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

[Benoit Panizzon]

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?

-Benoît Panizzon-
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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 efes9...@hotmail.com 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
  

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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



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Re: [asterisk-users] ringing in queues

[Ishfaq Malik]

On 13 March 2015 at 14:04, Matt Hamilton efes9...@hotmail.com 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

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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
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[asterisk-users] ringing in queues

[Matt Hamilton]

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
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Re: [asterisk-users] Ringing issue

[D'Arcy J.M. Cain]

On Tue, 13 May 2014 15:28:26 +0100
Gareth Blades mailinglist+aster...@dns99.co.uk 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

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Re: [asterisk-users] Ringing issue

[D'Arcy J.M. Cain]

On Tue, 13 May 2014 15:28:26 +0100
Gareth Blades mailinglist+aster...@dns99.co.uk 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

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[asterisk-users] Ringing issue

[D'Arcy J.M. Cain]

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.

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

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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.



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[asterisk-users] Ringing detection ?

[Nicolas Ross]

Hi !

I have an application where I originate a call with a call file and play 
some pre-recorded message when the person answers. And it's working 
correctly.


Now, I've been asked to add the support for extenstion numbers.

I've been able to actualy send the extension numbver via the SendDTMF 
command. It works perfectly. But after the dtmf have been sent, the 
dialplan shoud wait for someone to answer.


The problem is that during that time, the phone is rining. So withing 
north-america, it's typicly ring 2 secs, silence 4 seconds.


So, I could use AMD() WaitForSilence(4100) for exemple. But that would 
require the person at the other end to be silent for 4 seconds. That's 
unrealistic.


So, I'm searching for a way for my dialplan to detect ringing and only 
lauch Amd/WaitforSilence after the person answers...


My curent ael dialplan for my originated call is :

  500 = {
Answer();
Wait (1);
if (${LEN(${noPoste})})
{
  SendDTMF(${noPoste});
}
Background(silence/1);
AMD();
WaitForSilence(500);
for (x=0; ${x}  3; x=${x} + 1) {
  Background(outcall/outcall-${idJob});
  Background(outcall-confirm);
  WaitExten(5);
};
goto diffuseurappel|3|1;
  };

Any ideas ?

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

[Todd Routhier]

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.
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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.


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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 ja...@fivecats.org 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.

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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/300SIP/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.


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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...@carrierSIP/ukgeonum...@carrierSIP/PCSU200SIP/PCSU201SIP/PCSU202SIP/PCSU203SIP/PCSU204SIP/PCSU205SIP/PCSU206|15
 *** 9. row ***
  context: pcsu-Identifier-work
exten: s
 priority: 2
  app: Dial
  appdata: 
 SIP/ukgeonum...@carrierSIP/ukgeonum...@carrierSIP/ukgeonum...@carrierSIP/PCSU200SIP/PCSU201SIP/PCSU202SIP/PCSU203SIP/PCSU204SIP/PCSU205SIP/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

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[asterisk-users] Ringing issue

[Ishfaq Malik]

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...@carrierSIP/ukgeonum...@carrierSIP/PCSU200SIP/PCSU201SIP/PCSU202SIP/PCSU203SIP/PCSU204SIP/PCSU205SIP/PCSU206|15
*** 9. row ***
 context: pcsu-Identifier-work
   exten: s
priority: 2
 app: Dial
 appdata: 
SIP/ukgeonum...@carrierSIP/ukgeonum...@carrierSIP/ukgeonum...@carrierSIP/PCSU200SIP/PCSU201SIP/PCSU202SIP/PCSU203SIP/PCSU204SIP/PCSU205SIP/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
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

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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...@carrierSIP/ukgeonum...@carrierSIP/PCSU200SIP/PCSU201SIP/PCSU202SIP/PCSU203SIP/PCSU204SIP/PCSU205SIP/PCSU206|15
 *** 9. row ***
  context: pcsu-Identifier-work
exten: s
 priority: 2
  app: Dial
  appdata: 
 SIP/ukgeonum...@carrierSIP/ukgeonum...@carrierSIP/ukgeonum...@carrierSIP/PCSU200SIP/PCSU201SIP/PCSU202SIP/PCSU203SIP/PCSU204SIP/PCSU205SIP/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

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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 smit...@c-c-i.com


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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 smit...@c-c-i.com 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 smit...@c-c-i.com wrote:
   Dear All,
 
 snip
 
  
   ...
   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.
 
 snip
 
 i.e., after the Wait, but still no joy.
 
 Anything else I need to look at?
 
 Thanks,
 -- 
 Bob Smither smit...@c-c-i.com
 
 
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-- 
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

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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=provider host
context=inbound 
username=user name
secret=my 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)


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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
-- Local/s...@cci-7c61,2 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!



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[asterisk-users] Ringing for incoming call

[Bob Smither]

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 smit...@c-c-i.com


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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 smit...@c-c-i.com 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 smit...@c-c-i.com


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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 smit...@c-c-i.com wrote:
  Dear All,

snip

 
  ...
  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.

snip

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

Anything else I need to look at?

Thanks,
-- 
Bob Smither smit...@c-c-i.com


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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 smit...@c-c-i.com 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 smit...@c-c-i.com wrote:
  Dear All,

 snip

 
  ...
  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.

 snip

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

 Anything else I need to look at?

 Thanks,
 --
 Bob Smither smit...@c-c-i.com


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[asterisk-users] ringing... or lack thereof

[Jeff LaCoursiere]

Want to make sure I understand why a caller might not hear ringing when 
outbound calling.

A SIP phone is behind a firewall and is registered to an asterisk 
server on a public network.  Sometimes (but not always) when placing an 
outbound call there is no ringing before the remote party answers.  Its 
not that the remote party picks up very quickly - the delay may be as long 
as twenty seconds before it is answered (and I must assume that the remote 
phone had rung at least a few times).

I vaguely understand that part of the SIP call setup includes a ringing 
message, sent from asterisk to the originating phone.  If this is correct 
and the firewall for whatever reason isn't passing this message, will 
there be no ringing sound on the originating phone?  This is confusing to 
me, as I kind of assumed that the ringing sound was in audio, and would be 
part of the RTP stream.  But perhaps that isn't even flowing yet.

Guess I am showing my SIP ignorance.  Please enlighten ;)

Cheers,

j

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[asterisk-users] Ringing after console dsp hangup

[Jerry Geis]

I have a simple context that connects to the console dsp which works,
but then after I hangup I hear ringing on the console dsp. It rings 
until I stop asterisk.

Why is that and how can I stop it?

Thanks,
Jerry

[paging]
exten = s,1,Answer
exten = s,n,Playback(beep)
exten = s,n,Dial(Console/dsp)
exten = s,n,Hangup
exten = h,1,Hangup


   -- Executing [EMAIL PROTECTED]:1] Answer(SIP/192.168.1.8-089177a8, ) in 
new stack
-- Executing [EMAIL PROTECTED]:2] Playback(SIP/192.168.1.8-089177a8, 
beep) in new stack
-- SIP/192.168.1.8-089177a8 Playing 'beep' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Dial(SIP/192.168.1.8-089177a8, 
Console/dsp) in new stack
  Call placed to 'dsp' on console 
  Auto-answered 
-- Called dsp
-- ALSA/default answered SIP/192.168.1.8-089177a8
  Hangup on console 
  == Spawn extension (paging, s, 3) exited non-zero on 
'SIP/192.168.1.8-089177a8'
-- Executing [EMAIL PROTECTED]:1] Hangup(SIP/192.168.1.8-089177a8, ) in 
new stack
  == Spawn extension (paging, h, 1) exited non-zero on 
'SIP/192.168.1.8-089177a8'


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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.


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[asterisk-users] Ringing on Console after a page

[Josiah Bryan]

Hello, all -

Alright, after my fun with Asterisk crashing, I'm onto my next item in 
my checklist of stuff-to-fix-after-upgrading. I've noticed a very 
troubling problem when paging over Console/dsp.

(I'm not sure if this has anything to do with the Dial oddities that I 
experienced with the Crashing problem in my other thread or not...)

The problem is that after the user dials the extension, connects, speaks 
their page, hangsup, ringing is heard over the paging system (as in, the 
tone heard when you dial a person and you hear the phone ringing - that 
ringing tone - I don't know the proper term for it, but you get the 
drift.)

I've gone through the source code, trying to figure out what it could be 
doing - however, since this is the first time I've really looked at the 
source for asterisk, I really didn't know what to look for.

Here's the relevant context (which is included in a general context for 
all users):

[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

Here's the console output when I dial extension 249 to page. (I dial, 
paging answers, I say whatever (or even just hangup immediately) - then, 
right after the call termination, I hear the ringing over the paging 
system. I have to *manually* issue then hangup command seen below to 
stop it from ringing - however, the oddest thing is asterisk tells me 
that there is no call to hangup. Its not like the console got transfered 
to any extension - literally no channels active while the ringing is 
taking place (core show channels reports 0 active channels even while 
the ringing is heard.)

asterisk*CLI set verbose 99
Verbosity is at least 99
 -- Zap/1-1 answered SIP/236-09f0ea20
asterisk*CLI set debug 99
Core debug was  and is now 99
asterisk*CLI
 -- Executing [EMAIL PROTECTED]:1] Goto(SIP/josiah2-09f0ea20, 
paging|s|1) in new stack
 -- Goto (paging,s,1)
 -- Executing [EMAIL PROTECTED]:1] Playback(SIP/josiah2-09f0ea20, beep) 
in new stack
 -- SIP/josiah2-09f0ea20 Playing 'beep' (language 'en')
 -- Executing [EMAIL PROTECTED]:2] Dial(SIP/josiah2-09f0ea20, 
Console/dsp) in new stack
   Call placed to 'dsp' on console 
   Auto-answered 
 -- Called dsp
 -- ALSA/default answered SIP/josiah2-09f0ea20
   Hangup on console 
   == Spawn extension (paging, s, 2) exited non-zero on 
'SIP/josiah2-09f0ea20'
Really destroying SIP dialog '[EMAIL PROTECTED]' 
Method: BYE
asterisk*CLI  hangup
No call to hangup up


I'm open to any and all suggestions.

Thanks for your time and patience!

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


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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-

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


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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

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

[Russell Brown]

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])



[groups]
exten = 6061,1,Macro(QUEUEING_GROUP_WITH_NS,${EXTEN},Purchase)



[macro-QUEUEING_GROUP_WITH_NS]
...
exten = s,n,Dial(Sip/110Sip/111Sip/112Sip/113Sip/114)
...


If Sip/110 sets their SIP phone (SNOM 300 FWIW) to call forward to 6061
then all seems fine and calls to 110 end up in the group.

If Sip/113 *also* sets their SIP phone to call forward to 6061 then
Asterisk seems to get into a state where the calls bounce around,
ringing the phones but seemingly not allowing the call to be answered.

A 'restart now' is the only way out while this call storm is in
progress.

I'm guessing that having two SIP phones redirecting back into the
ringing group is what's causing the problem but can't think of a way
around it.

Can anyone suggest a cure?

-- 
 Regards,
 Russell
 
| Russell Brown  | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com  |
| Peterborough, England  | WWW Play: http://www.ruffle.me.uk |
 

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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

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[asterisk-users] Ringing sound doesn't work

[Simon Perreault]

Hi,

I have these extensions:

exten = 101,1,Dial(SIP/101,15)
exten = 102,1,Dial(SIP/102,15)
exten = 0,1,Dial(SIP/101SIP/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.

Thanks,
Simon
[globals]
SIPTRUNK=418555
IAXTRUNK=514555

[default]
exten = s,1,Answer()
exten = s,2,Background(viagenie)
exten = s,3,WaitExten()

exten = i,1,Background(invalid)
exten = i,n,Goto(s,1)

exten = t,1,Background(please-try-again)
exten = t,n,Goto(s,1)

[phones]
exten = 101,1,Dial(SIP/101,15)
exten = 101,n,Goto(201,1)

; Simon
exten = 102,1,Dial(SIP/102,15)
exten = 102,n,Voicemail(102)

exten = 201,n,Dial(SIP/[EMAIL PROTECTED],15)
exten = 201,n,Voicemail(101)

[ivr]
exten = 0,1,Dial(SIP/101SIP/102,15,r)
exten = 0,n,Goto(201,1)

exten = 8,1,Directory(default)
exten = #,1,Directory(default)
exten = 500,1,VoiceMailMain()

[voip_incoming]
exten = ${SIPTRUNK},1,Goto(s,1)

[voip_outgoing]
exten = _NXX,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _NXX,2,Dial(SIP/[EMAIL PROTECTED])

exten = _NXXNXX,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _NXXNXX,2,Dial(SIP/[EMAIL PROTECTED])

exten = _1NXXNXX,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _1NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten = _9NXX,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _9NXX,2,Dial(SIP/418${EXTEN:[EMAIL PROTECTED])

exten = _9NXXNXX,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _9NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten = _91NXXNXX,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten = _9.,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

exten = N11,1,Set(CALLERID(all)=Viagenie (418-555-))
exten = N11,2,Dial(SIP/[EMAIL PROTECTED])

[external]
include = default
include = phones
include = ivr
include = voip_incoming

[internal]
include = external
include = voip_outgoing
exten = 10,1,Goto(s,1)
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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/101SIP/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.

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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!

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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...
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[asterisk-users] Ringing Volume

[Jadrien Wauthier]

Hi,
 
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.
 
Thanks.
 

Jad

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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.

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[asterisk-users] Ringing Volume

[Jadrien Wauthier]

 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

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[asterisk-users] Ringing does not terminate on mISDN after pickup

[Arik Raffael Funke]

Hello,

I am having something of an odd problem: about every 100 calls or so, 
when a call comes in via an external mISDN interface and I route it to 
an internal mISDN interface by dialing an internal msn that is 
programmed for multiple phones on the internal bus, somtimes the other 
phones continue ringing for several minutes after the call has already 
been picked up by one (or even eventually hungup already...). As you 
imagine this is really annoying...


I don't seem to be able to narrow it down in any way: show channels is 
empty (provided, call was answered an hungup, other phones continue ringing)


The only way to terminate the ringing is: wait for several minutes... :-)
or: restart asterisk.

Can anybody point me in the right direction with this issue? Every 
attempt to get rid of the problem has failed. I have no clue what else I 
could try... - Does anybody else experience this as well?


Cheers,
Arik

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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

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[asterisk-users] Ringing oddity/stupidity

[J. Oquendo]

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)
exten = s,3,Dial(SIP/201,25,tr)
exten = 
s,4,DIal(SIP/211SIP/202SIP/203SIP/209SIP/211SIP/212SIP/213SIP/214,15,tr)

exten = s,5,Background(/etc/asterisk/day)
exten = s,6,Wait(3)
exten = s,7,Voicemail([EMAIL PROTECTED])
exten = s,8,Hangup

[main-night-aa]
exten = s,1,Answer
exten = s,2,Background(/etc/asterisk/night)
exten = s,3,Voicemail([EMAIL PROTECTED])
exten = s,4,Hangup
exten = 1,1,Directory(fakepbxname,internal,l)
exten = 00,1,VoicemailMain([EMAIL PROTECTED])

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


Asterisk 1.2.13 built by root @ fakepbxname on a i686 running Linux (FC5)

Any thoughts?

--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 


The happiness of society is the end of government.
John Adams



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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


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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/phone1SIP/phone2SIP/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

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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


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

[Carlos Chavez]

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


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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

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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/100SIP/101SIP/102SIP/103SIP/104SIP/105SIP106)

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


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[asterisk-users] Ringing phones

[Matt]

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!
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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!
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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  *



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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!
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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  *



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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
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 Those that sacrifice essential liberty to obtain a little temporary safetydeserve neither liberty nor safety.-- Ben Franklin (1759)  *Doug Crompton *
 *Richboro, PA 18954* *215-431-6307* ** * [EMAIL PROTECTED]* * 
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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
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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!
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  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  *
  
 
 
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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!
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   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  *
   
  
  
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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.
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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.
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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
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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.




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[asterisk-users] Ringing after answered on zaptel

[Don Fanning]

Greetings List,



Im 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








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[asterisk-users] Ringing all extensions

[J. Oquendo]

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/1201SIP/1202SIP/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/1201SIP/1202SIP/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.




--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
GPG Key ID 0x1383A743
Fingerprint:
7B02 28CF 24D3 ACA7 9907  789A 8772 7736 1383 A743

sil . infiltrated @ net
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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/1201SIP/1202SIP/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/1201SIP/1202SIP/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
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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/1201SIP/1202SIP/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/1201SIP/1202SIP/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.






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RE: [asterisk-users] Ringing timer

[Michel Zenone]

Ok.Thanks a lot! I will try!

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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 asterisk-users@lists.digium.com
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?


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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
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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 






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[asterisk-users] Ringing timer

[Zenone]

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


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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


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RE: [asterisk-users] Ringing timer

[Zenone]

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




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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.


--
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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 asterisk-users@lists.digium.com
Objet: Re: [asterisk-users] Ringing timer
Date: 26/07/06 12:54

 Zenone wrote:
 gt; But my question was, is it possible to free the channel if it rings
too
 gt; long?
 
 Yes.  quot;show application dialquot; 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?


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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 asterisk-users@lists.digium.com
Objet: Re: [asterisk-users] Ringing timer
Date: 26/07/06 12:54


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

too

gt; long?

Yes.  quot;show application dialquot; 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


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!DSPAM:500,44c76db5240132002735277!



--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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[Asterisk-Users] Ringing to Outside Line

[casasterisk]

Hello All,

I have the latest Asterisk and am using the TDM400 card with 4 FXO ports.  One 
problem I have is that I want one of my Queues to go to a cell phone if all 
agents for the queue don't answer.

The problem is that once the TDM400 goes to a POTS line it does not time out 
after 15 seconds like I have configured (and like the other agents), it will 
ring to voicemail which doesn't allow me to send the call to a special 
voicemail box that I have set up for just his purpose.

I read somewhere that the only way to recognize if a POTS line is busy/no 
answer/answer is if I get an ISDN line but that just seems unusual for such a 
powerful PBX system to not be able to detect that it has been ringing a POTS 
line for 15 seconds and it's time to hang up and try the next in queue.

The major problem with this is that I want it to fail over to a second and 
third cell phone on no answers and then capture to a voicemail box - obviously 
it just rings the first phone and goes to the cell phones voicemail.

Any suggestions?

Thank you,
Craig

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[Asterisk-Users] Ringing indication not working as expected

[Sebastian Kayser]

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/number,60,R)

or

Dial(Zap/g2/number,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
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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/number,60,R)

or

Dial(Zap/g2/number,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
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[Asterisk-Users] Ringing extensions in a call group.

[Brian McCarey]

Hi 
all,

I've got an Asterisk 
at home system running the new Free PBX front. It's solved all our small office 
VOIP phone system which we are using as our only source of telephone 
communications.

Anyway,I have 
set up a few ring groups.

The first rings the 
internal office extensions. After 15 seconds it switches to the second ring 
group which rings two cell (mobile) numbers and should ring the same extensions 
as in the first ring group

Ring group 1: 
extension 100, extension 101
Ring group 2: group 
200, group 201, extension 100, extension 101
Ring group 200: cell 
number A
Ring group 201: cell 
number B

All groups are set 
to ringall

When an incoming 
call arrives, extension 100 rings and after a delay of 5 seconds or so, 
extension 101 joins in. 10 seconds later both extensions stop and cell number A 
starts ringing.

The problem is that 
Cell number B or extensions 100 and 101do not ring.

I want 200, 201, 
Extension 100  101 all to ring together while on ring group 
2.

Why does asterisk 
not do this?

Kind 
regards

Brian
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[Asterisk-Users] ringing indication in handset when 2 extensions answer simultaneously?

[asterisk]

I have a bunch of grandstream gxp2000's in the office.

When I Dial() multiple extensions at once, eg
Dial(SIP/2000SIP/2001SIP/2002) and two extensions answer simultaneously,
one of them gets the call and the other hears a ringing indication in the 
handset.

Anyone else seen this bug?

-Dan
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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...???





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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...???


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[Asterisk-Users] Ringing Delay

[chan \(Alpha Trilogies Networls\)]

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...???





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[Asterisk-Users] Ringing a few phones

[Shidan]

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.
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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/5000SIP/5001SIP?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.
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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/5000SIP/5001SIP?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.
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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/5000SIP/5001SIP?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

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[Asterisk-Users] ringing CAPI SIP channels together

[Dimitris Kounalakis]

Hello,
I have an asterisk box with ISDN (CAPI) phones (using chan_capi that has 
them as Zap channels) and SIP phones connected to it. I want to have a 
extension that rings 1 CAPI and 2 SIP phones together as a group.

I tried the following but when either the capi phone either one SIP 
phone is busy, asterisk goes directly to the BUSY extension.

Any idea how to overcome this ?
Thank you in advance,
Dimitris
exten = s,1,Dial(Zap/g5/111694SIP/g4,20,Ttwr)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(su111694)
exten = s-NOANSWER,2,Hangup   
exten = s-BUSY,1,Voicemail(sb111694) 
exten = s-BUSY,2,Hangup  
exten = s-CHANUNAVAIL,1,Playtones(dial)
exten = s-CHANUNAVAIL,2,Hangup
exten = s-CONGESTION,1,Playtones(congestion)
exten = s-CONGESTION,2,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(111694)  
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup

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[Asterisk-Users] Ringing an extension on multiple phones

[Scott Henderson]

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

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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 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


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

[Listas]

You can Dial() extension SIP/line1SIP/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/line1SIP/line2,10)
etc...

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

bye,
M.


- Original Message - 
From: Scott Henderson [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
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



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

[Bill Seddon]

You can Dial() extension SIP/line1SIP/line2

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/line1SIP/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/line1SIP/line2,10)
etc...

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

bye,
M.


- Original Message - 
From: Scott Henderson [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
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



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[Asterisk-Users] ringing after hangup

[Jeremy Jones]

Hello,

I'm having a bit of trouble getting asterisk + tdm400p + rhino channel
bank set up just right.  The problem is a single ring on a telephone
connected to the channel bank after the line is hung up.  I've seen a
couple messages similar to this, but I haven't seen any resolutions to
the problem.

Using the tdm400p card, with the first port connected to a 24-port fxs
rhino channel bank; the second port connected to our provider's pri
line, here's /etc/zaptel.conf:

span=1,0,0,esf,b8zs
span=2,1,0,esf,b8zs
fxoks=1-24
bchan=36-47
dchan=48
loadzone=us
defaultzone=us

And /etc/asterisk/zapata.conf looks like this:

[channels]
group=1
context=default
signalling=pri_cpe
switchtype=ni1
echocancel=yes
echocancelwhenbridged=yes
callerid=asreceived
echotraining=400
channel=36-47

context=system
signalling=fxo_ks
echocancel=yes
echocancelwhenbridged=yes
callerid=asreceived
echotraining=400
threewaycalling=yes
transfer=yes
callwaiting=yes
callwaitingcallerid=yes
;
callerid = 1 1
channel = 1
;
callerid = 2 2
channel = 2
;
.. and so on...

I have tried switching the signaling from fxo_ks to fxo_ls.  When I do
that, there's no ringing after the phone is hung up, but there's no dtmf
picked up by asterisk either...

Anyone with ideas?



Thanks,
-- 
Jeremy Jones [EMAIL PROTECTED]


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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]

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[Asterisk-Users] Ringing multiline phone

[Joseph]

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?

-- 
#Joseph
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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
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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
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[Asterisk-Users] Ringing tone on calls going out on chan_modem

[Paolo Losi]

Hi all,
I set up a IS64PHHiSax: IS64PH Hisax compatible card to terminate sip
calls to PSTN. Everything works as expected but I can get no ringing tone
when the call attempt is in progress. As soon as the called party
answers everything works again. I've tried all three options for mode
parameter in modem.conf without any result.
Modem.conf is:
[interfaces]
context=remote
driver=i4l  ; isdn4linux - an alternative to i4l is to use chan_capi
language=en
type=autodetect
dialtype=tone
mode=ring
device = /dev/ttyI0
device = /dev/ttyI1
Do you have any suggestion?
Grazie!
Paolo
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[Asterisk-Users] Ringing() doesn't play sound while phone is ringing

[Warren Burstein]

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 dont 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 dont 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' 



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



show
application ringing doesnt describe any parameters to Ringing() 



Thanks.

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.
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