for that type of call.
That way you'd only hear the ringback from the SIP carrier(I think).

There might be a better way, we'll just have to see if anyone else chimes
in...

-
Andrew Thompson
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The opinions
stated above are yours. You cannot imagine why you ever felt otherwise.



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ringing

[Andrew Thompson]

- Original Message -
From: "Todd Wallace" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, December 18, 2003 6:27 PM
Subject: [Asterisk-Users] ringing


>
>
> How do I turn off the initial ringing in Asterisk.  I get an Euro sounding
> ringing prior the ringing from the carrier.  I don't get it on the X100P,
> but do on the SIP outbound side.

Are you dialing with "r"? If so, show application dial from the console.

If not, map out your connections and dial paths for us so we can help.


Andrew Thompson http://aktzero.com/

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Ringing Delay

[WipeOut]

Brian Mulligan wrote:

Sorry if this is a daft question but when a PSTN call comes in on my
X100P the console shows the following;
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
   -- Executing Dial("Zap/1-1", "Zap/2") in new stack
   -- Called 2
   -- Zap/2-1 is ringing
   -- Zap/2-1 is ringing
WARNING[1217602880]: File chan_zap.c, Line 2979 (zt_handle_event):
Didn't finish Caller-ID spill.  Cancelling.-- Zap/2-1 is ringing
   -- Zap/2-1 answered Zap/1-1
   -- Attempting native bridge of Zap/1-1 and Zap/2-1
   -- Hungup 'Zap/2-1'
 == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/1-1'
   -- Hungup 'Zap/1-1'
AS indicated, the call is switched to the context [incoming] which is
configured as follows;
[incoming]
exten=> s,1,Dial,Zap/2
The extension rings but not until the incoming line has rung three
times. If I hang-up the external line before the extension is answered
then the extension continues to ring three more times. Clearly, if I
pick up the extension during this time then nobody is there! 

I would like the extension to ring immediately when the call comes in.
Despite my best efforts I cannot find a configuration element which
addresses it. Could it be my hardware just being too slow?
Thanks
Brian
 

Brian,

AFAIK the delay is caused by the way the X100P detects ringing.. If you 
are using digital lines (ISDN) then there is a signal to tell the device 
connected to the line that it is ringing but with an analog line this is 
not the case..

So the X100P basically looks for the "swings" on the line that indicate 
that it is ringing, it goes through about 3 of them before it answers to 
avoid "phantom" calls which used to happen a lot and were very 
irritating especially in the middle of the night.. So what you are 
experiencing is normal..

Later..

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Ringing Delay

[Eric Wieling]

Chances are it's waiting to get the caller ID info (sent between the
first and the second ring)

On Wed, 2004-03-03 at 12:01, WipeOut wrote:
> Brian Mulligan wrote:
> 
> >Sorry if this is a daft question but when a PSTN call comes in on my
> >X100P the console shows the following;
> >
> >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> >(Ring/Answered)...
> >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> >(Ring/Answered)...
> >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> >(Ring/Answered)...
> >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> >(Ring/Answered)...
> >-- Executing Dial("Zap/1-1", "Zap/2") in new stack
> >-- Called 2
> >-- Zap/2-1 is ringing
> >-- Zap/2-1 is ringing
> >WARNING[1217602880]: File chan_zap.c, Line 2979 (zt_handle_event):
> >Didn't finish Caller-ID spill.  Cancelling.-- Zap/2-1 is ringing
> >-- Zap/2-1 answered Zap/1-1
> >-- Attempting native bridge of Zap/1-1 and Zap/2-1
> >-- Hungup 'Zap/2-1'
> >  == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/1-1'
> >-- Hungup 'Zap/1-1'
> >
> >
> >AS indicated, the call is switched to the context [incoming] which is
> >configured as follows;
> >
> >[incoming]
> >exten=> s,1,Dial,Zap/2
> >
> >The extension rings but not until the incoming line has rung three
> >times. If I hang-up the external line before the extension is answered
> >then the extension continues to ring three more times. Clearly, if I
> >pick up the extension during this time then nobody is there! 
> >
> >I would like the extension to ring immediately when the call comes in.
> >Despite my best efforts I cannot find a configuration element which
> >addresses it. Could it be my hardware just being too slow?
> >
> >Thanks
> >Brian
> >
> >  
> >
> Brian,
> 
> AFAIK the delay is caused by the way the X100P detects ringing.. If you 
> are using digital lines (ISDN) then there is a signal to tell the device 
> connected to the line that it is ringing but with an analog line this is 
> not the case..
> 
> So the X100P basically looks for the "swings" on the line that indicate 
> that it is ringing, it goes through about 3 of them before it answers to 
> avoid "phantom" calls which used to happen a lot and were very 
> irritating especially in the middle of the night.. So what you are 
> experiencing is normal..
> 
> Later..
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
For Asterisk PBX related documentation go to
http://www.digium.com/index.php?menu=documentation and look at the
"Unofficial Links" section also see
http://www.voip-info.org/wiki-Asterisk also see my site at
http://www.fnords.org/~eric/asterisk/

BTEL Consulting

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Ringing Delay

[Brian Mulligan]

Yup, that was it. Set usecallerid=no and it rings right out.
Thanks

On Wed, 2004-03-03 at 18:32, Eric Wieling wrote:
> Chances are it's waiting to get the caller ID info (sent between the
> first and the second ring)
> 
> On Wed, 2004-03-03 at 12:01, WipeOut wrote:
> > Brian Mulligan wrote:
> > 
> > >Sorry if this is a daft question but when a PSTN call comes in on my
> > >X100P the console shows the following;
> > >
> > >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> > >(Ring/Answered)...
> > >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> > >(Ring/Answered)...
> > >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> > >(Ring/Answered)...
> > >NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
> > >(Ring/Answered)...
> > >-- Executing Dial("Zap/1-1", "Zap/2") in new stack
> > >-- Called 2
> > >-- Zap/2-1 is ringing
> > >-- Zap/2-1 is ringing
> > >WARNING[1217602880]: File chan_zap.c, Line 2979 (zt_handle_event):
> > >Didn't finish Caller-ID spill.  Cancelling.-- Zap/2-1 is ringing
> > >-- Zap/2-1 answered Zap/1-1
> > >-- Attempting native bridge of Zap/1-1 and Zap/2-1
> > >-- Hungup 'Zap/2-1'
> > >  == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/1-1'
> > >-- Hungup 'Zap/1-1'
> > >
> > >
> > >AS indicated, the call is switched to the context [incoming] which is
> > >configured as follows;
> > >
> > >[incoming]
> > >exten=> s,1,Dial,Zap/2
> > >
> > >The extension rings but not until the incoming line has rung three
> > >times. If I hang-up the external line before the extension is answered
> > >then the extension continues to ring three more times. Clearly, if I
> > >pick up the extension during this time then nobody is there! 
> > >
> > >I would like the extension to ring immediately when the call comes in.
> > >Despite my best efforts I cannot find a configuration element which
> > >addresses it. Could it be my hardware just being too slow?
> > >
> > >Thanks
> > >Brian
> > >
> > >  
> > >
> > Brian,
> > 
> > AFAIK the delay is caused by the way the X100P detects ringing.. If you 
> > are using digital lines (ISDN) then there is a signal to tell the device 
> > connected to the line that it is ringing but with an analog line this is 
> > not the case..
> > 
> > So the X100P basically looks for the "swings" on the line that indicate 
> > that it is ringing, it goes through about 3 of them before it answers to 
> > avoid "phantom" calls which used to happen a lot and were very 
> > irritating especially in the middle of the night.. So what you are 
> > experiencing is normal..
> > 
> > Later..
> > 
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing Volume

[Eric \"ManxPower\" Wieling]

Jadrien Wauthier wrote:
Does anyone know how to adjust the volume of the ringing application?  I 
have done a lot of internet searching and have not found much.


You cannot do this in Asterisk.

Some SIP phones might allow you to do so by setting an option on the 
phone, but you would have to ask the company that makes that specific 
phone how to do that.

___
--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

Re: [asterisk-users] Ringing Volume

[Bob Chiodini]

Jadrien Wauthier wrote:


> Does anyone know how to adjust the volume of the ringing 
application?  I

> have done a lot of internet searching and have not found much.

You cannot do this in Asterisk.

Some SIP phones might allow you to do so by setting an option on the
phone, but you would have to ask the company that makes that specific
phone how to do that.






If Asterisk generates the audio, then it seems that there would be a 
source file that I could edit if nothing else.


I looked at the app_dial.c, but I didn't see anything.  Maybe I over 
looked something.


If I lower the volume on the phone, then all audio on the phone would 
be lower.  I am just interested in lowering the volume of the 
ringing.  Basically, rings from the pstn is at one level, and the 
rings from Asterisk are at another level.  I need to normalize the 
Asterisk volume.


Thank you so much for your help with this.

Jad



Jad,

Are you referring to the ring back (progress tones) when you call out?  
I have the same issue.  Depending on the type of interface you have to 
the PSTN, you could try raising the inbound gain from the PSTN to match 
that of asterisk.


Bob...
___
--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

Re: [asterisk-users] Ringing issue

[Ishfaq Malik]

Ishfaq Malik wrote:
> Hi
>
> We run a hosted VoIP service for multiple customers off the same server 
> and I'm having an odd issue with just one customer in particular. We're 
> using realtime in a MySQL DB  and this is their dialplan
>
> *** 1. row ***
>  context: pcsu-Identifier
>exten: s
> priority: 1
>  app: Answer
>  appdata:
> *** 2. row ***
>  context: pcsu-Identifier
>exten: s
> priority: 2
>  app: Wait
>  appdata: 2
> *** 3. row ***
>  context: pcsu-Identifier
>exten: s
> priority: 3
>  app: Set
>  appdata: CALLERID(num)=${CALLERID(num)}
> *** 4. row ***
>  context: pcsu-Identifier
>exten: s
> priority: 4
>  app: GotoIfTime
>  appdata: 08:30-17:30|mon-fri|*|*?pcsu-Identifier-work|s|1
> *** 5. row ***
>  context: pcsu-Identifier
>exten: s
> priority: 5
>  app: Playback
>  appdata: pcsu-voicemail-file
> *** 6. row ***
>  context: pcsu-Identifier
>exten: s
> priority: 6
>  app: Voicemail
>  appdata: 2...@pcsu-local|s
> *** 7. row ***
>  context: pcsu-Identifier
>exten: s
> priority: 8
>  app: Hangup
>  appdata:
> *** 8. row ***
>  context: pcsu-Identifier-work
>exten: s
> priority: 1
>  app: Dial
>  appdata: 
> SIP/ukgeonum...@carrier&SIP/ukgeonum...@carrier&SIP/PCSU200&SIP/PCSU201&SIP/PCSU202&SIP/PCSU203&SIP/PCSU204&SIP/PCSU205&SIP/PCSU206|15
> *** 9. row ***
>  context: pcsu-Identifier-work
>exten: s
> priority: 2
>  app: Dial
>  appdata: 
> SIP/ukgeonum...@carrier&SIP/ukgeonum...@carrier&SIP/ukgeonum...@carrier&SIP/PCSU200&SIP/PCSU201&SIP/PCSU202&SIP/PCSU203&SIP/PCSU204&SIP/PCSU205&SIP/PCSU206|20
> *** 10. row ***
>  context: pcsu-Identifier-work
>exten: s
> priority: 3
>  app: Playback
>  appdata: pcsu-voicemail-file
> *** 11. row ***
>  context: pcsu-Identifier-work
>exten: s
> priority: 4
>  app: Voicemail
>  appdata: 2...@pcsu-local|s
> *** 12. row ***
>  context: pcsu-Identifier-work
>exten: s
> priority: 5
>  app: Hangup
>  appdata:
>
>
> I know how daft it looks but they insisted on ringing real UK geographic 
> numbers in the same step as SIP extensions. A while back I changed the 
> initial Answer step to NoOp as the Answer step was distorting our CDR 
> and I hadn't realised that Answer wasn't implicitly required. After I 
> did this the caller stopped hearing a ringing tone when ringing into 
> this dial plan. When I put the Answer step back in instead of the NoOp 
> the caller could hear the ringing tone when dialling in again.
>
> I've tried replacing the Answer with Ringing but I still got silence 
> while the extensions and numbers were ringing.
>
> Any thoughts on this would be helpful and I will be trying to replicate 
> this on out test system.
>
> Thanks in advance
>
> Ish
>   
I should also add, we have no problems with the caller hearing ringing 
with any of the other dial plans on this server even though they start 
with NoOp and not Answer

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing issue

[Ishfaq Malik]

Ishfaq Malik wrote:
> Ishfaq Malik wrote:
>   
>> Hi
>>
>> We run a hosted VoIP service for multiple customers off the same server 
>> and I'm having an odd issue with just one customer in particular. We're 
>> using realtime in a MySQL DB  and this is their dialplan
>>
>> *** 1. row ***
>>  context: pcsu-Identifier
>>exten: s
>> priority: 1
>>  app: Answer
>>  appdata:
>> *** 2. row ***
>>  context: pcsu-Identifier
>>exten: s
>> priority: 2
>>  app: Wait
>>  appdata: 2
>> *** 3. row ***
>>  context: pcsu-Identifier
>>exten: s
>> priority: 3
>>  app: Set
>>  appdata: CALLERID(num)=${CALLERID(num)}
>> *** 4. row ***
>>  context: pcsu-Identifier
>>exten: s
>> priority: 4
>>  app: GotoIfTime
>>  appdata: 08:30-17:30|mon-fri|*|*?pcsu-Identifier-work|s|1
>> *** 5. row ***
>>  context: pcsu-Identifier
>>exten: s
>> priority: 5
>>  app: Playback
>>  appdata: pcsu-voicemail-file
>> *** 6. row ***
>>  context: pcsu-Identifier
>>exten: s
>> priority: 6
>>  app: Voicemail
>>  appdata: 2...@pcsu-local|s
>> *** 7. row ***
>>  context: pcsu-Identifier
>>exten: s
>> priority: 8
>>  app: Hangup
>>  appdata:
>> *** 8. row ***
>>  context: pcsu-Identifier-work
>>exten: s
>> priority: 1
>>  app: Dial
>>  appdata: 
>> SIP/ukgeonum...@carrier&SIP/ukgeonum...@carrier&SIP/PCSU200&SIP/PCSU201&SIP/PCSU202&SIP/PCSU203&SIP/PCSU204&SIP/PCSU205&SIP/PCSU206|15
>> *** 9. row ***
>>  context: pcsu-Identifier-work
>>exten: s
>> priority: 2
>>  app: Dial
>>  appdata: 
>> SIP/ukgeonum...@carrier&SIP/ukgeonum...@carrier&SIP/ukgeonum...@carrier&SIP/PCSU200&SIP/PCSU201&SIP/PCSU202&SIP/PCSU203&SIP/PCSU204&SIP/PCSU205&SIP/PCSU206|20
>> *** 10. row ***
>>  context: pcsu-Identifier-work
>>exten: s
>> priority: 3
>>  app: Playback
>>  appdata: pcsu-voicemail-file
>> *** 11. row ***
>>  context: pcsu-Identifier-work
>>exten: s
>> priority: 4
>>  app: Voicemail
>>  appdata: 2...@pcsu-local|s
>> *** 12. row ***
>>  context: pcsu-Identifier-work
>>exten: s
>> priority: 5
>>  app: Hangup
>>  appdata:
>>
>>
>> I know how daft it looks but they insisted on ringing real UK geographic 
>> numbers in the same step as SIP extensions. A while back I changed the 
>> initial Answer step to NoOp as the Answer step was distorting our CDR 
>> and I hadn't realised that Answer wasn't implicitly required. After I 
>> did this the caller stopped hearing a ringing tone when ringing into 
>> this dial plan. When I put the Answer step back in instead of the NoOp 
>> the caller could hear the ringing tone when dialling in again.
>>
>> I've tried replacing the Answer with Ringing but I still got silence 
>> while the extensions and numbers were ringing.
>>
>> Any thoughts on this would be helpful and I will be trying to replicate 
>> this on out test system.
>>
>> Thanks in advance
>>
>> Ish
>>   
>> 
> I should also add, we have no problems with the caller hearing ringing 
> with any of the other dial plans on this server even though they start 
> with NoOp and not Answer
>
> Ish
>   
Fixed it by using an explicit r option in the dial steps

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE : [Asterisk-Users] Ringing Delay

[f6hqz-m]

Hi Chan,

1/ be sure to have correctly inputed your country zone
2/ disable the fax recognition in zapata.conf

Best Regards,
Francois BERGERET,
France.

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de chan (Alpha
Trilogies Networls)
Envoyé : lundi 27 février 2006 08:35
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Ringing Delay


Hi,
Can some one advice me that how can I make the FXO channels port answer an
incoming calls, means when I call from Lan line to Asterisk TDM400, my phone
get ring immediately. When POT FXO port is ringing, Asterisk seems like
studying the incoming ringing pattern even it did answer the call. I did not
activate the usedestingtive, but why it seems delaying an incoming calls?
Normal PBX, say will only delay 1 cycle as max in analog line, but Asterisk
is about 2 sec...???





___
--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

RE : [Asterisk-Users] Ringing Delay

[chan \(Alpha Trilogies Networls\)]

Hi,
I did change the RING parameters to my country, but seems like no
improvement, so how to confirm the ringing frequency than from Telco, any
device to test it out?


Date: Mon, 27 Feb 2006 09:28:15 +0100
From: <[EMAIL PROTECTED]>
Subject: RE : [Asterisk-Users] Ringing Delay

Hi Chan,

1/ be sure to have correctly inputed your country zone
2/ disable the fax recognition in zapata.conf

Best Regards,
Francois BERGERET,
France.

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de chan (Alpha
Trilogies Networls)
Envoyi : lundi 27 fivrier 2006 08:35
@ : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Ringing Delay


Hi,
Can some one advice me that how can I make the FXO channels port answer an
incoming calls, means when I call from Lan line to Asterisk TDM400, my phone
get ring immediately. When POT FXO port is ringing, Asterisk seems like
studying the incoming ringing pattern even it did answer the call. I did not
activate the usedestingtive, but why it seems delaying an incoming calls?
Normal PBX, say will only delay 1 cycle as max in analog line, but Asterisk
is about 2 sec...???


___
--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

Re: [asterisk-users] Ringing phones

[Doug Crompton]

You did not mention what your FXO (connection to PSTN) hardware is???
Depending on what it is there may be configuration options for things like
'ring thru' and wether the fxo answers or passes the call to *

Doug

On Wed, 8 Nov 2006, Matt wrote:

> Hi,
> I have a system that connects to the PSTN.What do I need to do so
> that when a call comes in, the system will start ringing the hunt
> group I have setup but not actually answer the call?  The problem is
> the system is answering the call, and then passing 'ringing tones'
> back to the caller, so this makes the phone companies
> call-forward-no-answer not work since the telco thinks they have
> answered!
> ___
> --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
>


"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



___
--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

Re: [asterisk-users] Ringing phones

[Matt]

Apologies.. we are using a sangom 4 port FXO card.   It used to work
(or so the company claims that has the PBX), but they are saying it
stopped.. yet nothing has changed on the PBX system.  I have verified
it IS picking up and then passing the call onto the ringgroup (hence
taking it out of the phone companies domain).

On 11/8/06, Doug Crompton <[EMAIL PROTECTED]> wrote:

You did not mention what your FXO (connection to PSTN) hardware is???
Depending on what it is there may be configuration options for things like
'ring thru' and wether the fxo answers or passes the call to *

Doug

On Wed, 8 Nov 2006, Matt wrote:

> Hi,
> I have a system that connects to the PSTN.What do I need to do so
> that when a call comes in, the system will start ringing the hunt
> group I have setup but not actually answer the call?  The problem is
> the system is answering the call, and then passing 'ringing tones'
> back to the caller, so this makes the phone companies
> call-forward-no-answer not work since the telco thinks they have
> answered!
> ___
> --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
>


"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



___
--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

Re: [asterisk-users] Ringing phones

[Andrew Joakimsen]

Why don't you post your configuration?On 11/8/06, Matt <[EMAIL PROTECTED]> wrote:
Apologies.. we are using a sangom 4 port FXO card.   It used to work(or so the company claims that has the PBX), but they are saying itstopped.. yet nothing has changed on the PBX system.  I have verifiedit IS picking up and then passing the call onto the ringgroup (hence
taking it out of the phone companies domain).On 11/8/06, Doug Crompton <[EMAIL PROTECTED]> wrote:> You did not mention what your FXO (connection to PSTN) hardware is???
> Depending on what it is there may be configuration options for things like> 'ring thru' and wether the fxo answers or passes the call to *>> Doug>> On Wed, 8 Nov 2006, Matt wrote:
>> > Hi,> > I have a system that connects to the PSTN.What do I need to do so> > that when a call comes in, the system will start ringing the hunt> > group I have setup but not actually answer the call?  The problem is
> > the system is answering the call, and then passing 'ringing tones'> > back to the caller, so this makes the phone companies> > call-forward-no-answer not work since the telco thinks they have
> > answered!> > ___> > --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> >>>
> "Those that sacrifice essential liberty to obtain a little temporary safety>  deserve neither liberty nor safety."  -- Ben Franklin (1759)>> > *  Doug Crompton   *
> *  Richboro, PA 18954  *> *  215-431-6307*> *  *> * [EMAIL PROTECTED]*> * 
http://www.crompton.com  *> >>> ___> --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 listTo 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

Re: [asterisk-users] Ringing phones

[Time Bandit]

Apologies.. we are using a sangom 4 port FXO card.   It used to work
(or so the company claims that has the PBX), but they are saying it
stopped.. yet nothing has changed on the PBX system.  I have verified
it IS picking up and then passing the call onto the ringgroup (hence
taking it out of the phone companies domain).

Matt,

check in your incoming context that you don't have an "Answer" before
you dial the ringgroup.

If you don't answer and just dial the ringgroup, Asterisk won't pickup
the incoming call until a phone in the ringgroup answers it.

hth
___
--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

Re: [asterisk-users] Ringing phones

[Matt]

The config is pretty simple.. when a call comes in it does an
Answer(), which obviously is going to stop the phone companies
no-answer-call-forward from working.  My question, better perhaps,
is.. is there a way to cause asterisk to push the ringing through to
my ring group, without actually answering the line?

On 11/8/06, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:

Why don't you post your configuration?


On 11/8/06, Matt <[EMAIL PROTECTED]> wrote:
> Apologies.. we are using a sangom 4 port FXO card.   It used to work
> (or so the company claims that has the PBX), but they are saying it
> stopped.. yet nothing has changed on the PBX system.  I have verified
> it IS picking up and then passing the call onto the ringgroup (hence
> taking it out of the phone companies domain).
>
> On 11/8/06, Doug Crompton <[EMAIL PROTECTED]> wrote:
> > You did not mention what your FXO (connection to PSTN) hardware is???
> > Depending on what it is there may be configuration options for things
like
> > 'ring thru' and wether the fxo answers or passes the call to *
> >
> > Doug
> >
> > On Wed, 8 Nov 2006, Matt wrote:
> >
> > > Hi,
> > > I have a system that connects to the PSTN.What do I need to do so
> > > that when a call comes in, the system will start ringing the hunt
> > > group I have setup but not actually answer the call?  The problem is
> > > the system is answering the call, and then passing 'ringing tones'
> > > back to the caller, so this makes the phone companies
> > > call-forward-no-answer not work since the telco thinks they have
> > > answered!
> > > ___
> > > --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
> > >
> >
> >
> > "Those that sacrifice essential liberty to obtain a little temporary
safety
> >  deserve neither liberty nor safety."  -- Ben Franklin (1759)
> >
> > 
> > *  Doug Crompton   *
> > *  Richboro, PA 18954  *
> > *  215-431-6307*
> > *  *
> > * [EMAIL PROTECTED]*
> > * http://www.crompton.com  *
> > 
> >
> >
> > ___
> > --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
>


___
--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

Re: [asterisk-users] Ringing phones

[Noah Miller]

Hi Matt -


The config is pretty simple.. when a call comes in it does an
Answer(), which obviously is going to stop the phone companies
no-answer-call-forward from working.  My question, better perhaps,
is.. is there a way to cause asterisk to push the ringing through to
my ring group, without actually answering the line?


Yes, as suggested earlier, just don't use the Answer() statement.
Just skip it and go directly to the Dial() command for your ring
group.  The only real reason to do an Answer() before a Dial() is if
you're getting audio-skippage (a very technical term) at the beginning
of a call.  This can happen on some FXO cards and phone lines, but it
should generally work without the Answer().

- Noah





On 11/8/06, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:
> Why don't you post your configuration?
>
>
> On 11/8/06, Matt <[EMAIL PROTECTED]> wrote:
> > Apologies.. we are using a sangom 4 port FXO card.   It used to work
> > (or so the company claims that has the PBX), but they are saying it
> > stopped.. yet nothing has changed on the PBX system.  I have verified
> > it IS picking up and then passing the call onto the ringgroup (hence
> > taking it out of the phone companies domain).
> >
> > On 11/8/06, Doug Crompton <[EMAIL PROTECTED]> wrote:
> > > You did not mention what your FXO (connection to PSTN) hardware is???
> > > Depending on what it is there may be configuration options for things
> like
> > > 'ring thru' and wether the fxo answers or passes the call to *
> > >
> > > Doug
> > >
> > > On Wed, 8 Nov 2006, Matt wrote:
> > >
> > > > Hi,
> > > > I have a system that connects to the PSTN.What do I need to do so
> > > > that when a call comes in, the system will start ringing the hunt
> > > > group I have setup but not actually answer the call?  The problem is
> > > > the system is answering the call, and then passing 'ringing tones'
> > > > back to the caller, so this makes the phone companies
> > > > call-forward-no-answer not work since the telco thinks they have
> > > > answered!
> > > > ___
> > > > --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
> > > >
> > >
> > >
> > > "Those that sacrifice essential liberty to obtain a little temporary
> safety
> > >  deserve neither liberty nor safety."  -- Ben Franklin (1759)
> > >
> > > 
> > > *  Doug Crompton   *
> > > *  Richboro, PA 18954  *
> > > *  215-431-6307*
> > > *  *
> > > * [EMAIL PROTECTED]*
> > > * http://www.crompton.com  *
> > > 
> > >
> > >
> > > ___
> > > --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
> >
>
>
> ___
> --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


___
--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

Re: [asterisk-users] Ringing phones

[Eric \"ManxPower\" Wieling]

Matt wrote:

The config is pretty simple.. when a call comes in it does an
Answer(), which obviously is going to stop the phone companies
no-answer-call-forward from working.  My question, better perhaps,
is.. is there a way to cause asterisk to push the ringing through to
my ring group, without actually answering the line?


Yes, don't execute Answer() before the Dial.
___
--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

Re: [asterisk-users] Ringing phones

[Matt]

Ahh ok.. thanks.

On 11/8/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:

Matt wrote:
> The config is pretty simple.. when a call comes in it does an
> Answer(), which obviously is going to stop the phone companies
> no-answer-call-forward from working.  My question, better perhaps,
> is.. is there a way to cause asterisk to push the ringing through to
> my ring group, without actually answering the line?

Yes, don't execute Answer() before the Dial.
___
--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

RE: [asterisk-users] Ringing timer

[Alexander Lopez]

If by ringing duration you mean how long a device will ring, then look
at options to Dial

If you mean how long the ring sounds to the callee look at
indications.conf

Alex


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zenone
Sent: Wednesday, July 26, 2006 5:38 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Ringing timer

Hi!
Does a ringing timer exist in asterisk to control ringing duration? If
not,
is there a way to control ringing duration?
Thanks in advance for your help,
Michel


Message sent using UebiMiau 2.7.8


___
--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

RE: [asterisk-users] Ringing timer

[Zenone]

But my question was, is it possible to free the channel if it rings too
long?
Michel




Message sent using UebiMiau 2.7.8


___
--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

Re: [asterisk-users] Ringing timer

[Eric \"ManxPower\" Wieling]

Zenone wrote:

But my question was, is it possible to free the channel if it rings too
long?


Yes.  "show application dial" in the Asterisk CLI will show you where 
the timeout goes on the Dial line.


--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

___
--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

Re: [asterisk-users] Ringing timer

[Zenone]

- Message d'origine 
De: Eric ManxPower Wieling <[EMAIL PROTECTED]>
A: Zenone <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial
Discussion 
Objet: Re: [asterisk-users] Ringing timer
Date: 26/07/06 12:54

> Zenone wrote:
> > But my question was, is it possible to free the channel if it rings
too
> > long?
> 
> Yes.  "show application dial" in the Asterisk CLI will show you
where 
> the timeout goes on the Dial line.
> 
> -- 
> Now accepting new clients in Birmingham, Atlanta, Huntsville, 
> Chattanooga, and Montgomery.
> 
> 
Thanks! I already read 'Unless there is a timeout specified, the Dial
application will wait indefinitely until one of the called channels answers,
the user hangs up, or
if all of the called channels are busy or unavailable. Dialplan executing
will
continue if no requested channels can be called, or if the timeout expires.'
But did the channel answer when its status is 'ringing'? I think yes but I'm
maybe wrong. If I'm rigth the timeout option can't help me...What about you?


Message sent using UebiMiau 2.7.8


___
--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

Re: [asterisk-users] Ringing timer

[Mojo with Horan & Company, LLC]

Yes, on a Zap FXO channel, when you can hear ringing, the timeout is 
counting down, even if the remote party hasn't answered yet.


Zenone wrote:

- Message d'origine 
De: Eric ManxPower Wieling <[EMAIL PROTECTED]>
A: Zenone <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial
Discussion 
Objet: Re: [asterisk-users] Ringing timer
Date: 26/07/06 12:54


Zenone wrote:
> But my question was, is it possible to free the channel if it rings

too

> long?

Yes.  "show application dial" in the Asterisk CLI will show you
where 

the timeout goes on the Dial line.

--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.




Thanks! I already read 'Unless there is a timeout specified, the Dial
application will wait indefinitely until one of the called channels answers,
the user hangs up, or
if all of the called channels are busy or unavailable. Dialplan executing
will
continue if no requested channels can be called, or if the timeout expires.'
But did the channel answer when its status is 'ringing'? I think yes but I'm
maybe wrong. If I'm rigth the timeout option can't help me...What about you?


Message sent using UebiMiau 2.7.8


___
--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

!DSPAM:500,44c76db5240132002735277!



--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
___
--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

Re: [asterisk-users] Ringing timer

[Zenone]

- Message d'origine 
De: Mojo with Horan & Company, LLC <[EMAIL PROTECTED]>
A: Zenone <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial
Discussion 
Objet: Re: [asterisk-users] Ringing timer
Date: 26/07/06 22:39

> Yes, on a Zap FXO channel, when you can hear ringing, the timeout is 
> counting down, even if the remote party hasn't answered yet.
> 
Thanks!
But I don't understand why, when I wrote this:
exten => _0X,2,Dial(${TRUNK}/${NUMPH},5,H|g)
the called phone rings more than 5 seconds and finally goes on voicemail?


Message sent using UebiMiau 2.7.8


___
--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

Re: [asterisk-users] Ringing timer

[Ralph Liebessohn]

On 7/26/06, Zenone <[EMAIL PROTECTED]> wrote:
But my question was, is it possible to free the channel if it rings toolong?MichelUsing this thread, is there a way to make differents rings? When receiving a call from a internal user () rings different when a external agent calls ().
-- Ralph LiebessohnICQ: 74835911Skype: liebessohn
___
--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

RE: [asterisk-users] Ringing timer

[Alexander Lopez]

Use a variable that is set when the call
comes in such as:

 

Exten => s,n,Set(OUTSIDECALL=1)

 

Then in your dial macro test for variable existence
and change ring via alert info or other distinctive ring methods. It is unfortunate
that it is heavily dependant on technology of the channel used.

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ralph Liebessohn
Sent: Thursday, July 27, 2006 5:28
PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Ringing timer



 

On 7/26/06, Zenone
<[EMAIL PROTECTED]> wrote:





But my question was, is it possible to free the channel if it rings too
long?
Michel






Using this thread, is there a way to make differents rings? 
When receiving a call from a internal user () rings different when a
external agent calls (). 

-- 
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn 






___
--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

RE: [asterisk-users] Ringing timer

[Michel Zenone]

Ok.Thanks a lot! I will try!

___
--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

Re: [asterisk-users] Ringing issue

[Gareth Blades]

You would need to provide more information. Mobiles and landlines are 
not SIP and yet you say calls are coming into your asterisk over SIP. So 
what or who is doing the translation?


Initial thoughts are that it could be you are sending back SIP/180 with 
no session progress and indicating ringing but the other end is 
misconfiguration and not generating its own ring tone. This is possible 
if you have multiple providers sending you calls or one provider using 
different kit for different geographic areas.


On 13/05/14 12:01, D'Arcy J.M. Cain wrote:

I have an issue with ringing.  Some users who call my switch hear
ringing and others don't.  I have researched this and understand the
issue of firewalling and RTP.  My switch has UDP ports 1 to 2
open.  In any case I think that blocked RTP would block all ringing,
not just some.

I have one origination provider.  As far as I can tell the issue is
related to the remote user's provider.  My sister does not hear ringing
when she calls from her Roger's cell phone but she does from her Vonage
phone.  I hear ringing when calling in from my Koodo cell phone.  Some
land lines work and others do not.

The server is not behind a NAT and neither is the origination
provider.  There is a firewall but port 5060 is open (UDP and, just in
case, TCP) as well as the RTP ports mentioned above.

I am not sure where to look next.  I assume that there is some sort of
signaling that I am not doing but I can't figure out where.  Can anyone
suggest what area I should be looking?

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] Ringing issue

[D'Arcy J.M. Cain]

On Tue, 13 May 2014 15:28:26 +0100
Gareth Blades  wrote:
> You would need to provide more information. Mobiles and landlines are 
> not SIP and yet you say calls are coming into your asterisk over SIP.
> So what or who is doing the translation?

My origination provider.  While I do have a SIP address, no one is
calling it and other than local sets (which don't seem to have this
issue) all calls are coming through my single origination provider.
This is why I am confused.  Virtually all calls are coming from the
PSTN through one connection.  If all callers had the problem it would
almost make more sense.

> Initial thoughts are that it could be you are sending back SIP/180
> with no session progress and indicating ringing but the other end is 
> misconfiguration and not generating its own ring tone. This is
> possible if you have multiple providers sending you calls or one
> provider using different kit for different geographic areas.

Geographic doesn't seem to be the issue.  Most calls are coming from
Toronto, Canada where I am.  They come from major carriers.  Rogers is
the largest cell carrier here and that appears to be one place where it
fails.  I am on Koodo which uses the Telus network, the second largest,
and mine works fine.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

-- 
_
-- 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] Ringing issue

[D'Arcy J.M. Cain]

On Tue, 13 May 2014 15:28:26 +0100
Gareth Blades  wrote:
> Initial thoughts are that it could be you are sending back SIP/180
> with no session progress and indicating ringing but the other end is 
> misconfiguration and not generating its own ring tone. This is
> possible if you have multiple providers sending you calls or one
> provider using different kit for different geographic areas.

I seem to have solved this, sorta.  My Provider, Thinktel in Canada,
normally sets "PBX plays ringback" to false meaning that they generate
the ring tone in all cases.  By mistake it was set to true on my
trunk.  They changed that and now the callers are hearing a ring tone.

It's still an interesting question I think.  What if I wanted to do
something with early media?  That is not possible with this setup.

Anyway, here it is for future searchers.  Talk to your origination
provider if you have this problem.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

-- 
_
-- 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] Ringing tones oh323

[Michael Manousos]

Jorge Cisneros wrote:
 
 
When i make a call using oh323 channels, how i can send a ringing sounds 
to indicate to the users that the call is in progress
This is generated by the IP phone. You don't have to do anything
special. In the case that the call has already been answered, and
you want to give back some kind of indication, you could
playback a prerecorded tone.
 
 
thanks
 


Michael.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Ringing multiline phone

[Christopher L. Wade]

Joseph wrote:
Is there a way to ring selective line on multi-line phone.
For example if I'm on the phone talking internally on line 1 and the
calls comes-in the line 2 will automatically ring.
The phone P104 allow extension to be assign each line.
Is there a way to call certain line (example line 3) on multi-line phone
instead of line 1 when the phone is not busy?  

For example the Sip phone P104 has 10-lines when I call this phone
ext.12 line one rings, if line one is busy line 2 will ring.  Is the a
way I to dial straight line 2?
use unique sip account for each line, then use the dialplan to pick what 
line to dial.  i use the same technique for having a 'Line' side line 
button and a 'ICM' side line button on my Cisco 7940's.  Calls from PSTN 
come in on 'Line' side lines, internal intercom calls ring the 'ICM' lines.

-Chris
--
Christopher L. Wade Unistar-Sparco Computers, Inc.
Senior Systems Administratordba Sparco.com
Email: [EMAIL PROTECTED] 7089 Ryburn Drive
Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053
Fax:   (901) 872 8482  USA
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Ringing multiline phone

[Joseph]

On Tue, 2004-12-07 at 16:34 -0600, Christopher L. Wade wrote:
> Joseph wrote:
> > Is there a way to ring selective line on multi-line phone.
> > For example if I'm on the phone talking internally on line 1 and the
> > calls comes-in the line 2 will automatically ring.
> > The phone P104 allow extension to be assign each line.
> > 
> > Is there a way to call certain line (example line 3) on multi-line phone
> > instead of line 1 when the phone is not busy?  
> > 
> > For example the Sip phone P104 has 10-lines when I call this phone
> > ext.12 line one rings, if line one is busy line 2 will ring.  Is the a
> > way I to dial straight line 2?
> > 
> 
> use unique sip account for each line, then use the dialplan to pick what 
> line to dial.  i use the same technique for having a 'Line' side line 
> button and a 'ICM' side line button on my Cisco 7940's.  Calls from PSTN 
> come in on 'Line' side lines, internal intercom calls ring the 'ICM' lines.
> 
> -Chris
> 
I don't think so that is possible on that phone ACT P104. The phone
allows to setup four SIP accounts but they are not linked to 10-line
extensions.  Though I'm not 100% sure. The phone has a feature called:
"Line Key Settings" but what it does is just a wild guess.  Manual
(short guide) does not mention anything about it.

The manual the came with the phone is outdated and ACT is not reponding
to customer emails.

-- 
#Joseph
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ringing after hangup

[Jeremy Jones]

TYPO ALERT...

On Fri, 2004-12-10 at 14:25 -0700, Jeremy Jones wrote:
..snip...
> Using the tdm400p card, with the first port connected to a 24-port fxs
..snip...

That should read "...T400P card...", not "...tdm400p card..."

-- 
Jeremy Jones <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing oddity/stupidity

[Leif Neland]

J. Oquendo wrote:

Anyone experience ring oddities with extensions.conf rollovers? Let me 
summarize...

One of my extensions.conf file is built to ring during the day, ring/go
to voicemail after a certain time:
[main-aa]
exten => s,1,GotoIfTime(17:00-8:30|mon-fri|*|*|*?main-night-aa,s,1)
exten => s,2,GotoIfTime(*|sat-sun|*|*|*?main-night-aa,s,1)

...

[main-night-aa]
exten => s,1,Answer
exten => s,2,Background(/etc/asterisk/night)
exten => s,3,Voicemail([EMAIL PROTECTED])
exten => s,4,Hangup



When in night mode, if someone called, while Asterisk would show the
phone as ringing (and INDEED the phone would ring) the caller wouldn't
hear the phone ring. No music, no ringing no thing until the amount of
time the rings ran out and then be transferred into voicemail. So...
(un)Leet ASCII explanation:
Caller (after hours) --> Dials in --> Press extension --> Asterisk makes
transfer --> Caller hears dead air --> No one answers --> Voicemail -->
Caller now hears voicemail prompts


According to the dialplan, there should be no ring at all, it should go 
directly to voicemail.

How long is the " Caller hears dead air --> No one answers " time?

To comfort the caller you could add
exten => s,1,ringing
exten => s,2,wait(2)
exten => s,3,answer()
exten => s,4,Background(/etc/asterisk/night)
exten => s,5,Voicemail([EMAIL PROTECTED])

Leif

___
--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

Re: [asterisk-users] Ringing all extensions

[Joshua Colp]

- Original Message -
From: J. Oquendo
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thu,
03 Aug 2006 11:20:19 -0300
Subject: [asterisk-users] Ringing all extensions

> I've set up a "ring all" context on my gateway on extensions.conf:
> 
> [EMAIL PROTECTED] ~]# grep "*7" ast/extensions.conf
> exten => *7,1,Dial(SIP/1201&SIP/1202&SIP/1203,15,tr)
> 
> Asterisk shows that it rings the lines but in reality nothing happens... 
> Any thoughts on this?
> 
> 2006-08-02 17:07:20 VERBOSE[7027] logger.c: -- Executing 
> Dial("Zap/1-1", "SIP/1201&SIP/1202&SIP/1203|15|tr") in new stack
> 2006-08-02 17:07:20 NOTICE[7027] app_dial.c: Unable to create channel of 
> type 'SIP' (cause 3 - No route to destination)
> 2006-08-02 17:07:20 VERBOSE[7027] logger.c: -- Called 1201
> 2006-08-02 17:07:20 VERBOSE[7027] logger.c: -- Called 1202
> 2006-08-02 17:07:20 VERBOSE[7027] logger.c: -- Called 1203
> 2006-08-02 17:07:24 VERBOSE[7027] logger.c:   == Spawn extension 
> (main-aa, *7, 1) exited non-zero on 'Zap/1-1'
> 2006-08-02 17:07:24 VERBOSE[7027] logger.c: -- Hungup 'Zap/1-1'
> 
> I truncated to phones to ring to 3 lines but in reality there are 42 
> lines that are supposed to ring at once when *7 is pressed.
> 

When you call each phone individually do they report back that they are 
ringing? What's curious here is that none of them report back a progress 
indication.

Joshua Colp
Digium
___
--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

RE:[asterisk-users] Ringing all extensions

[\(AstATN\)]

Hi "J.
Oquendo" <[EMAIL PROTECTED]>,

What LAN switch that you are
using, and what type of IP phones that you are using? 

 

>I've set up a "ring
all" context on my gateway on extensions.conf:

 

>[EMAIL PROTECTED] ~]# grep
"*7" ast/extensions.conf

>exten =>
*7,1,Dial(SIP/1201&SIP/1202&SIP/1203,15,tr)

 

>Asterisk shows that it
rings the lines but in reality nothing happens... 

>Any thoughts on this?

 

>2006-08-02 17:07:20
VERBOSE[7027] logger.c: -- Executing 

>Dial("Zap/1-1",
"SIP/1201&SIP/1202&SIP/1203|15|tr") in new stack

>2006-08-02 17:07:20
NOTICE[7027] app_dial.c: Unable to create channel of 

>type 'SIP' (cause 3 - No
route to destination)

>2006-08-02 17:07:20
VERBOSE[7027] logger.c: -- Called 1201

>2006-08-02 17:07:20
VERBOSE[7027] logger.c: -- Called 1202

>206-08-02 17:07:20
VERBOSE[7027] logger.c: -- Called 1203

>2006-08-02 17:07:24
VERBOSE[7027] logger.c:   == Spawn extension 

(main-aa, *7, 1) exited
non-zero on 'Zap/1-1'

>2006-08-02 17:07:24
VERBOSE[7027] logger.c: -- Hungup 'Zap/1-1'

 

>I truncated to phones to
ring to 3 lines but in reality there are 42 

>lines that are supposed
to ring at once when *7 is pressed.






___
--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

Re: [asterisk-users] ringing in queues

[Ishfaq Malik]

On 13 March 2015 at 14:04, Matt Hamilton  wrote:

> We use the ringall strategy for a small queue with 4 members. When a call
> comes in, if one of the members is busy, all the phones except the busy
> phone rings (as intended). While the other phones are ringing, if this busy
> phone becomes available again, we would like to have it start ringing.
> Right now it just sits idle.
>
> Is this possible? I played with ringinuse (queues.conf) and callcounter
> (sip.conf) values, but wasn't able to get it going.
>
> Thanks,
> Matt
>
> --
> _
> -- 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
>


Reduce the timeout in the queue configuration (but not in the Queue
application in the dialplan), when the timeout (and the retry) value has
elapsed, all available members will be rung again.

Regards

Ish

-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
-- 
_
-- 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] ringing in queues

[Matt Hamilton]

> Reduce the timeout in the queue 
configuration (but not in the Queue application in the dialplan), when 
the timeout > (and the retry) value has elapsed, all available members 
will be rung again.
>


Thanks, that should do it.



Date: Fri, 13 Mar 2015 14:16:34 +
From: i...@pack-net.co.uk
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ringing in queues



On 13 March 2015 at 14:04, Matt Hamilton  wrote:



We use the ringall strategy for a small queue with 4 members. When a call comes 
in, if one of the members is busy, all the phones except the busy phone rings 
(as intended). While the other phones are ringing, if this busy phone becomes 
available again, we would like to have it start ringing. Right now it just sits 
idle. 

Is this possible? I played with ringinuse (queues.conf) and callcounter 
(sip.conf) values, but wasn't able to get it going. 

Thanks,
Matt
  

--

_

-- 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


Reduce the timeout in the queue configuration (but not in the Queue application 
in the dialplan), when the timeout (and the retry) value has elapsed, all 
available members will be rung again.
Regards
Ish 
-- 
Ishfaq Malik 
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street 
Manchester, M1 2JW
COMPANY REG NO. 04920552



-- 
_
-- 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] Ringing a few phones

[Jennifer Hales]

If you want to dial a number of phones at the same time do "exten =>
5000,1,Dial(SIP/5000&SIP/5001&SIP?5002).  The & value is what does the job.

Kind regards
Jenn

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shidan
Sent: Thursday, June 09, 2005 11:01 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Ringing a few phones

I have a client requirement that multiple phones can be dialed,
however they don't want the pstn phone to pick up automatically
because of  voicemail etc, nothing can be changed on the phones, how
can I handle this requirement, by the way no zap channels are
involved, all the pstn phones are behing another sip gateway.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Ringing a few phones

[Shidan]

Hi Jen thanks for the info but I already knew that, what I want is for
it to not get picked up by voicemail on one of the channels. dialing
them in sequence is not an option either, and as I mentioned changing
the settings on the actual phones isn't an option either.  I remember
there was an option for the user to hit * to accept the call but I
think thats only with ZAP, anyone know of a solution to this problem
or something similar for SIP.


Shidan

On 6/8/05, Jennifer Hales <[EMAIL PROTECTED]> wrote:
> If you want to dial a number of phones at the same time do "exten =>
> 5000,1,Dial(SIP/5000&SIP/5001&SIP?5002).  The & value is what does the job.
> 
> Kind regards
> Jenn
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Shidan
> Sent: Thursday, June 09, 2005 11:01 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Ringing a few phones
> 
> I have a client requirement that multiple phones can be dialed,
> however they don't want the pstn phone to pick up automatically
> because of  voicemail etc, nothing can be changed on the phones, how
> can I handle this requirement, by the way no zap channels are
> involved, all the pstn phones are behing another sip gateway.
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
>
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Ringing a few phones

[Robert Goodyear]

On Jun 8, 2005, at 7:19 PM, Shidan wrote:





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shidan
Sent: Thursday, June 09, 2005 11:01 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Ringing a few phones

I have a client requirement that multiple phones can be dialed,
however they don't want the pstn phone to pick up automatically
because of  voicemail etc, nothing can be changed on the phones, how
can I handle this requirement, by the way no zap channels are
involved, all the pstn phones are behing another sip gateway.


On 6/8/05, Jennifer Hales <[EMAIL PROTECTED]> wrote:

If you want to dial a number of phones at the same time do "exten =>
5000,1,Dial(SIP/5000&SIP/5001&SIP?5002).  The & value is what does 
the job.


Kind regards
Jenn

Hi Jen thanks for the info but I already knew that, what I want is for
it to not get picked up by voicemail on one of the channels. dialing
them in sequence is not an option either, and as I mentioned changing
the settings on the actual phones isn't an option either.  I remember
there was an option for the user to hit * to accept the call but I
think thats only with ZAP, anyone know of a solution to this problem
or something similar for SIP.


Shidan




More details needed. If you cannot control the behavior of the phones 
behind the other SIP GW (as you described it) then your only option is 
to control the duration of ringing to just below the threshold of 
pickup on those phones. Also, what happens when one of those phones is 
busy? If it goes straight to VM then that'll blow the whole timeout 
trick.




Robert Goodyear
Brand Up LLC
http://www.brand-up.com

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing sound doesn't work

[Eric "ManxPower" Wieling]

Simon Perreault wrote:
> Hi,
> 
> I have these extensions:
> 
> exten => 101,1,Dial(SIP/101,15)
> exten => 102,1,Dial(SIP/102,15)
> exten => 0,1,Dial(SIP/101&SIP/102,15,r)
> 
> They work fine and I get the ringing sound if I dial them directly. However, 
> I 
> also have this extension:
> 
> exten => s,1,Answer()
> exten => s,2,Background(viagenie)
> exten => s,3,WaitExten()
> 
> The ringing sound doesn't work for any extension if I use this one. I just 
> get 
> silence until someone answers. How come?
> 
> I use Asterisk 1.4.10. I have attached my extensions.conf file to this email.

You do not have a /etc/asterisk/indications.conf  This file is used to 
provide ringing sounds AFTER a channel has been answered.

BTW, don't use "r" option to Dial.  It doesn't work.

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing sound doesn't work

[Simon Perreault]

On Wednesday 29 August 2007 10:46:18 Eric "ManxPower" Wieling wrote:
> You do not have a /etc/asterisk/indications.conf  This file is used to
> provide ringing sounds AFTER a channel has been answered.

Thanks a million times!

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing for incoming call

[Steve Johnson]

Try putting the wait before the Answer.

...
exten => s,n,Wait(10)
exten => s,n,Answer
...

On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither  wrote:
> Dear All,
>
> I am using Asterisk 1.4 on CentOS 5.  I have an incoming DID provided by
> Vitelity.  When the number is called it goes to my Asterisk box.  The
> protocol is SIP.  This all works just fine if I answer the call and
> begin a playback.
>
> I want to let the number ring for a few seconds before it is answered,
> and would like the caller to hear it ringing.  I have tried:
>
> ...
> exten => s,n,Answer
> exten => s,n,Playtones(ring)
> exten => s,n,Wait(10)
> exten => s,n,StopPlaytones()
> exten => s,n,BackGround(sound file)
> ...
>
> also
>
> ...
> exten => s,n,Answer
> exten => s,n,Ringing()
> exten => s,n,Wait(10)
> exten => s,n,BackGround(sound file)
> ...
>
> I have also tried moving the Answer app to right before the BackGround
> app.
>
> In all cases when I call the number I never hear it ringing.  After the
> 10 second delay, the BackGround app does run.  Connecting to the CLI
> does not give me any useful information - for example the Ringing app is
> shown to run, but the caller does not hear it.
>
> Any suggestions?
>
> Many thanks!
>
> --
> Bob Smither 
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing for incoming call

[Bob Smither]

On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote:
> Try putting the wait before the Answer.
> 
> ...
> exten => s,n,Wait(10)
> exten => s,n,Answer
> ...

Thanks Steve.  I tried that:

> On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither  wrote:
> > Dear All,



> >
> > ...
> > exten => s,n,Answer
> > exten => s,n,Ringing()
> > exten => s,n,Wait(10)
> > exten => s,n,BackGround(sound file)
> > ...
> >
> > I have also tried moving the Answer app to right before the BackGround
> > app.



i.e., after the Wait, but still no joy.

Anything else I need to look at?

Thanks,
-- 
Bob Smither 


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing for incoming call

[Steve Johnson]

If you try just this, what does the caller hear? It should be ringing
for the first 20 sec, and then maybe the congestion tone afterwards.
exten => s,1,Wait(20)
exten => s,n,Hangup

You shouldn't need/use the Ringing() command at all, as the initial
ring before your system answers would be generated by the provider.

If "wait ... answer" doesn't work for you, you'll have to provide more
output from the CLI and tell us more about your configuration.


On Fri, Dec 18, 2009 at 10:29 PM, Bob Smither  wrote:
>
> On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote:
>> Try putting the wait before the Answer.
>>
>> ...
>> exten => s,n,Wait(10)
>> exten => s,n,Answer
>> ...
>
> Thanks Steve.  I tried that:
>
>> On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither  wrote:
>> > Dear All,
>
> 
>
>> >
>> > ...
>> > exten => s,n,Answer
>> > exten => s,n,Ringing()
>> > exten => s,n,Wait(10)
>> > exten => s,n,BackGround(sound file)
>> > ...
>> >
>> > I have also tried moving the Answer app to right before the BackGround
>> > app.
>
> 
>
> i.e., after the Wait, but still no joy.
>
> Anything else I need to look at?
>
> Thanks,
> --
> Bob Smither 
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing for incoming call

[covici]

I have a strange suggestion -- have one extension answer the call and
dial the extension you want -- then it should ring before dialing the
second one.

Bob Smither  wrote:

> 
> On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote:
> > Try putting the wait before the Answer.
> > 
> > ...
> > exten => s,n,Wait(10)
> > exten => s,n,Answer
> > ...
> 
> Thanks Steve.  I tried that:
> 
> > On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither  wrote:
> > > Dear All,
> 
> 
> 
> > >
> > > ...
> > > exten => s,n,Answer
> > > exten => s,n,Ringing()
> > > exten => s,n,Wait(10)
> > > exten => s,n,BackGround(sound file)
> > > ...
> > >
> > > I have also tried moving the Answer app to right before the BackGround
> > > app.
> 
> 
> 
> i.e., after the Wait, but still no joy.
> 
> Anything else I need to look at?
> 
> Thanks,
> -- 
> Bob Smither 
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing for incoming call

[Bob Smither]

On Fri, 2009-12-18 at 23:56 -0600, Steve Johnson wrote:
> If you try just this, what does the caller hear? It should be ringing
> for the first 20 sec, and then maybe the congestion tone afterwards.
> exten => s,1,Wait(20)
> exten => s,n,Hangup

Dialplan:

[cci]
exten => s,1,Wait(10)
exten => s,n,Hangup()

When the number is dialed, here is the CLI output:

Connected to Asterisk 1.4.21.1 currently running on k6-2 (pid = 6283)
Verbosity was 0 and is now 3
-- Executing [8772709...@inbound:1] Goto("SIP/smither-03390860",
"cci|s|1") in new stack
-- Goto (cci,s,1)
-- Executing [...@cci:1] Wait("SIP/smither-03390860", "10") in new
stack
-- Executing [...@cci:2] Hangup("SIP/smither-03390860", "") in new
stack
  == Spawn extension (cci, s, 2) exited non-zero on
'SIP/smither-03390860'

The caller hears silence for 10 seconds.  When the Hangup is executed,
as reported on the CLI, the caller _then_ hears ringing (!?) which
continues until the caller hangs up.

Here is the entry in sip.conf (Asterisk registers with the provider):

[vitel-inbound-cci]
type=friend  
dtmfmode=auto
host=
context=inbound 
username=
secret=
allow=all
insecure=very
nat=yes

Context in extensions.conf:

[inbound]
exten => 8772709688,1,Goto(cci,s,1)

The context [cci] is shown above.

I appreciate the help, as I am confused!

-- 
Bob Smither, PhD   Circuit Concepts, Inc.
=

There are only 10 kinds of people in the world
--Those who understand binary, and those who don't...

=
smit...@c-c-i.com  http://www.C-C-I.Com  281-331-2744(office)  -4616(fax)


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing for incoming call

[Bob Smither]

On Sat, 2009-12-19 at 08:26 -0500, cov...@ccs.covici.com wrote:
> I have a strange suggestion -- have one extension answer the call and
> dial the extension you want -- then it should ring before dialing the
> second one.

Actually, that is pretty close to what I do on a *1.6 box and it works.
Here's what I tried on my *1.4 box (in extensions.conf):

[inbound]
exten => 8772709688,1,Dial(Local/s...@cci,15)
exten => 8772709688,n,Hangup()

[cci]
exten => s,1,Set(CallerContext=${CONTEXT}) ; capture context
; document time of call to console
exten => s,n,NoOp(Time is: ${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
; document caller id to console
exten => s,n,NoOp(CallerID is ${CALLERID(all)})
exten => s,n,Set(TIMEOUT(digit)=3)  ; Set Digit Timeout
exten => s,n,Set(TIMEOUT(response)=10)  ; Set Response Timeout
; create unique call id for this call
exten => s,n,Set(GLOBAL(cid)=${EPOCH})
;
;exten => s,n,Playtones(ring)
exten => s,n,Wait(10)
;exten => s,n,StopPlaytones()
exten => s,n,Answer()
exten => s,n(start),Wait(0.5)
exten => s,n,BackGround(cci/prompt00)
exten => s,n,WaitExten  ; Wait for an extension to be dialed.

I tried both with and without the Playtones(ring) / StopPlaytones()
lines.

Here is what I get from the CLI:

Connected to Asterisk 1.4.21.1 currently running on k6-2 (pid = 8998)
Verbosity was 0 and is now 3
-- Executing [8772709...@inbound:1] Dial("SIP/smither-173b4940",
"Local/s...@cci|15") in new stack
-- Called s...@cci
-- Executing [...@cci:1] Set("Local/s...@cci-7c61,2",
"CallerContext=cci") in new stack
-- Executing [...@cci:2] NoOp("Local/s...@cci-7c61,2", "Time is:
2009-12-19 09:43:10") in new stack
-- Executing [...@cci:3] NoOp("Local/s...@cci-7c61,2", "CallerID is
"*" <*>") in new stack
-- Executing [...@cci:4] Set("Local/s...@cci-7c61,2", "TIMEOUT(digit)=3")
in new stack
-- Digit timeout set to 3
-- Executing [...@cci:5] Set("Local/s...@cci-7c61,2",
"TIMEOUT(response)=10") in new stack
-- Response timeout set to 10
-- Executing [...@cci:6] Set("Local/s...@cci-7c61,2",
"GLOBAL(cid)=1261237390") in new stack
  == Setting global variable 'cid' to '1261237390'
-- Executing [...@cci:7] PlayTones("Local/s...@cci-7c61,2", "ring") in
new stack
-- Executing [...@cci:8] Wait("Local/s...@cci-7c61,2", "10") in new stack
-- Executing [...@cci:9] StopPlayTones("Local/s...@cci-7c61,2", "") in
new stack
-- Executing [...@cci:10] Answer("Local/s...@cci-7c61,2", "") in new
stack
-- Executing [...@cci:11] Wait("Local/s...@cci-7c61,2", "0.5") in new
stack
-- Local/s...@cci-7c61,1 answered SIP/smither-173b4940
-- Executing [...@cci:12] BackGround("Local/s...@cci-7c61,2",
"cci/prompt00") in 
new stack
--  Playing 'cci/prompt00' (language 'en')

This all looks as expected to me, but the caller hears nothing until the
BackGround statement is executed.  There still is no ringing back to the
caller.

Thanks!



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing for incoming call

[Andrew Thomas]

exten => did,1,Answer
exten => did,n,Playtones(ring)
exten => did,n,Wait(10)
exten => did,n,StopPlaytones()
exten => did,n,BackGround(sound file)

did = the DID number as presented and note the '1' before Answer.

This works for me.

exten => 820055,1,Answer()
exten => 820055,n,PlayTones(ring)
exten => 820055,n,Wait(5)
exten => 820055,n,StopPlayTones()
exten => 820055,n,[do something interesting from now on]

That's my DID (820055) being answered first and then waiting for 5
seconds.  I use it for fax detect this way.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob
Smither
Sent: 18 December 2009 23:11
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Ringing for incoming call

Dear All,

I am using Asterisk 1.4 on CentOS 5.  I have an incoming DID provided by
Vitelity.  When the number is called it goes to my Asterisk box.  The
protocol is SIP.  This all works just fine if I answer the call and
begin a playback.

I want to let the number ring for a few seconds before it is answered,
and would like the caller to hear it ringing.  I have tried:

...
exten => s,n,Answer
exten => s,n,Playtones(ring)
exten => s,n,Wait(10)
exten => s,n,StopPlaytones()
exten => s,n,BackGround(sound file)
...

also

...
exten => s,n,Answer
exten => s,n,Ringing()
exten => s,n,Wait(10)
exten => s,n,BackGround(sound file)
...

I have also tried moving the Answer app to right before the BackGround
app.

In all cases when I call the number I never hear it ringing.  After the
10 second delay, the BackGround app does run.  Connecting to the CLI
does not give me any useful information - for example the Ringing app is
shown to run, but the caller does not hear it.

Any suggestions?

Many thanks!

-- 
Bob Smither 


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Ringing multiple instruments: sudden cutoff

[Mark Spencer]

> It all seems to work OK, with one unfortunate major exception: when the
> two phones are ringing, if I pick up the ATA on the local LAN here, I
> can talk for as long as I want and everything is cool.  But so far, in 4
> or 5 incoming calls, if I pick up the S100U attached across the net via
> IAX, after a certain variable amount of time (never more than 2 minutes)
> the call suddenly drops, with nothing more than the usual "Hangup"
> messages on the consoles of both instances of Asterisk.
>
> I have some other quality-related problems talking between the ATA here
> and the S100U there, which I will document in a separate mail.

If you run iax debug, can you send me a log of what you see?

Mark

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing after console dsp hangup

[Doug Lytle]

Jerry Geis wrote:
> Why is that and how can I stop it?
>   

I've never tried paging directly to the console, since it can introduce 
too much feedback.  Try recording the page and then play it back to the 
console:


exten => s,1,Set(active=${DB(paging/active)})
exten => s,n,GotoIf($["${active}" = "YES"]?7:4)

;
;* Set database entry for
;* paging active to YES
;

exten => s,n,Set(DB(paging/active)=YES)

;*
;* If paging currently in use,
;* jump to paging-inuse
;* context.
;*

exten => s,n,Goto(paging-inuse,s,1)


;**
;* Start recording to paging.gsm,
;* no longer then 30 seconds if
;* silence for 5 seconds, terminate
;* recording
;***

exten => s,n,Record(paging:gsm|5|30)
exten => s,n,Hangup()

;*
;* On hangup from paging, Playback paging file
;* then set paging/active to NO.
;*

exten => h,1,Dial(Console/dsp)
exten => h,n,Playback(paging)
exten => h,n,Set(DB(paging/active)=NO)

[paging-inuse]

exten => s,1,Congestion
exten => s,n,Hangup()

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 --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing after answered on zaptel

[Brodie Macleod]

Try setting:

progressinband=no

in your sip.conf

-Brodie

On Monday 14 August 2006 10:20 pm, Don Fanning wrote:
> Greetings List,
>
>
>
> I'm having a strange problem with my X100p card still ringing after the
> call is connected.  Any idea on how to solve this?
>
>
>
> Using latest asterisk (not svn) along with latest zaptel driver.
>
>
>
> Thanks,
> Don
___
--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

Re: [asterisk-users] Ringing after answered on zaptel

[Eric \"ManxPower\" Wieling]

That's kind of useless since progressinband only applies to digital 
interfaces.


Try callprogress=no

Brodie Macleod wrote:

Try setting:

progressinband=no

in your sip.conf

-Brodie

On Monday 14 August 2006 10:20 pm, Don Fanning wrote:

Greetings List,



I'm having a strange problem with my X100p card still ringing after the
call is connected.  Any idea on how to solve this?



Using latest asterisk (not svn) along with latest zaptel driver.




--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

___
--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

Re: [Asterisk-Users] Ringing an extension on multiple phones

[Alexander Lopez]

Title: Re: [Asterisk-Users] Ringing an extension on multiple phones






There are several options here.

You can set up a queue and have the phones ring un the order you like.

Setup an additional extension on every phone.

Set up an AGI script that allows them to login to the receptionist calls. That way they can turn it on and off when they want.

-Original Message-
From: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com 
Sent: Fri Jan 07 11:45:37 2005
Subject: [Asterisk-Users] Ringing an extension on multiple phones

I am using Cisco 7960 phones and have had a request to have the
receptionist phone ring on multiple phones just in case she is not around.

Call pickup is the theory here but the issue is that not all the people
that need to hear the ring would here the receptionist phone ring so I
think I need to have a second line appearance on the phones in question
so that line will ring.

Can this be done or is there a better way.

--
Scott Henderson

Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
http://www.finite-tech.com
http://www.chillywall.com
http://www.virtuale.cc
http://www.mphage.com
Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Ringing an extension on multiple phones

[Listas]

You can Dial() extension SIP/line1&SIP/line2

even more you can and that will call both extensions only after a 5 seconds
timeout
exten => xxx,1,Dial(SIP/line1,5)
exten => xxx,2,Dial(SIP/line1&SIP/line2,10)
etc...

that's if I understood what ou needed...

bye,
M.


- Original Message - 
From: "Scott Henderson" <[EMAIL PROTECTED]>
To: 
Sent: Friday, January 07, 2005 1:45 PM
Subject: [Asterisk-Users] Ringing an extension on multiple phones


> I am using Cisco 7960 phones and have had a request to have the
> receptionist phone ring on multiple phones just in case she is not around.
>
> Call pickup is the theory here but the issue is that not all the people
> that need to hear the ring would here the receptionist phone ring so I
> think I need to have a second line appearance on the phones in question
> so that line will ring.
>
> Can this be done or is there a better way.
>
> -- 
> Scott Henderson
>

> Finite Technologies Incorporated
> 3763 Image Drive, Anchorage, Alaska 99504
> Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
> http://www.finite-tech.com
> http://www.chillywall.com
> http://www.virtuale.cc
> http://www.mphage.com
> Current Local Time:
http://www.worldtimeserver.com/time.asp?locationid=US-AK
>

>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Ringing an extension on multiple phones

[Bill Seddon]

<>

Yes, and if the multiple extensions that ring are members of the same group
then any one of the phones can pickup the call.

So the next question is: how does the receptionist put the system into
"group ring" mode.  The answer is to have the receptionist call a nominated
number such as **221 (enable group ringing) and **222 (to disable group
ringing).

When the receptionist calls **221 a global variable (or an entry in the
registry is created) is made to contain a value that indicates group ringing
is in effect.  When **222 is called, calls ring on the operator extension.

We use a similar approach to have support calls forwarded to mobile phones
out of office hours.

Bill Seddon

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Listas
Sent: January 07, 2005 6:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Ringing an extension on multiple phones

You can Dial() extension SIP/line1&SIP/line2

even more you can and that will call both extensions only after a 5 seconds
timeout
exten => xxx,1,Dial(SIP/line1,5)
exten => xxx,2,Dial(SIP/line1&SIP/line2,10)
etc...

that's if I understood what ou needed...

bye,
M.


- Original Message - 
From: "Scott Henderson" <[EMAIL PROTECTED]>
To: 
Sent: Friday, January 07, 2005 1:45 PM
Subject: [Asterisk-Users] Ringing an extension on multiple phones


> I am using Cisco 7960 phones and have had a request to have the
> receptionist phone ring on multiple phones just in case she is not around.
>
> Call pickup is the theory here but the issue is that not all the people
> that need to hear the ring would here the receptionist phone ring so I
> think I need to have a second line appearance on the phones in question
> so that line will ring.
>
> Can this be done or is there a better way.
>
> -- 
> Scott Henderson
>

> Finite Technologies Incorporated
> 3763 Image Drive, Anchorage, Alaska 99504
> Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
> http://www.finite-tech.com
> http://www.chillywall.com
> http://www.virtuale.cc
> http://www.mphage.com
> Current Local Time:
http://www.worldtimeserver.com/time.asp?locationid=US-AK
>

>
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing on Console after a page

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Josiah Bryan wrote:

> [paging]
> exten => 249,1,Goto(paging,s,1)
> exten => s,1,Playback(beep)
> exten => s,n,Dial(Console/dsp)
> exten => s,n,Playback(vm-goodbye)
> exten => s,n,Hangup

If the caller has hung up, to whom are you playing the "vm-goodbye"
message?  Also, why the Goto?

[paging]
exten => 249,1,Playback(beep)
exten => 249,n,Dial(Console/dsp)

Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)

iD8DBQFIvvVQCFu3bIiwtTARAp9XAJ0Ra8LLo2COS89loyBFgWutV5SxcgCbB6Md
54ve7snza6SLYZ1ufR4BVJY=
=Y8MF
-END PGP SIGNATURE-

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing on Console after a page

[Josiah Bryan]

Good questions - the only answer to the Goto is that this was a legacy 
dialplan that I first wrote 3+ years ago when I first set up asterisk - 
and I havn't gone go back and re-work it after learning more about 
asterisk - it worked up till the upgrade to 1.4 and that was that.

However, you're right - simpler is better anyway. I changed it to the 
249,n,Dial(Console/dsp) format (as you described below) and it still 
plays the ringing indicator over the console after I hangup my phone.

As an aside, In the 3+ years that the system has been online, users know 
that when they dialed 249 and heard "Goodbye!" right away, they weren't 
going to be able to page and "Something was wrong." (Usually, someone 
had put 249 on hold or something like that.) Thats the primary reason I 
left the goodbye in there.

Anyway, thoughts on how to debug?

Thanks for your help and your suggestions.
-josiah



Barry L. Kline wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Josiah Bryan wrote:
> 
>> [paging]
>> exten => 249,1,Goto(paging,s,1)
>> exten => s,1,Playback(beep)
>> exten => s,n,Dial(Console/dsp)
>> exten => s,n,Playback(vm-goodbye)
>> exten => s,n,Hangup
> 
> If the caller has hung up, to whom are you playing the "vm-goodbye"
> message?  Also, why the Goto?
> 
> [paging]
> exten => 249,1,Playback(beep)
> exten => 249,n,Dial(Console/dsp)
> 
> Barry
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.5 (GNU/Linux)
> 
> iD8DBQFIvvVQCFu3bIiwtTARAp9XAJ0Ra8LLo2COS89loyBFgWutV5SxcgCbB6Md
> 54ve7snza6SLYZ1ufR4BVJY=
> =Y8MF
> -END PGP SIGNATURE-
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 

-- 
Josiah Bryan
IT Manager
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Ringing indication not working as expected

[Eric \"ManxPower\" Wieling]

"R" is not a valid Dial option.  "r" is the option you wanted.  HOWEVER, 
if you are not hearing ringback, "r" will almost never fixes the issue.


Make sure you have a /etc/asterisk/indications.conf   In some situations 
if you do not have that file you will not hear ringback.


Sebastian Kayser wrote:

Hi all,

are there any caveats regarding ringing indication with Asterisk?

I have got an asterisk installation with a quadBRI driven by BRIstuff.
Internal phones are various snoms (320 / 360) connected via SIP and
Idefisk softphones connected via IAX2. Outgoing calls are "routed"
through the Zap interfaces.

When i set up the action for an external extension as

Dial(Zap/g2/,60,R)

or

Dial(Zap/g2/,60)

and initiate an outgoing call, Asterisk tells me that the called party
is ringing (Zap/4-1 is ringing) but there is no ringing indicated to the
calling party. No matter whether the calling party is a snom hardphone
or an idefisk softphone.

Am i missing something?

asterisk*CLI> show version
Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1p

- Sebastian
___
--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




--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

___
--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

Re: [asterisk-users] Ringing agents cell as an alert?

[James Sharp]

On 01/03/2012 01:06 PM, Todd Routhier wrote:

Happy New Year to all!

Asterisk 1.8.x

I have a queue to which I add agent channels like SIP/300 dynamically
using the manager interface. Once logged in, there SIP/300 of course
rings when a call is distributed to them.

How can I also get the agents cell phone to ring without actually adding
it to the queue? I mean id I add something goofy like
SIP/MyProvider/1555444 to the queue, I don't know what will happen
at this point, haven't tested it. Even if it works (asterisk channel
state etc) it will mess with the queue and treat the cell phone like a
separate agent, messing up call distribution etc.

I am trying to be as clear as possible, sorry if my questions are cloudy.

Basically, I have the queue doing what I want right now, I just want to
add the ability to have an agent's cell phone ring as a means of
alerting them if they are away from their desk. If they can answer the
call and the queue will handle it just as they answered it from their
SIP device, that would be a bonus. I know this can all be done, just not
sure how to tackle it at the moment.

Any guidance would be appreciated, thanks in advance.


Perhaps blend the agent's SIP phone + cell phone together as a local 
channel and then add that local channel to the queue instead of SIP 
phone + cell phone.  Asterisk will see the local channel as one agent 
rather than two.


--
_
-- 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] Ringing agents cell as an alert?

[Todd Routhier]

Sounds perfect, I will need to look into how to blend them together like
that.

I wonder though, will channel state still work using that method? I think
it's needed by something in the queue but I can't remember at the moment.



On Tue, Jan 3, 2012 at 12:15 PM, James Sharp  wrote:

> On 01/03/2012 01:06 PM, Todd Routhier wrote:
>
>> Happy New Year to all!
>>
>> Asterisk 1.8.x
>>
>> I have a queue to which I add agent channels like SIP/300 dynamically
>> using the manager interface. Once logged in, there SIP/300 of course
>> rings when a call is distributed to them.
>>
>> How can I also get the agents cell phone to ring without actually adding
>> it to the queue? I mean id I add something goofy like
>> SIP/MyProvider/1555444 to the queue, I don't know what will happen
>> at this point, haven't tested it. Even if it works (asterisk channel
>> state etc) it will mess with the queue and treat the cell phone like a
>> separate agent, messing up call distribution etc.
>>
>> I am trying to be as clear as possible, sorry if my questions are cloudy.
>>
>> Basically, I have the queue doing what I want right now, I just want to
>> add the ability to have an agent's cell phone ring as a means of
>> alerting them if they are away from their desk. If they can answer the
>> call and the queue will handle it just as they answered it from their
>> SIP device, that would be a bonus. I know this can all be done, just not
>> sure how to tackle it at the moment.
>>
>> Any guidance would be appreciated, thanks in advance.
>>
>
> Perhaps blend the agent's SIP phone + cell phone together as a local
> channel and then add that local channel to the queue instead of SIP phone +
> cell phone.  Asterisk will see the local channel as one agent rather than
> two.
>
> --
> __**__**_
> -- 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] Ringing agents cell as an alert?

[James Sharp]

On 01/03/2012 01:22 PM, Todd Routhier wrote:

Sounds perfect, I will need to look into how to blend them together like
that.


Put them in extensions conf like so:

[agentblends]
exten => bob,1,Dial(SIP/300&SIP/12102263232@myprovider)

Then put Local/bob@agentblends into your queue.



I wonder though, will channel state still work using that method? I
think it's needed by something in the queue but I can't remember at the
moment.



Channel state for ringing/answer?  The state will be ringing until Bob 
either answers his cell phone (or his cell voicemail answers, that may 
be a quirk.  Adjust the "timeout" parameter on the dial command to work 
that out) or his SIP phone.


--
_
-- 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] Ringing Groups, SIP Forward and looping problem

[Robert Lister]

On Fri, Sep 28, 2007 at 09:57:52AM +0100, Russell Brown wrote:
> 
> I've a big problem with SIP forwarding back into 'ringing groups'
> creating what can only be described as call storms :-(
> 
> I have a 'ringing groups' of SIP phones with an effective dialplan (much
> simplified) like so:
> 
>   ;   Purchase ledger
>   [ptsn_inbound]
>   exten => _846061,1,Dial(Local/[EMAIL PROTECTED])

I am not sure why you are doing it like this but it seems awkward.

Relying on handset diverts seems fraught with danger as you can't be sure 
what's going to happen from a dialplan perspective.

Why don't you set up a queue in queues.conf strategy ringall:

[purchase]
; Dynamic group for users logging on in London Office
strategy = ringall
maxlen = 1
retry = 1
timeout = 20
musiconhold = default
joinempty = strict
leavewhenempty = yes
timeoutrestart = yes
member => SIP/110
member => SIP/111
member => SIP/112
member => SIP/113
member => SIP/114

Then route calls to that queue from the dialplan:-

exten => _846061,1,Queue(purchase|rn|||40)
...

[...variety of options you can do here if there is no answer all busy
 in the queue etc, see variable ${QUEUESTATUS}. Here's what I've got:-
 
exten => s,n,GotoIf($["${QUEUESTATUS}" = "UNKNOWN"]?200)
exten => s,n,GotoIf($["${QUEUESTATUS}" = "BUSY"]?200)
exten => s,n,GotoIf($["${QUEUESTATUS}" = "FULL"]?200)
exten => s,n,GotoIf($["${QUEUESTATUS}" = "JOINUNAVAIL"]?200)
exten => s,n,GotoIf($["${QUEUESTATUS}" = "LEAVEUNAVAIL"]?200)
exten => s,n,GotoIf($["${QUEUESTATUS}" = "LEAVEEMPTY"]?200)
exten => s,n,GotoIf($["${QUEUESTATUS}" = "TIMEOUT"]?200)

]

Then you could set up some features in the dial plan to allow your users
to go in and out of the group as required. Something like:-

exten => _*71,2,Macro(togglegroup,${CALLERID(num)})

( *71 will toggle in and out of group, so you could program a button on
  your phones for example, to set them in and out of group. This set of 
  macros keeps track for each user in and out group state and toggles
  it in and out. It keeps track of it with a db variable.)


[macro-outofgroup]
exten => s,1,NoOp("macro-outofgroup reached: ${ARG1}")
exten => s,n,NoOp( -- DND pausing queue member:  Local/${ARG1} --- )
exten => s,n,PauseQueueMember(|Local/[EMAIL PROTECTED])
exten => s,n,Set(DB(${ARG1}/outofgroup)=1)
exten => s,n,Answer
exten => s,n,Playback(extras/dnd-out-of-group)
exten => s,n,Hangup

[macro-ingroup]
exten => s,1,NoOp("macro-ingroup reached: ${ARG1}")
exten => s,n,NoOp( -- DND unpausing queue member:  Local/${ARG1} --- )
exten => s,n,UnPauseQueueMember(|Local/[EMAIL PROTECTED])
exten => s,n,DBdel(${ARG1}/outofgroup)
exten => s,n,Answer
exten => s,n,Playback(extras/dnd-now-in-group)
exten => s,n,Hangup

[macro-togglegroup]
exten => s,1,NoOp("macro-togglegroup reached: ${ARG1}")
exten => s,n,GotoIf($["${DB(${ARG1}/outofgroup)}" = ""]?900)
exten => s,n,Macro(ingroup,${ARG1})
exten => s,n,Hangup

exten => s,900,Macro(outofgroup,${ARG1});
exten => s,n,Hangup

(I've got those sounds if you want them, let me know, if you don't mind 
plummy british accent we re-recorded all our sounds files in, plus a few 
custom ones, or you could just play a tone so the user knows the group 
action has been carried out.)

Let me know if this is any use to you.


Regards,


Rob


-- 
Robert Lister - London Internet Exchange - http://www.linx.net/
sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ringing Groups, SIP Forward and looping problem

[Robert Lister]

Whoops! Forgot to change it for SIP devices. 

Of course you need to change your queue member devices to SIP and not 
Local/${ARG1} as I've got agents and other complications in mine.

You might need a context or not, see what happens!

Rob

Here is corrected version (I think will work, untested though!)

> [macro-outofgroup]
> exten => s,1,NoOp("macro-outofgroup reached: ${ARG1}")
> exten => s,n,NoOp( -- DND pausing queue member:  SIP/${ARG1} --- )
> exten => s,n,PauseQueueMember(|SIP/${ARG1})
> exten => s,n,Set(DB(${ARG1}/outofgroup)=1)
> exten => s,n,Answer
> exten => s,n,Playback(extras/dnd-out-of-group)
> exten => s,n,Hangup
> 
> [macro-ingroup]
> exten => s,1,NoOp("macro-ingroup reached: ${ARG1}")
> exten => s,n,NoOp( -- DND unpausing queue member:  SIP/${ARG1} --- )
> exten => s,n,UnPauseQueueMember(|SIP/${ARG1})
> exten => s,n,DBdel(${ARG1}/outofgroup)
> exten => s,n,Answer
> exten => s,n,Playback(extras/dnd-now-in-group)
> exten => s,n,Hangup
> 
> [macro-togglegroup]
> exten => s,1,NoOp("macro-togglegroup reached: ${ARG1}")
> exten => s,n,GotoIf($["${DB(${ARG1}/outofgroup)}" = ""]?900)
> exten => s,n,Macro(ingroup,${ARG1})
> exten => s,n,Hangup
> 
> exten => s,900,Macro(outofgroup,${ARG1});
> exten => s,n,Hangup


-- 
Robert Lister - London Internet Exchange - http://www.linx.net/
sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Ringing() doesn't play sound while phone is ringing

[Bart Coppens]

try
exten => 101,2,Dial(Zap/1,10,r)
in stead of
exten => 101,2,Dial(Zap/1,10)
BC



From: "Warren Burstein" <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Ringing() doesn't play sound while phone is 
ringing
Date: Wed, 11 Aug 2004 15:22:45 +0400

I have:
RedHat 9.0
TDM40B
asterisk-0.9.0 compiled from sources
zaptel-0.9.1 likewise

/etc/zaptel.conf contains
fxoks=1-4
loadzone = us
defaultzone=us

loaded modules zaptel and wcfxs

/etc/askterisk/zapata.conf contains
[channels]
language = en
signalling = fxo_ks
context = phones
channel => 1-4

/etc/askterisk/extensions.conf contains
[general]
static=yes
writeprotect=yes
[phones]
exten => 101,1,Ringing()
exten => 101,2,Dial(Zap/1,10)
exten => 101,3,Congestion

I also uncommented the "noload => chan_oss.so" in 
/etc/asterisk/modules.conf
because I don't have a sound card.  Other than that, all conf files are the
originals from "make samples".


But when I dial 101 from Zap/2, Zap/1 rings (and if I pick it up, I can 
have
a conversation with myself), but I don't hear a ringing tone out of Zap/2.
I commented out the Dial and Congestion, and then I heard a two ringing
tones, a click, and a congestion tone, while the console said:


pbx.c:1836 ast_pbx_run: Timeout, but no rule 't' in context 'phones'

I'm guessing that Dial stops Ringing.  How do I tell Ringing to continue
while Dial is working, and if it isn't stopped by Dial, not to time out
after two rings?

"show application ringing" doesn't describe any parameters to Ringing() .

Thanks.
_
Make your own website http://webbuilder.msn.be
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Ringing() doesn't play sound while phone is ringing

[Warren Burstein]

That did it.  I thought Ringing() did that, but I guess it's just for when
you want to fake a ringing tone.  I'll add a comment to
http://www.voip-info.org/wiki-Asterisk+cmd+Ringing.

thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bart Coppens
Sent: Wednesday, August 11, 2004 5:28 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Ringing() doesn't play sound while phone is
ringing

try
exten => 101,2,Dial(Zap/1,10,r)

instead of
exten => 101,2,Dial(Zap/1,10)

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Ringing a group of phones but not if they arebusy

[Chris Bagnall]

>   I need to ring a group of 8 phones, but not if they are already on 
> another call.  How can I determine which of those 8 phones are busy so 
> I only ring the others?

I've done this in the past by disabling call waiting on the phones and put
all 8 phones into a "ringall" queue. Then, when you call that queue, the
phones already on calls return SIP BUSY,whilst the others ring as normal.

It's not perfect, but for most of our users the call waiting noise in the
earpiece is an annoyance anyway.

Hope that helps.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


___
--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

Re: [asterisk-users] Ringing a group of phones but not if they are busy

[Steven Ringwald]

Carlos Chavez wrote:

I need to ring a group of 8 phones, but not if they are already on
another call.  How can I determine which of those 8 phones are busy so I
only ring the others?



chanIsAvail

Steve

___
--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

Re: [asterisk-users] Ringing a group of phones but not if they are busy

[Carlos Chavez]

On Fri, 2006-11-17 at 16:03 -0800, Steven Ringwald wrote:
> Carlos Chavez wrote:
> > I need to ring a group of 8 phones, but not if they are already on
> > another call.  How can I determine which of those 8 phones are busy so I
> > only ring the others?
> 
> 
> chanIsAvail
> 

The problem with ChanIsAvail is that if ig give it a line like this:

s,1,ChanIsAvail(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105&SIP&106)

the resulting variable only lists the first available channel and not
all the available channels so I cannot ring all the available channels.

-- 
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chàvez Prats
Director de Tecnologìa
+52-55-91169161 ext 2001


signature.asc
Description: This is a digitally signed message part
___
--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

Re: [asterisk-users] Ringing a group of phones but not if they are busy

[Alberto Pastore]

Chris Bagnall ha scritto:
	I need to ring a group of 8 phones, but not if they are already on 
another call.  How can I determine which of those 8 phones are busy so 
I only ring the others?



I've done this in the past by disabling call waiting on the phones and put
all 8 phones into a "ringall" queue. Then, when you call that queue, the
phones already on calls return SIP BUSY,whilst the others ring as normal.

It's not perfect, but for most of our users the call waiting noise in the
earpiece is an annoyance anyway.

Hope that helps.

Regards,

Chris
  

If you disable call waiting, then you don't need a queue.
With grandstream gxp-2000 phones, calling
Dial(SIP/phone1&SIP/phone2&SIP/phone3)
rings only off-hook phones.

However, I have also SPA-941 phones.
Is it possible to disable the call waiting feature on Linksys SPA-941?
I haven't succeeded so far... and the multiple Dial() method or
the Queue are not working either.

I had to change my extensions.conf macro to do ChanIsAvail sequentially,
that is, for each phone I call ChanIsAvail and then check the results
to see if the phone is busy. If not, I add it to the dialstring to
pass to Dial() eventually.

Since there are 10 phones to check, and the process is not atomic,
it can (very rarely) occur that a phone is included in the dialstring
but has just become busy, and the user gets the annoying call waiting
tone.

Any clue?

--
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508 
Fax 0321-492974

http://www.msoft.it

___
--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

Re: [asterisk-users] Ringing a group of phones but not if they are busy

[C F]

Try something like this:

exten => s,1,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]&Local/[EMAIL 
PROTECTED])

[callphones]
exten => _X.,1,ChanIsAvail(Sip/${EXTEN},js)
exten => _X.,2,Dial(Sip/${EXTEN})
exten => _X.,102,Noop(${EXTEN} is on a call)


On 11/17/06, Carlos Chavez <[EMAIL PROTECTED]> wrote:

I need to ring a group of 8 phones, but not if they are already on
another call.  How can I determine which of those 8 phones are busy so I
only ring the others?

--
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chàvez Prats
Director de Tecnologìa
+52-55-91169161 ext 2001


___
--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

Re: [asterisk-users] Ringing (180) no SDP to progress(183) with SDP transition => no audio.

[Richard Mudgett]

On Mon, Nov 20, 2017 at 7:31 AM, Benoit Panizzon 
wrote:

> Dear List
>
> I am testing various early audio scenarios with different voice IC's,
> phones and pbxes.
>
> In Switzerland, when you operate a value added number, you have to
> announce the price of the call, usually in early audio, before the call
> is established.
>
> In 'dialplan' terms this would be:
>
> exten => XX,1,Ringing
> exten => XX,n,Wait(15)
> exten => XX,n,Progress
> exten => XX,n,Playback(price-announce,noanswer)
> exten => XX,n,Wait(5)
> exten => XX,n,Answer
>
> I see the asterisk playing the early announcement audio in the rtp
> stream. Some devices (arris EMTA) calling the asterisk also do play it
> to the caller.
>
> But!
>
> Most other devices I have tested just keep playing the locally generated
> ringtone despite getting an 183 with SDP and the announcement is never
> to be heard by the caller.
>
> If I do to force inband ringback tone, this works with all devices I
> have tested so far.
>
> exten => XX,1,Progress
> exten => XX,n,Ringing
> exten => XX,n,Wait(15)
> exten => XX,n,Playback(price-announce,noanswer)
> exten => XX,n,Wait(5)
> exten => XX,n,Answer
>
> Is anything wrong with the transition of ringing without SDP (to have
> the local device generating ringback tone) and then start sending early
> audio with 183?
>

Both orderings of Ringing and Progress are valid.  It is up to the calling
device to handle it.  As you have seen, there is quite a difference in
how devices handle it.  I have even seen where the calling device needs
Ringing before Progress to handle the call correctly.  I think that case was
because the device was converting ISDN to SIP.  I do think that the devices
that don't stop local ringback in favor of the incoming RTP stream following
the 183 are broken.  Unfortunately it is something that is out of your
control.

Richard
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users