[asterisk-users] Call waiting for Queue Agents.

[Aziz TestAccount]

Hi All,

I have a question about the Queues.

I'm using Asterisk 11.13.0 , and I want to configure the following setup :

When there is an incoming call to the queue all agents should ring even
those that are already in call, they should receive a second call.

Is this doable in any Asterisk version ?

Thanks in advance.
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Re: [asterisk-users] Call waiting for Queue Agents.

[Aziz TestAccount]

Hi,

Thanks for your reply.

It's working.  I forgot to enable call waiting under  extensions in
Asterisk.


Best regards


On Mon, Sep 21, 2015 at 3:36 PM, Ishfaq Malik  wrote:

>
>
> On 21 September 2015 at 15:27, Aziz TestAccount 
> wrote:
>
>> Hi All,
>>
>> I have a question about the Queues.
>>
>> I'm using Asterisk 11.13.0 , and I want to configure the following setup
>> :
>>
>> When there is an incoming call to the queue all agents should ring even
>> those that are already in call, they should receive a second call.
>>
>> Is this doable in any Asterisk version ?
>>
>> Thanks in advance.
>>
>>
>>
> In 1.8 there is a ring in use option at the queue level. I doubt this will
> have been removed in 11.
>
> ; If you want the queue to avoid sending calls to members whose devices are
> ; known to be 'in use' (via the channel driver supporting that device
> state)
> ; uncomment this option. (Note: only the SIP channel driver currently is
> able
> ; to report 'in use'.)
> ;
> ; ringinuse = no
>
>
> Regards
>
> Ish
>
>
> --
>
> Ishfaq Malik
> Department: VOIP Support
> Company: Packnet Limited
> t: +44 (0)161 660 2350
> 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] Call waiting for Queue Agents.

[Ishfaq Malik]

On 21 September 2015 at 15:27, Aziz TestAccount 
wrote:

> Hi All,
>
> I have a question about the Queues.
>
> I'm using Asterisk 11.13.0 , and I want to configure the following setup :
>
> When there is an incoming call to the queue all agents should ring even
> those that are already in call, they should receive a second call.
>
> Is this doable in any Asterisk version ?
>
> Thanks in advance.
>
>
>
In 1.8 there is a ring in use option at the queue level. I doubt this will
have been removed in 11.

; If you want the queue to avoid sending calls to members whose devices are
; known to be 'in use' (via the channel driver supporting that device state)
; uncomment this option. (Note: only the SIP channel driver currently is
able
; to report 'in use'.)
;
; ringinuse = no


Regards

Ish


-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
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] call-waiting

[C F]

Interface type?

On Fri, May 28, 2010 at 1:47 AM, bhrugu mehta mehtabhr...@gmail.com wrote:
 hi, all

 Is ther any way to set up call-waiting feature in asterisk using dialplan or
 any other ways. I want to use only
 asterisk for that not any other gui.

 I am using asterisk 1.4.28.

 Regards,

 --
 Bhrugu Mehta
 Sr. S/W Engineer (DD)
 VOIP,Telephony Team
 India

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[asterisk-users] call-waiting

[bhrugu mehta]

hi, all

Is ther any way to set up call-waiting feature in asterisk using dialplan or
any other ways. I want to use only
asterisk for that not any other gui.

I am using asterisk 1.4.28.

Regards,

-- 
Bhrugu Mehta
Sr. S/W Engineer (DD)
VOIP,Telephony Team
India
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[asterisk-users] Call-Waiting, implementation ideas

[Harel Cohen]

Hi all,
How can I implement a full-featured Call-Waiting behavior on the Asterisk level 
(e.g. I don't want to relay on end-equipment capabilities)?
I found it very strange that such a basic feature is not built-in in Asterisk 
(and I've googled a lot in search for this).

Here is what I need:
SomeuserX is calling MyUserA. They are on conversation (assumption: voice is 
via the Asterisk)
SomeuserY is calling MyUserA.
SomeuserY gets a special ringing tone. Meaning - Asterisk opens voice channel 
towards SomeuserY (progress with SDP) and plays SpecialRingBack.wav/gsm etc.
MyUserA Gets voice notification (e.g. beep-beep) during his call to SomeuserX. 
Meaning - Asterisk barge-in the rtp stream and play the file beepbeep.wav/gsm 
on the MyUserA channel. This is done periodically for as long as SomeuserY is 
waiting to be answered (i.e. doesn't hang-up).
Asterisk is monitoring the state of the call SomeuserX - MyUserA.
If MyUserA will signal (e.g. hook-flash or some digit sequence) that he wants 
to answer the 2nd call then Asterisk will put on hold SomeuserX and bridge 
SomeuserY to MyUserA with the option for MyUserA to toggle between the two 
channels.
If the conversation SomeuserX with MyUserA is terminated Asterisk will INVITE 
MyUserA and when picked up will bridge SomeuserY with MyUserA.
I hope there is a solution for that…
I tried using DEVICE_STATE for this purpose however I keep getting status 
NOT_INUSE even if the extension IS in use (I'll open a different thread on this 
issue if needed).
Thanks in advance for any ideas provided,
Harel

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Re: [asterisk-users] Call-Waiting, implementation ideas

[C F]

If you use zap then asterisk already does it. With sip the phones will
not tell asterisk about the hook flash. However you can play around
with dynamic features and assign a key that will mimic hook flash.
Injecting the beep sound might be hard though. Playing a different
ring to 2nd caller based on if the recipient is on the phone can be
accomplished using chanavail or whatever that app is called can't
recall at the moment and I'm typing this on my BB

On 4/30/10, Harel Cohen ha...@easycall.gi wrote:
 Hi all,
 How can I implement a full-featured Call-Waiting behavior on the Asterisk
 level (e.g. I don't want to relay on end-equipment capabilities)?
 I found it very strange that such a basic feature is not built-in in
 Asterisk (and I've googled a lot in search for this).

 Here is what I need:
 SomeuserX is calling MyUserA. They are on conversation (assumption: voice is
 via the Asterisk)
 SomeuserY is calling MyUserA.
 SomeuserY gets a special ringing tone. Meaning - Asterisk opens voice
 channel towards SomeuserY (progress with SDP) and plays
 SpecialRingBack.wav/gsm etc.
 MyUserA Gets voice notification (e.g. beep-beep) during his call to
 SomeuserX. Meaning - Asterisk barge-in the rtp stream and play the file
 beepbeep.wav/gsm on the MyUserA channel. This is done periodically for as
 long as SomeuserY is waiting to be answered (i.e. doesn't hang-up).
 Asterisk is monitoring the state of the call SomeuserX - MyUserA.
 If MyUserA will signal (e.g. hook-flash or some digit sequence) that he
 wants to answer the 2nd call then Asterisk will put on hold SomeuserX and
 bridge SomeuserY to MyUserA with the option for MyUserA to toggle between
 the two channels.
 If the conversation SomeuserX with MyUserA is terminated Asterisk will
 INVITE MyUserA and when picked up will bridge SomeuserY with MyUserA.
 I hope there is a solution for that…
 I tried using DEVICE_STATE for this purpose however I keep getting status
 NOT_INUSE even if the extension IS in use (I'll open a different thread on
 this issue if needed).
 Thanks in advance for any ideas provided,
 Harel



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[asterisk-users] Call Waiting With Draytek ATA

[Tim Nelson]

Greetings all-

I've got a rather odd situation and would like to know if anyone can shed some 
light on the issue.

Some background- I've got an * system running 1.4.11 (yes I know it's older.. 
upgrades are planned at some point...). I also have a remote user with a 
cordless phone connected to a Draytek ATA device.

When this user is on a call and receives another call via call waiting, they 
use the 'flash' button on their phone to switch to the other call. When this 
occurs, music on hold is started for the first call, and the second call is 
connected. However, at this point music on hold suddenly stops and audio from 
both calls can be heard together (and is rather garbled). Then, hitting flash 
again, call 2 is disconnected and call 1 is connected again. BUT, only one way 
audio(inbound to the user) is available on the first call now.

I thought it could be a problem with MoH and ensured that was setup properly. 
Still the same problem. Then, I thought it could be a problem with the version 
of Asterisk I was running. As it turns out, a separate system running 1.2.13 
works perfectly.

So, at this point, I have to ask... are there any known issues like this that 
have been fixed in later versions than what I'm running? I know I'll probably 
receive a general blanket statement like upgrade to the latest but what I'm 
looking for is solid proof that an upgrade will fix it (something from the bug 
tracker maybe?). Or, maybe I'm going about this the wrong way and its something 
configured wrong elsewhere and * is not at fault?

All thought and comments welcome. Thank you!

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

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[asterisk-users] Call Waiting

[John covici]

Are you doing *0 after you flash the hook?  This will flash the fxo
line for you.  I do wish there was a way to get Asterisk to answer the
call waiting on the fxo, all I ever get is the call waiting beep and I
get to answer it myself, otherwise it goes to telco's voicemail.

on Wednesday 07/11/2007 Joe acquisto([EMAIL PROTECTED]) wrote
  Since the beginning (of my Asterisk life) I have an install that is, 
  supposedly, set up for call waiting.
  
  Using a TDM400p, with FXO and FXS modules.
  
  On the Analog phones, I can hear the Incoming call (call waiting) tone, but 
  the system does not respond to a hook flash, to place the current call on 
  hold and answer the incoming call.   I have not attempted, nor research 
  how/if this can be done on SIP.
  
  What am I not grasping here? About the Analog phone/Asterisk actions.
  
  Not too vague, I hope.
  
  joe a.
  
  
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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
 [EMAIL PROTECTED]

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Re: [asterisk-users] Call Waiting

[Joe acquisto]

 On 7/11/2007 at 11:04 AM, Joe acquisto [EMAIL PROTECTED] wrote:
 Since the beginning (of my Asterisk life) I have an install that is, 
 supposedly, set up for call waiting.
 
 Using a TDM400p, with FXO and FXS modules.
 
 On the Analog phones, I can hear the Incoming call (call waiting) tone, but 
 the system does not respond to a hook flash, to place the current call on 
 hold and answer the incoming call.   I have not attempted, nor research 
 how/if this can be done on SIP.
 
 What am I not grasping here? About the Analog phone/Asterisk actions.
 
 Not too vague, I hope.
 
 joe a.
 

OK, so the secret seems to be to flash (press hook button briefly) as normal, 
the do *0.  That takes me to the waiting call.   But how to switch back, is 
still a mystery.Do to various constraints intense testing is not possible 
at this time.

Anyone?

joe a.


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Re: [asterisk-users] Call Waiting

[Mojo with Horan Company, LLC]

try to transfer current call to parking spot, i.e. exten 700, then deal 
with new incoming call, then go back to parking space to pick up old 
caller when you're free.  Just set the parking extension timeout to 
something long so they don't fall out right away.

Moj

Joe acquisto wrote:
 On 7/11/2007 at 11:04 AM, Joe acquisto [EMAIL PROTECTED] wrote:
 Since the beginning (of my Asterisk life) I have an install that is, 
 supposedly, set up for call waiting.

 Using a TDM400p, with FXO and FXS modules.

 On the Analog phones, I can hear the Incoming call (call waiting) tone, but 
 the system does not respond to a hook flash, to place the current call on 
 hold and answer the incoming call.   I have not attempted, nor research 
 how/if this can be done on SIP.

 What am I not grasping here? About the Analog phone/Asterisk actions.

 Not too vague, I hope.

 joe a.

 
 OK, so the secret seems to be to flash (press hook button briefly) as normal, 
 the do *0.  That takes me to the waiting call.   But how to switch back, is 
 still a mystery.Do to various constraints intense testing is not possible 
 at this time.
 
 Anyone?
 
 joe a.
 
 
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[asterisk-users] Call Waiting

[Joe acquisto]

Since the beginning (of my Asterisk life) I have an install that is, 
supposedly, set up for call waiting.

Using a TDM400p, with FXO and FXS modules.

On the Analog phones, I can hear the Incoming call (call waiting) tone, but the 
system does not respond to a hook flash, to place the current call on hold 
and answer the incoming call.   I have not attempted, nor research how/if this 
can be done on SIP.

What am I not grasping here? About the Analog phone/Asterisk actions.

Not too vague, I hope.

joe a.


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Re: [asterisk-users] Call Waiting curiosity...

[Mojo with Horan Company, LLC]

Is your incoming context using chanisavail, while your internal-dialing 
context is not, and just sends the call, without checking?

Mojo

Michael Wareman wrote:
 Hi,
 
 I have (to me) an interesting problem.
 
 There are 3 physical extensions, 11, 12 and 13. All hang off Sipura 
 adapters.
 There is also extension 10 which simply uses 
 'Dial(SIP/11SIP/12SIP/13)' to call all phones in the house.
 
 Incoming calls from outside get sent to 10 in order that they can be 
 answered from any phone..
 
 Now - if (say) 11 is on a call externally, and 12 calls 11 - 11 get's 
 the call waiting beeps, and can 'flash' over to the new incoming call. 
 No problem there.
 
 However, if 12 instead calls 10, in the log I see the Dial command sees 
 11 as 'In Use' and the call never causes the call waiting beep in 11.
 
 Any way to change this? 
 
 Many thanks,
 
 Michael.
 
 
 
 
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[asterisk-users] Call Waiting curiosity...

[Michael Wareman]

Hi,

I have (to me) an interesting problem.

There are 3 physical extensions, 11, 12 and 13. All hang off Sipura
adapters.
There is also extension 10 which simply uses 'Dial(SIP/11SIP/12SIP/13)' to
call all phones in the house.

Incoming calls from outside get sent to 10 in order that they can be
answered from any phone..

Now - if (say) 11 is on a call externally, and 12 calls 11 - 11 get's the
call waiting beeps, and can 'flash' over to the new incoming call. No
problem there.

However, if 12 instead calls 10, in the log I see the Dial command sees 11
as 'In Use' and the call never causes the call waiting beep in 11.

Any way to change this?

Many thanks,

Michael.
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[asterisk-users] Call waiting / hook flash on ZAP trunk from SIP phone?

[Sean M. Pappalardo]

Hello.

After doing much web searching and searching archives of this mailing 
list, I see that my question has been asked at least 6 separate times 
but no answers have been attached.


In a nutshell, is there a way for a SIP phone to easily hook flash a ZAP 
analog trunk mid-call? (This is important when trying to make use of 
features on a PSTN analog line such as call waiting, call forwarding, 
3-way calling, etc.)


I've seen the *3 trick (mentioned in the comments here 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Flash ), but I 
don't know in which context to put the extensions.conf stuff:


exten = s,n,Set(DYNAMIC_FEATURES=zapflash)
exten = s,n,Dial(SIP/,15,tw)

...to make it take effect while on a call.

My configuration:

- 1 X100P card with one analog line attached with call waiting, caller 
ID, 3-way calling enabled on the line.

- 1 Snom 300 SIP desk phone

My Zapata.conf:

[trunkgroups]
; define any trunk groups

[channels]
; hardware channels
; default

usecallingpres=yes
usecallerid=yes
cidstart=ring
cidsignalling=bell
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=no
callreturn=no
immediate=no   ; for Caller ID, allows time for telco to send digits
hidecallerid=no
echocancel=yes
echotraining=yes
useincomingcalleridonzaptransfer=yes
context=from-pstn   ; Incoming calls go to [from-pstn]
signalling=fxs_ks   ; Use FXS signalling for an FXO channel
group=0 ; Use with Zap/g0
channel = 1; PSTN attached to port 1


Thank you very much for your time. Hopefully we can get an answer to 
this and put it on the Wiki for all to see!


Sincerely,
Sean M. Pappalardo

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Re: [asterisk-users] Call waiting tone when calling a busy station?

[Edoardo Serra]

Yehavi Bourvine +972-8-9489444 ha scritto:


This is not what I meant. I want the called party to get a sign of a waiting
call and answer it if he/she wants.

Ok, that's an UAC option

 I want the caller to know that he on a
waiting call (here it is customary to play a stuttered ring tone).
in short - can I signal in the 183 ringing packet that this is a second call?
  

I don't think SIP has an implementation of that

My suggestion is to use a queue in which you would put callers if the 
called party is busy

(you can check that with ome AGI scripting)
You can then record a stuttered 'ring' tone and put that as background 
music for the queue.


Queues are the best way to handle you situation even if it's not an 
elegant solution for playing the stuttered ring tone


My 2 cents

  Thanks! __Yehavi:
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--
Ing. Edoardo Serra
WeBRainstorm S.r.l.
Via Pio Foà 83/C
10126 - Torino

Tel: +39 011 678 100
Fax: +39 011 678 275

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[asterisk-users] Call waiting tone when calling a busy station?

[Yehavi Bourvine +972-8-9489444]

Hello,

  When dialling a SIP phone which is already in a call the caller hears a
regular ringing tone and does not know that the called party is engaged in
another call. Is there a supported way inside SIP to tell the calling party to
play a stuttered ringing tone?

   Thanks! __Yehavi:
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Re: [asterisk-users] Call waiting tone when calling a busy station?

[Edoardo Serra]

Hello,
   this is a SIP phone configuration issue.

You should tell the UAC to not accept a second call while the line is 
engaged (look for a 'Call Waiting' option in the configuration of the UAC)
The UAC will send back a 486 Busy Here error code and the calling 
party will get a busy signal  from asterisk


The calling party will then play a busy tone, or Asterisk will emulate 
it in case of analog zaptel devices


Regards

Edoardo Serra
WeBRainstorm S.r.l.

Yehavi Bourvine +972-8-9489444 ha scritto:

Hello,

  When dialling a SIP phone which is already in a call the caller hears a
regular ringing tone and does not know that the called party is engaged in
another call. Is there a supported way inside SIP to tell the calling party to
play a stuttered ringing tone?

   Thanks! __Yehavi:
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--
Ing. Edoardo Serra
WeBRainstorm S.r.l.
Via Pio Foà 83/C
10126 - Torino

Tel: +39 011 678 100
Fax: +39 011 678 275

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Re: [asterisk-users] Call waiting tone when calling a busy station?

[Yehavi Bourvine +972-8-9489444]

 this is a SIP phone configuration issue.

 You should tell the UAC to not accept a second call while the line is
 engaged (look for a 'Call Waiting' option in the configuration of the UAC)
 The UAC will send back a 486 Busy Here error code and the calling
 party will get a busy signal  from asterisk

This is not what I meant. I want the called party to get a sign of a waiting
call and answer it if he/she wants. I want the caller to know that he on a
waiting call (here it is customary to play a stuttered ring tone).

in short - can I signal in the 183 ringing packet that this is a second call?

  Thanks! __Yehavi:
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Re: [asterisk-users] Call Waiting problems

[Lachek Butalek]

On 3/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote:

If you are using sip then you should look for the call-limit option in
sip.conf file.


Using IAX. Is that a problem?
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Re: [asterisk-users] Call Waiting problems

[Rizwan Hisham]

If you are using sip then you should look for the call-limit option in
sip.conf file.

On 3/30/07, Lachek Butalek [EMAIL PROTECTED] wrote:


Situation, simple home setup:

* Trixbox 2.0
* Feature Codes installed
* GNet PA-168V based ATA
* Cheesy cordless analogue phone

From what I gather, dialing *70 from the handset should activate Call
Waiting. All it seems to do is change the message The person at
extension is on the phone to ring ring The person at
extension is unavailable. The person speaking on the phone at the
time of the second incoming call hears no indication that another call
is incoming.

Part of the problem is that I have no idea how the feature should work
when it's functional. Could someone help me troubleshoot this, or
point me in the right direction? It seems as though, as a very basic
feature, not a lot of documentation is written about it.

Thanks!
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--
Regards
Rizwan Hisham
Software Engineer
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[asterisk-users] Call Waiting problems

[Lachek Butalek]

Situation, simple home setup:

* Trixbox 2.0
* Feature Codes installed
* GNet PA-168V based ATA
* Cheesy cordless analogue phone


From what I gather, dialing *70 from the handset should activate Call

Waiting. All it seems to do is change the message The person at
extension is on the phone to ring ring The person at
extension is unavailable. The person speaking on the phone at the
time of the second incoming call hears no indication that another call
is incoming.

Part of the problem is that I have no idea how the feature should work
when it's functional. Could someone help me troubleshoot this, or
point me in the right direction? It seems as though, as a very basic
feature, not a lot of documentation is written about it.

Thanks!
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[asterisk-users] Call Waiting broken on ZAP

[John Hyde]

Problem: *Call* *waiting* comes in, I press flash to answer it, and the
first caller gets disconnected after 3 seconds. This is all ZAP - no VOIP.

System:
Analog stations and trunks running on a pair of TDM400's. It does NOT have *
call* *waiting* from the phone company, and I have enabled it in all my conf
files. The trunks are set to FXSKS and the stations are FXOKS. I am not
using *call* progress or busy detect. Also its * 1.2.13 w/ FreePBX2.2.  I
have scoured the net for this, and nobody seems to know.

Here is some logging from a *call*:

Feb 1 17:41:53 DEBUG[6765] chan_zap.c: Requested indication 3 on channel
Zap/5-1
Feb 1 17:41:53 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:41:53 DEBUG[6765] pbx.c: Function result is 's'
Feb 1 17:41:53 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:41:53 DEBUG[6765] pbx.c: Function result is '5'
Feb 1 17:41:53 DEBUG[6765] db.c: Unable to find key '5187152626' in family
'blacklist'
Feb 1 17:41:53 DEBUG[6765] pbx.c: Expression result is '0'
Feb 1 17:41:53 DEBUG[6765] pbx.c: Not taking any branch
Feb 1 17:41:53 DEBUG[6765] chan_zap.c: Took Zap/5-1 off hook
Feb 1 17:41:53 DEBUG[6765] chan_zap.c: Enabled echo cancellation on channel
5
Feb 1 17:41:53 DEBUG[6765] chan_zap.c: Engaged echo training on channel 5
Feb 1 17:41:54 DEBUG[6765] channel.c: Scheduling timer at 160 sample
intervals
Feb 1 17:42:00 DEBUG[6765] chan_zap.c: DTMF digit: 5 on Zap/5-1
Feb 1 17:42:00 DEBUG[6765] channel.c: Scheduling timer at 0 sample intervals

Feb 1 17:42:00 DEBUG[6765] pbx.c: Oooh, got something to jump out with
('5')!
Feb 1 17:42:01 DEBUG[6765] chan_zap.c: DTMF digit: 0 on Zap/5-1
Feb 1 17:42:02 DEBUG[6765] chan_zap.c: DTMF digit: 0 on Zap/5-1
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is ''
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is '5187152626'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '0'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Not taking any branch
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '0'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Not taking any branch
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is '5187152626'
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '5187152626/user' in
family 'DEVICE'
Feb 1 17:42:02 DEBUG[6765] func_db.c: DB: DEVICE/5187152626/user not found
in database.
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is ''
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '/cidname' in family
'AMPUSER'
Feb 1 17:42:02 DEBUG[6765] func_db.c: DB: AMPUSER//cidname not found in
database.
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is ''
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '0'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Not taking any branch
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '-1'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is '64'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is ' 5187152626'
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '500' in family 'CFU'
Feb 1 17:42:02 DEBUG[6765] func_db.c: DB: CFU/500 not found in database.
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is ''
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '0'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '1'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is '15'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Function result is '0'
Feb 1 17:42:02 DEBUG[6765] pbx.c: Expression result is '0'
Feb 1 17:42:02 VERBOSE[6765] logger.c:
recordingcheck|20070201-174202|1170380512.151: Inbound recording enabled.
Feb 1 17:42:02 VERBOSE[6765] logger.c:
recordingcheck|20070201-174202|1170380512.151: CALLFILENAME=
20070201-174202-1170380512.151
Feb 1 17:42:02 DEBUG[6765] channel.c: Spy MixMonitor added to channel
Zap/5-1
Feb 1 17:42:02 VERBOSE[6765] logger.c: dialparties.agi: Starting New
Dialparties.agi
Feb 1 17:42:02 VERBOSE[6765] logger.c: dialparties.agi: priority is 1
Feb 1 17:42:02 VERBOSE[6765] logger.c: dialparties.agi: Caller ID name is
'unknown' number is '5187152626'
Feb 1 17:42:02 VERBOSE[6765] logger.c: dialparties.agi: Methodology of ring
is 'none'
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '500' in family 'CF'
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '500' in family 'DND'
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '500' in family 'CFB'
Feb 1 17:42:02 DEBUG[6765] db.c: Unable to find key '500' in family 'CFU'
Feb 1 17:42:02 DEBUG[6765] chan_zap.c: Requested indication 3 on channel
Zap/5-1
Feb 1 17:42:02 DEBUG[6765] channel.c: Building translator from ulaw to
SLINEAR for spies on channel Zap/5-1
Feb 1 17:42:03 DEBUG[6765] chan_zap.c: Exception on 13, channel 2
Feb 1 17:42:03 DEBUG[6765] chan_zap.c: Got event Ring/Answered(2) on channel
2 (index 0)
Feb 1 17:42:03 DEBUG[6765] chan_zap.c: Enabled echo cancellation on channel
2

Re: [asterisk-users] Call waiting notification

[Kevin Smith]

Hi Kevin,

Thanks, that's what I thought but sometimes you need a second opinion 
from someone with more experience to get administration off your back 
about an issue such as this.


Kevin



Kevin P. Fleming wrote:

Kevin Smith wrote:
  

We are running Polycom 601's. I can't seem to find anything to say one
way or another on this issue, so I figured I would ask. I have call
waiting notification working on the phones when a user is on the phone.
However, is it possible to see the notification on the screen or hear it
on the line when it is in the dial status, IE I just pick the receiver
off the hook and I am about to dial a number.



Nope.
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[asterisk-users] Call waiting notification

[Kevin Smith]

Hi everyone,

We are running Polycom 601's. I can't seem to find anything to say one 
way or another on this issue, so I figured I would ask. I have call 
waiting notification working on the phones when a user is on the phone. 
However, is it possible to see the notification on the screen or hear it 
on the line when it is in the dial status, IE I just pick the receiver 
off the hook and I am about to dial a number.


Kevin
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Re: [asterisk-users] Call waiting notification

[Kevin P. Fleming]

Kevin Smith wrote:
 We are running Polycom 601's. I can't seem to find anything to say one
 way or another on this issue, so I figured I would ask. I have call
 waiting notification working on the phones when a user is on the phone.
 However, is it possible to see the notification on the screen or hear it
 on the line when it is in the dial status, IE I just pick the receiver
 off the hook and I am about to dial a number.

Nope.
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Re: [asterisk-users] call waiting

[Tzafrir Cohen]

On Tue, Sep 12, 2006 at 09:55:17PM -0700, Christopher Corn wrote:

 Christopher Corn [EMAIL PROTECTED] wrote:

  i've got trixbox installed and grandstream 101 phones.
 
out of my 4 phones, one of them has call waiting working. 
  they all the same version of firmware and settings. i tried 
  looking in asterisk to see if anything could be doing this, 
  but can't find anything. suggestions? Thanks.

 nevermind i figured it out :)

What was it, then?

The archives of the list would like you to tell them ;-)

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] call waiting

[Christopher Corn]

well this is my guess :) i had call features (call waiting) enabled on my budgetone 100. i guess when i would dial *70 to enable call waiting, it wasn't reaching my asterisk server. I then turned off call features on my phone then when I would dial *70, I could hear a voice response telling me call waiting was then enabled. it seems that one step in configuring is to enable call waiting, as it appears to be disabled by default.Tzafrir Cohen [EMAIL PROTECTED] wrote:  On Tue, Sep 12, 2006 at 09:55:17PM -0700, Christopher Corn wrote: Christopher Corn <[EMAIL PROTECTED]>wrote:   i've got trixbox installed and grandstream 101 phones.   out of my 4 phones, one of them has call waiting working.   they all the same version
 of firmware and settings. i tried   looking in asterisk to see if anything could be doing this,   but can't find anything. suggestions? Thanks. nevermind i figured it out :)What was it, then?The archives of the list would like you to tell them ;-)-- Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755 iax:[EMAIL PROTECTED]+972-50-7952406 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] http://www.xorcom.com___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___
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[asterisk-users] call waiting

[Christopher Corn]

i've got trixbox installed and grandstream 101 phones.  out of my 4 phones, one of them has call waiting working. they all the same version of firmware and settings. i tried looking in asterisk to see if anything could be doing this, but can't find anything. suggestions? Thanks.___
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Re: [asterisk-users] call waiting

[Christopher Corn]

nevermind i figured it out :)Christopher Corn [EMAIL PROTECTED] wrote:i've got trixbox installed and grandstream 101 phones.  out of my 4 phones, one of them has call waiting working. they all the same version of firmware and settings. i tried looking in asterisk to see if anything could be doing this, but can't find anything. suggestions? Thanks.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___
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[Asterisk-Users] Call waiting using free PBX

[Dumpolid Exeplish]

hi list,
i have tried to set the call waiting function using freePBX but it dosent work. i think there is something wrong with the coding. Has anyone experienced this sort of problems?
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Re: [Asterisk-Users] Call waiting using free PBX

[Raymond McKay]

hi list,
i have tried to set the call waiting function using freePBX but it dosent 
work. i think there is something wrong with the coding. Has anyone

 experienced this sort of problems?


Can you expand a bit more on your problem?  What versions of software are 
you running?  What have you tried so far?


Regards,


Raymond McKay
President
RAYNET Technologies LLC
http://www.raynettech.com
(860) 693-2226 x 31
Toll Free (877) 693-2226


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[Asterisk-Users] call waiting announcement on agent phone

[Pavel Jezek]

Hello, is there any way to anounce on agents phone (e.g. beep tone like 
in gsm), that some call is waiting in queue?

PJ
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[Asterisk-Users] Call Waiting Issues

[Brad Glonka]

I have two call waiting problems.

I have a POTS line into and FXO port
and telephones on an FXS port

1) I can't seem to use the flash button(on the phone) to answer a call
waiting call.
 I see the callerid coming though and here the call waiting tone,
but I just can't seem to answer it.  The flash button seems to have no
effect.

 I have:
callwaiting=yes in zapata.conf


2) When the PSTN line is in use and a call comes though via call waiting.
   I don't think it hits my   exten = s
   Instead it rings the phone (but as I mentioned above I
can't seem to answer it)

Thanks for any suggestions.
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Re: [Asterisk-Users] Call Waiting Issues

[C F]

You have to flash the FXO not your phone, use features.conf to accomplish that.

On 3/27/06, Brad Glonka [EMAIL PROTECTED] wrote:
 I have two call waiting problems.

 I have a POTS line into and FXO port
 and telephones on an FXS port

 1) I can't seem to use the flash button(on the phone) to answer a call
 waiting call.
  I see the callerid coming though and here the call waiting tone,
 but I just can't seem to answer it.  The flash button seems to have no
 effect.

  I have:
 callwaiting=yes in zapata.conf


 2) When the PSTN line is in use and a call comes though via call waiting.
I don't think it hits my   exten = s
Instead it rings the phone (but as I mentioned above I
 can't seem to answer it)

 Thanks for any suggestions.
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Re: [Asterisk-Users] Call Waiting Issues

[Mojo with Horan Company, LLC]

Brad Glonka wrote:

I have two call waiting problems.

I have a POTS line into and FXO port
and telephones on an FXS port

1) I can't seem to use the flash button(on the phone) to answer a call
waiting call.
 I see the callerid coming though and here the call waiting tone,
but I just can't seem to answer it.  The flash button seems to have no
effect.

 I have:
callwaiting=yes in zapata.conf


2) When the PSTN line is in use and a call comes though via call waiting.
   I don't think it hits my   exten = s
   Instead it rings the phone (but as I mentioned above I
can't seem to answer it)
When a call comes in through call waiting and you hear the tone, it 
can't hit your exten = s.  I think *0 or whatever is configured in 
features.conf for 'disconnect' might send the flash down the pstn line.




Thanks for any suggestions.
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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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[Asterisk-Users] Call Waiting? Should this just work?

[Martin Joseph]

I realized today that my call waiting isn't working properly.  If I am 
using the FXS attached phone and a call comes in the FXO,  it just goes 
directly to voicemail, with no indication (call waiting beep).


If I flash there is a second dial tone, and I can initiate a second 
call.


If I am dialed out through the FXO and another call comes in I hear the 
call waiting beep, when I flash over I hear another dialtone rather 
then the incoming call.


With the HT-488 this feature seemed to work.  With my new Wellgate 
3701a I don't know what to do to make it go?


Thoughts, ideas, suggestions welcomed.
Marty

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RE: [Asterisk-Users] Call Waiting x100P and Cisco IP Phone

[Chuck Smith]

OK with that being said how can you modify the phone to use the second line
button as a speed dial? Then you can label it has flash.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Thursday, February 02, 2006 11:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Call Waiting x100P and Cisco IP Phone

 How can I send the hook flash to the x100P card to switch to the call
 coming in from the PSTN?

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

Scroll down to Re: X100P + Call-Waiting how-to

Enjoy.

Nabeel

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[Asterisk-Users] Call Waiting x100P and Cisco IP Phone

[Chuck Smith]

OK I have looked everywhere and I can't get a clear understanding on how to
do this. If I have an x100P card connected to my home phone line and I am
receiving calls on my Cisco 7940 IP phone with a SIP firmware loaded on it.
How can I send the hook flash to the x100P card to switch to the call coming
in from the PSTN? I am using [EMAIL PROTECTED] 2.4. I can hear the call waiting
tone coming over the line but the phone doesn't recognize it.



Thanks



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RE: [Asterisk-Users] Call Waiting x100P and Cisco IP Phone

[kevin ling]

Hi,

In AAH, you can setup the Incoming Calls to ring your extension. Or to
ring extensions in a ring group. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Smith
Sent: Friday, February 03, 2006 6:53 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Call Waiting x100P and Cisco IP Phone

OK I have looked everywhere and I can't get a clear understanding on how to
do this. If I have an x100P card connected to my home phone line and I am
receiving calls on my Cisco 7940 IP phone with a SIP firmware loaded on it.
How can I send the hook flash to the x100P card to switch to the call coming
in from the PSTN? I am using [EMAIL PROTECTED] 2.4. I can hear the call waiting
tone coming over the line but the phone doesn't recognize it.



Thanks



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RE: [Asterisk-Users] Call Waiting x100P and Cisco IP Phone

[Nabeel Jafferali]

 How can I send the hook flash to the x100P card to switch to the call
 coming in from the PSTN?

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

Scroll down to Re: X100P + Call-Waiting how-to

Enjoy.

Nabeel

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[Asterisk-Users] Call Waiting CallerID

[Andy Kuo]

Hi,

According to the wiki, we need to have both callwaiting=yes and
callwaitingcallerid=yes , and that's what I have in zapata.conf.

I can hear the call waiting alert tone when a 2nd call comes in during
an established call, and I can switch between the calls without
problems.  However, CallerID on the 2nd call does not show up with the
call waithing alert tones.

Am I missing something?  Can anyone help?
Thank you in advance.
Andy
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[Asterisk-Users] Call Waiting CallerID not showing up

[Andy Kuo]

Hi All,

According to the wiki, we need to have both callwaiting=yes and
callwaitingcallerid=yes , and that's what I have in zapata.conf.

I can hear the call waiting alert tone when a 2nd call comes in during
an established call, and I can switch between the calls without
problems.  However, CallerID on the 2nd call does not show up with the
call waithing alert tones.

Can anyone help?
Thank you in advance.
Andy
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RE: [Asterisk-Users] Call waiting issue

[Steve Totaro]

A simple sql command will do this.  

 -Original Message-
 From: Kerry Garrison [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, November 22, 2005 1:10 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] Call waiting issue
 
 Whenever I restart Asterisk, I then have to go to each phone and dial
*70
 to turn call waiting back on so that the multiple lines on the phones
will
 ring through instead of getting a busy when the user is only on a
single
 call. Is there a simple way to have call waiting be On by default?
 
 -Kerry
 
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RE: [Asterisk-Users] Call waiting issue

[Kerry Garrison]

I'm not following, must be too tired. Are you saying that on startup I could
run a SQL command that toggles everyone's call waiting status?
-Kerry 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Tuesday, November 22, 2005 5:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call waiting issue

A simple sql command will do this.  

 -Original Message-
 From: Kerry Garrison [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, November 22, 2005 1:10 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] Call waiting issue
 
 Whenever I restart Asterisk, I then have to go to each phone and dial
*70
 to turn call waiting back on so that the multiple lines on the phones
will
 ring through instead of getting a busy when the user is only on a
single
 call. Is there a simple way to have call waiting be On by default?
 
 -Kerry
 
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[Asterisk-Users] Call waiting issue

[Kerry Garrison]

Whenever I restart Asterisk, I then have to go to each 
phone and dial *70 to turn call waiting back on so that the multiple lines on 
the phones will ring through instead of getting a busy when the user is only on 
a single call. Is there a simple way to have call waiting be On by 
default?

-Kerry

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[Asterisk-Users] call waiting not working on PAP2

[Andy Kuo]

Hi,

I have callwaiting=yes in my zapata.conf, and Call Waiting Serv: Yes in the PAP2s.
However,there's sitllno callwaitingon the PAP2s. Everything else work fine.

Any ideas? Am I missing something somewhere?

Thank you.
AK
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Re: [Asterisk-Users] call waiting not working on PAP2

[Tom Vile]

Have a look at the CW Act Code: CW Per Call Act Code: and remove the
entries in there. I have a sipura so I dont know if they are
using the same terminology buts it the same hardware.On 10/13/05, Andy Kuo [EMAIL PROTECTED] wrote:
Hi,

I have callwaiting=yes in my zapata.conf, and Call Waiting Serv: Yes in the PAP2s.
However,there's sitllno callwaitingon the PAP2s. Everything else work fine.

Any ideas? Am I missing something somewhere?

Thank you.
AK

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http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony
www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856
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Re: [Asterisk-Users] call waiting not working on PAP2

[Andy Kuo]

Hi,

I can't seem to find CW Act Code: and CW Per Call Act Code: in PAP2.
Does anyone know what they are in PAP2?

thanks.
AK
On 10/13/05, Tom Vile [EMAIL PROTECTED] wrote:
Have a look at the CW Act Code: CW Per Call Act Code: and remove the entries in there. I have a sipura so I dont know if they are using the same terminology buts it the same hardware.


On 10/13/05, Andy Kuo 
[EMAIL PROTECTED] wrote: 


Hi,

I have callwaiting=yes in my zapata.conf, and Call Waiting Serv: Yes in the PAP2s.
However,there's sitllno callwaitingon the PAP2s. Everything else work fine.

Any ideas? Am I missing something somewhere?

Thank you.
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony
www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 
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[Asterisk-Users] Call Waiting Tracking?

[Nathan E. Pralle]

Hi all.

Searched the archives but couldn't find anything on this:

I want to track 2nd incoming calls on a single line but don't want to pass the 
Call Waiting pips along to the engaged user.  IE:  I want Asterisk to detect 
that CW is currently being transmitted on the line, and track it, but not 
pass it on.

Is there a way to do this?

TIA,
Nathan



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Re: [Asterisk-Users] Call waiting setup/Confenencing problems in AAH

[Gurminder Arora]

Hi,
   Sorry I can't help you in your questions but actually I have one. 
I m using TDM22B card. I am in india.
I want to know are you able to get callerd ID? 
What cidsignalling you have set for in zaptel.conf.?
On my system when a call comes it checks for caller ID and returns and error. 
#error
ERROR[12656]: callerid.c:260 callerid_feed: fsk_serie made mylen  0
(-24)
#
I am not able to understand what it is?
Do tell me how yours is working. 
I am working in Indian Institute of science, Bangalore

Gurminder

On 8/30/05, Raj, Ashok [EMAIL PROTECTED] wrote:
 Hello
 
 I have couple issues with AAH. 1.5
 
 1. Flash panel doesn't show proper status. Sometime accessing with IP
 seems to work and it shows current line status etc. Sometimes accessing
 with hostname of the asterisk server seems to show lines, but it doesn't
 show off hook etc when we pickup a extension and talk.
 
 In /var/www/html/panel/op_server.cfg I have tried setting manager_host
 to all possible values.
 
 127.0.0.1 and its own ip address or its hostname. I have tried to reload
 with asterisk -rx reload, and also a system reboot, none help to get FOP
 working properly.
 
 2. Call waiting. - Does the default configuration disable call waiting?
 I remembered with the same setup when I call myself with X-lite, I used
 to have an incoming call at line3. Now I get forwarded to the busy
 message. Any idea how I can get call-waiting to work?
 
 3. Do we need special hardware to conference? I tried pulling an
 extension to an already in progress call, but it asks for a password.
 Don't know which one of the default passwords would work. Is there a
 default password we need to set?
 
 Cheers,
 ashok raj
 - Open Source Technology Center
 
 
 
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[Asterisk-Users] Call waiting setup/Confenencing problems in AAH

[Raj, Ashok]

Hello

I have couple issues with AAH. 1.5

1. Flash panel doesn't show proper status. Sometime accessing with IP
seems to work and it shows current line status etc. Sometimes accessing
with hostname of the asterisk server seems to show lines, but it doesn't
show off hook etc when we pickup a extension and talk.

In /var/www/html/panel/op_server.cfg I have tried setting manager_host
to all possible values.

127.0.0.1 and its own ip address or its hostname. I have tried to reload
with asterisk -rx reload, and also a system reboot, none help to get FOP
working properly.

2. Call waiting. - Does the default configuration disable call waiting?
I remembered with the same setup when I call myself with X-lite, I used
to have an incoming call at line3. Now I get forwarded to the busy
message. Any idea how I can get call-waiting to work?

3. Do we need special hardware to conference? I tried pulling an
extension to an already in progress call, but it asks for a password.
Don't know which one of the default passwords would work. Is there a
default password we need to set?

Cheers,
ashok raj
- Open Source Technology Center
   


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[Asterisk-Users] call waiting beep on PSTN and TDM400P FXO line hook flash

[Jeff Otterson]

 I have been looking for the answer to this question for a 
while.  Google-ing and reading the archives of Asterisk-Users has not 
enlightened me.


  It seems that this question has been asked many times, and many times it 
has gone unanswered.


 I have call waiting and three way calling on my PSTN line from Verizon 
(the local telco).  This is connected to a FXO port on a TDM400P.  I also 
have two FXS ports on the TDM400P.


  So my problem is, how do I flash the Verizon PSTN line when I hear the 
call waiting beep?  How can I send a hook flash to the Verizon trunk to 
activate their 3-way calling feature.


  I have seen some stuff like hook flash then send *0 to get the bridged 
Zap trunk to flash but I can't get it to work.  I get the reorder 
signal.  I need something my wife and kid can do.


  Can anybody help?

  Thanks,

  Jeff

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[Asterisk-Users] Call waiting beeps

[Chris Johnson]

Any ideas on how to disable the audible incoming call beeps
with *. We have a dial pool and if
someone calls into the pilot number and were talking to them, if another call
comes in, both us and the far end users hear the incoming call beep.






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[Asterisk-Users] Call waiting?

[Adam Collard]

 
Is call waiting supported on an analog incoming line? I have a customer
that has a line with call waiting that wants to go to Asterisk, but
wants to keep the call waiting. If it is, how would I set it up in
asterisk.

Adam Collard
General Manager, ER Wireless
(800) 757-5669 x4861
(810) 496-0161 Fax
(517) 242-1800 Cell
Nextel DC 131*256784*19
[EMAIL PROTECTED]



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RE: [Asterisk-Users] Call waiting?

[Dean Collins]

Yes it is possible, if a call comes in it will beep on the extension
that is speaking to the first call.

Keep in mind individual extensions can also turn off call waiting so you
need to check both pbx and extension settings.

Cheers,
Dean


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Adam Collard
 Sent: Friday, 27 May 2005 9:49 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Call waiting?
 
 
 Is call waiting supported on an analog incoming line? I have a
customer
 that has a line with call waiting that wants to go to Asterisk, but
 wants to keep the call waiting. If it is, how would I set it up in
 asterisk.
 
 Adam Collard
 General Manager, ER Wireless
 (800) 757-5669 x4861
 (810) 496-0161 Fax
 (517) 242-1800 Cell
 Nextel DC 131*256784*19
 [EMAIL PROTECTED]
 
 
 
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Re: [Asterisk-Users] Call waiting?

[Kim Culhan]

On Fri, May 27, 2005 9:48 am, Adam Collard wrote:

 Is call waiting supported on an analog incoming line? I have a customer
 that has a line with call waiting that wants to go to Asterisk,
 but wants to keep the call waiting. If it is, how would I set it up in 
 asterisk.

We have callwaiting=yes
in sections of zapata.conf for both fxo and fxs interfaces.

With a call in progress on the pstn, a second call [waiting]
results in an immediate disconnect of the first pstn call.

On the asterisk end you hear a loud glitch and a burst of
caller ID data, then it disconnects.

This is with the zapata-bsd drivers, maybe with the linux
drivers it would be different.

-kim

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RE: [Asterisk-Users] Call waiting?

[Adam Collard]

I will be installing [EMAIL PROTECTED], if that helps.  


Adam Collard
General Manager, ER Wireless
(800) 757-5669 x4861
(810) 496-0161 Fax
(517) 242-1800 Cell
Nextel DC 131*256784*19
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kim Culhan
Sent: Friday, May 27, 2005 2:29 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Call waiting?

On Fri, May 27, 2005 9:48 am, Adam Collard wrote:

 Is call waiting supported on an analog incoming line? I have a 
 customer that has a line with call waiting that wants to go to 
 Asterisk, but wants to keep the call waiting. If it is, how would I
set it up in asterisk.

We have callwaiting=yes
in sections of zapata.conf for both fxo and fxs interfaces.

With a call in progress on the pstn, a second call [waiting] results in
an immediate disconnect of the first pstn call.

On the asterisk end you hear a loud glitch and a burst of caller ID
data, then it disconnects.

This is with the zapata-bsd drivers, maybe with the linux drivers it
would be different.

-kim

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[Asterisk-Users] Call waiting on TDM-400 FXO

[Kim Culhan]

Is pstn call waiting working on a Digium TDM-400 with FXO ?

Configuration in zapata.conf:

callwaiting=yes
callwaitingcallerid=yes
callprogress=yes
 
If an incoming call happens while the FXO channel has a call in progress,
and the call is routed to a FXS channel (which has callwaiting=yes in
zapata.conf) the call on the FXO is interrupted.

Is anyone else seeing this ?

-kim

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[Asterisk-Users] call waiting signal

[Klaus Marbach]

Hello,

how can i realise a 
call waiting signal in extensions.conf
I use VoIP phone 
snom 360 and the docs say this phones supports this feature.
Is it implemented 
into SIP?

Thanks in 
advance
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RE: [Asterisk-Users] call waiting signal

[Jay Milk]

it should work once it's enabled on your phone.  I'm using SPA-2000s
with Call-Waiting enabled, and I get call-waiting beeps all the time --
no special setup needed.

-Original Message-
From: Klaus Marbach [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, May 17, 2005 4:03 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] call waiting signal


Hello,

how can i realise a call waiting signal in extensions.conf
I use VoIP phone snom 360 and the docs say this phones supports this
feature.
Is it implemented into SIP?

Thanks in advance

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[Asterisk-Users] Call Waiting

[Christopher Kenna]

Is there a way to enable call waiting by default in asterisk? Every timeI create an extension, it is disabled by default. Having to go to every phone is becoming quite annoying. I havent restarted the server yet, butI am afraid of all my extensions changing back to disabled again. Making it the default would just solve all my issues.

Chris

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Re: [Asterisk-Users] Call Waiting

[C F]

Really??
call waiting disalbed? as far as I know there isn't even a function in
asterisk to enable or disable call waiting. There are lots of
workarounds but no function. So you have gone the extra mile to create
such a workaround that (you set it up that way) by default disables
call waiting, so change it.
Do you remember the story with the guy that came to the doctor that
whenever he drinks a coffee hi eye hurts, so the doctor let him make a
coffee and drink it in his office, the doctor then realizes that he
doesn't take out the spoon, why don't you rewrite your dialplan?

On 5/9/05, Christopher Kenna [EMAIL PROTECTED] wrote:
  
 Is there a way to enable call waiting by default in asterisk? Every time I
 create an extension, it is disabled by default. Having to go to every phone
 is becoming quite annoying. I havent restarted the server yet, but I am
 afraid of all my extensions changing back to disabled again. Making it the
 default would just solve all my issues. 
   
 Chris 
   
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Re: [Asterisk-Users] Call waiting

[[EMAIL PROTECTED]]

dial *70

check out the handbook for all the feature codes

http://asteriskathome.sourceforge.net/handbook


--- Sascha Ferley [EMAIL PROTECTED] wrote:
 Hi,
 
 
 I am trying to figure out how to setup call waiting
 on a [EMAIL PROTECTED]
 box. We get the call waiting signal from the telco
 and would like to be
 able to switch calls.
 Our setup right now is as following:
 
 [PSTN] - [EMAIL PROTECTED] - [sip to Cisco ATA 188]
 - Siemens 8825 (Analog)
 
 When we had the siemens plugged into the PSTN
 directly, we could switch by
 just pressing the line button again. Now this
 doesn't seem to work going
 through the sip channel. Does anyone know how to
 enable this such that the
 switch line signal is propagate back the the PSTN
 and be able to switch
 the call to the call waiting one?
 
 
 Please let me know
 
 Thanks
 Sascha
 
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[Asterisk-Users] Call waiting

[Sascha Ferley]

Hi,


I am trying to figure out how to setup call waiting on a [EMAIL PROTECTED]
box. We get the call waiting signal from the telco and would like to be
able to switch calls.
Our setup right now is as following:

[PSTN] - [EMAIL PROTECTED] - [sip to Cisco ATA 188] - Siemens 8825 (Analog)

When we had the siemens plugged into the PSTN directly, we could switch by
just pressing the line button again. Now this doesn't seem to work going
through the sip channel. Does anyone know how to enable this such that the
switch line signal is propagate back the the PSTN and be able to switch
the call to the call waiting one?


Please let me know

Thanks
Sascha

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[Asterisk-Users] Call waiting

[Sascha Ferley]

Hi,

I am trying to figure out how to setup call waiting on a [EMAIL PROTECTED]
box. We get the call waiting signal from the telco and would like to be
able to switch calls.
Our setup right now is as following:

[PSTN] - [EMAIL PROTECTED] - [sip to Cisco ATA 188] - Siemens 8825 (Analog)

When we had the siemens plugged into the PSTN directly, we could switch by
just pressing the line button again. Now this doesn't seem to work going
through the sip channel. Does anyone know how to enable this such that the
switch line signal is propagate back the the PSTN and be able to switch
the call to the call waiting one?


Please let me know

Thanks
Sascha

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Re: [Asterisk-Users] Call waiting

[Henry Devito]

You have to do a flash on the Siemens which gives you * dialtone then Dial 
*0 which flashes the line.  So the steps are flash *0
- Original Message - 
From: Sascha Ferley [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, April 20, 2005 6:32 PM
Subject: [Asterisk-Users] Call waiting


Hi,
I am trying to figure out how to setup call waiting on a [EMAIL PROTECTED]
box. We get the call waiting signal from the telco and would like to be
able to switch calls.
Our setup right now is as following:
[PSTN] - [EMAIL PROTECTED] - [sip to Cisco ATA 188] - Siemens 8825 
(Analog)

When we had the siemens plugged into the PSTN directly, we could switch by
just pressing the line button again. Now this doesn't seem to work going
through the sip channel. Does anyone know how to enable this such that the
switch line signal is propagate back the the PSTN and be able to switch
the call to the call waiting one?
Please let me know
Thanks
Sascha
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[Asterisk-Users] Call Waiting

[Chris]

I have a pretty simple setup. I have a box running
[EMAIL PROTECTED] with an X100P card and 2 Grandstream BudgeTone-100 phones.
I thought it would be rather simple but I seem to be having a hell of a time
getting call waiting to work. I have come across some solutions in the
mailing list that have to do with making an extension that you dial that gives
a Flash() cmd and then dials the extension your on. I have somewhat
gotten this to work, but only to another extension.. 



My question is if it is at all possible to perhaps have the
Flash button on the phone to send the Flash() command that does a flash on the
Zap channel instead of to the asterisk pbx. This would make things much
more simple. I have a feeling this is not possible though. That
being true, I was wondering if someone could please show me some type of
example of what I would need in order to use call waiting with this
phone. 



exten=600,1,Flash()

exten=600,2,Dial(SIP/202)



Currently I have that in extensions.custom.conf and if I am
on a call and another call comes in via call waiting, I press flash and dial
600 it will do the Flash() command and send it to extension 202. Now
Ive only done this from my phone which is extension 200, so that doesnt
help much. But at least I know that it can somewhat be done. 



Also, I have tried pressing flash and *0 which doesnt
seem to be working at all.








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[Asterisk-Users] Call Waiting and FXO

[mj]

What is the best way to get call waiting working with a
TDM11B. Ive tried a macro to flash the line, but I never get call
waiting to work completely. Any suggestions?






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Re: [Asterisk-Users] Call waiting in Australia

[PHP Mechanic]

Has anyone had problems with Call Waiting signals causing Zap channel or
bridging hangups in AU.
I was on a call the other day (Zap channel to PSTN) and the call
suddenly hung up on my side.  I dialled the calling party and got the
call again, it seems that the bridge had dropped and that the other
party had not lost the connection.
As soon as I got the bridging again the other party mentioned that they
had had a call waiting signal immediately before I went off the air.
Any one had similar experiences, or have fixes?
I'm in Australia, I have the same setup, and I had the exact same thing 
happen twice in the space of a few minutes, just then, while calling the 
same person. The person who I was calling says they don't have call waiting 
and were disconnected from me without warning, as I was. I have disabled 
call waiting with my telco. I rebooted asterisk today.

Personally, I've come to the conclusion that these digium cards are a bit 
flaky - dunno? 

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[Asterisk-Users] Call waiting in Australia

[Howard Lowndes]

Has anyone had problems with Call Waiting signals causing Zap channel or
bridging hangups in AU.

I was on a call the other day (Zap channel to PSTN) and the call
suddenly hung up on my side.  I dialled the calling party and got the
call again, it seems that the bridge had dropped and that the other
party had not lost the connection.

As soon as I got the bridging again the other party mentioned that they
had had a call waiting signal immediately before I went off the air.

Any one had similar experiences, or have fixes?

-- 
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.


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[Asterisk-Users] call waiting notification and cisco 7960 phone

[Jeremy Hinton]

This is probably better suited to a cisco forum, but thought i'd drop it 
in here also. Using a 7960 with *, and have a very specific need. If the 
end user is currently on a call on the 7960, and a new call comes in, i 
need the phone to:

show a visual indicator of the call (pref flash the call light)
emit a ringing tone
*NOT* play a call waiting beep inline on the current call.
currently the phones do the exact opposite, ie no notification of an 
incoming call on the phone base (other than on the screen), and they 
play a call waiting tone inline on the current call.

I've tried disabling call waiting on the phone, and configuring the 
dialplan to roll from one line on the phone to the next if the first 
line is busy. This works, but the phone still acts the same way as when 
call waiting is enabled.

If anyone knows if this is possible (and even better how to do it), i 
would greatly appreciate any info. Thanks!

- jeremy
--
Jeremy Hinton A little nonsense
Senior Network Manager   now and then
Continental VisiNet Broadband   is relished by
[EMAIL PROTECTED]the wisest men
757 873 4500
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Re: [Asterisk-Users] Call Waiting on X100P

[Lyle Giese]

http://voip-info.org/tiki-index.php?page=Asterisk%20vertical%20service%20activation%20codes

I have never done this, so YMMV

But searching the wiki can be quite usefull and enlightening.

Lyle

- Original Message - 
From: Derek Whitten [EMAIL PROTECTED]
To: asterisk-users asterisk-users@lists.digium.com
Sent: Saturday, February 05, 2005 9:59 PM
Subject: Re: [Asterisk-Users] Call Waiting on X100P


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Re: [Asterisk-Users] Call Waiting on X100P

[Derek Whitten]

*0 doesn't work.. tried that already...



On Sun, 2005-02-06 at 08:24, Lyle Giese wrote:
 http://voip-info.org/tiki-index.php?page=Asterisk%20vertical%20service%20activation%20codes
 
 I have never done this, so YMMV
 
 But searching the wiki can be quite usefull and enlightening.
 
 Lyle
 
 - Original Message - 
 From: Derek Whitten [EMAIL PROTECTED]
 To: asterisk-users asterisk-users@lists.digium.com
 Sent: Saturday, February 05, 2005 9:59 PM
 Subject: Re: [Asterisk-Users] Call Waiting on X100P
 
 
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[Asterisk-Users] Call Waiting on X100P

[Derek Whitten]

Is there any documentation for getting call waiting to work on an x100p?

I have all the callwaiting settings turned on in zapata.conf.

I get the Call Waiting tones coming through when an incoming call comes
in, but when i try to click over to pick up the other call, i just get a
dialtone like i was trying to make a 3-way call..

* zapata.conf **
[channels]

busydetect=1
busycount=7

relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes

usecallerid=yes

echocancel=yes
echotraining=yes
echocancelwhenbridged=yes

rxgain=8.0
txgain=0.0

immediate=no
context=default
signalling=fxs_ks
callerid=asreceived

channel=1


Thanks in advance for any assistance..

Derek


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Re: [Asterisk-Users] Call Waiting on X100P

[Lyle Giese]

http://voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash

You are wanting to flash the zap trunk, not your analog pbx extension.

Lyle

- Original Message - 
From: Derek Whitten [EMAIL PROTECTED]
To: asterisk-users asterisk-users@lists.digium.com
Sent: Saturday, February 05, 2005 5:37 PM
Subject: [Asterisk-Users] Call Waiting on X100P

Is there any documentation for getting call waiting to work on an x100p?

I have all the callwaiting settings turned on in zapata.conf.

I get the Call Waiting tones coming through when an incoming call comes
in, but when i try to click over to pick up the other call, i just get a
dialtone like i was trying to make a 3-way call..

* zapata.conf **
[channels]

busydetect=1
busycount=7

relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes

usecallerid=yes

echocancel=yes
echotraining=yes
echocancelwhenbridged=yes

rxgain=8.0
txgain=0.0

immediate=no
context=default
signalling=fxs_ks
callerid=asreceived

channel=1


Thanks in advance for any assistance..

Derek


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Re: [Asterisk-Users] Call Waiting on X100P

[Derek Whitten]

Yes, but i was playing with the flash command yesterday and all i was
able to accomplish was to hang up the line and not answer the other
incoming call... 

Are there some call waiting examples available ?


On Sat, 2005-02-05 at 18:16, Lyle Giese wrote:
 http://voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash
 
 You are wanting to flash the zap trunk, not your analog pbx extension.
 
 Lyle
 
 - Original Message - 
 From: Derek Whitten [EMAIL PROTECTED]
 To: asterisk-users asterisk-users@lists.digium.com
 Sent: Saturday, February 05, 2005 5:37 PM
 Subject: [Asterisk-Users] Call Waiting on X100P
 
 Is there any documentation for getting call waiting to work on an x100p?
 
 I have all the callwaiting settings turned on in zapata.conf.
 
 I get the Call Waiting tones coming through when an incoming call comes
 in, but when i try to click over to pick up the other call, i just get a
 dialtone like i was trying to make a 3-way call..
 
 * zapata.conf **
 [channels]
 
 busydetect=1
 busycount=7
 
 relaxdtmf=yes
 callwaiting=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 cancallforward=yes
 
 usecallerid=yes
 
 echocancel=yes
 echotraining=yes
 echocancelwhenbridged=yes
 
 rxgain=8.0
 txgain=0.0
 
 immediate=no
 context=default
 signalling=fxs_ks
 callerid=asreceived
 
 channel=1
 
 
 Thanks in advance for any assistance..
 
 Derek
 
 
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S***I***G***N***A***T***U***R***E***

   ,(),
 ,(,.   huh-huh ,---,,,_
 ()))//((\ Check it out,   ( ))
(\\( \))( \(/)Beavis...we're, ()
/(  \\  like, in ASS-kee.   ()
//   _   \huh-huh-huh (_(_ )
//   \  /\   / (, \)
\   (.  .\  /  |   /   )   )
(, |,) Yeah. heh-heh   |\ /(   )
 \   ^\/^   /  That's COOL! Hey,   (.(.)S  )
 \  / Butt-Head...you're/_   \ )
  \ (--) / an ASS-kee.  \  /__)   ^   \/
   \  --  /   heh-heh   //|
\ __ / )__|
 |  |   //\/\\/\//\/\//\/\\\/\\   |
  __-|__|-__   \  / __-\__|-__
 (  )   BEAVIS AND BUTT-HEAD (  )
 |_|AC//DC|_|  /  \|_| MTVu |_|
 | |  | |   \/\//\/\\/\\/\//\//\/\ TM  | |  | |



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RE: [Asterisk-Users] Call Waiting Audio Prompt

[Alex Barnes]

Thanks for the replies everyone


 how do you expect to get the indication that you have a 
 callwaiting call? 

The whole point is I don't want it.

The beep is a guard that hides the 
 caller-id fsk spill also. So you can't get 
 callwaiting-callerid and not have a beep. 
 

I don't really need that either, users can see whos waiting on hold for
them via the web page so anything caller waiting related is fine but
only as long as it doesn't have negative impact on the call quality.

Setting the indication durations to zero has helped hugely as the sound
is far less intrusive now.

Also all of the end points are essentially SIP phones (or DECT phones
plugged into 2102's) so callerID doesn't need to be passed inbound.

I will try what Jon sugggests:

Zaptel.conf
callwaiting=no
callwaitingcallerid=no

It didn't really occur to me as was looking at configuring the SIP side.


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RE: [Asterisk-Users] Call Waiting Audio Prompt

[Steven Critchfield]

On Mon, 2005-01-31 at 09:37 +, Alex Barnes wrote:
 Thanks for the replies everyone
 
 
  how do you expect to get the indication that you have a 
  callwaiting call? 
 
 The whole point is I don't want it.
 
 The beep is a guard that hides the 
  caller-id fsk spill also. So you can't get 
  callwaiting-callerid and not have a beep. 
  
 
 I don't really need that either, users can see whos waiting on hold for
 them via the web page so anything caller waiting related is fine but
 only as long as it doesn't have negative impact on the call quality.
 
 Setting the indication durations to zero has helped hugely as the sound
 is far less intrusive now.
 
 Also all of the end points are essentially SIP phones (or DECT phones
 plugged into 2102's) so callerID doesn't need to be passed inbound.
 
 I will try what Jon sugggests:
 
 Zaptel.conf
 callwaiting=no
 callwaitingcallerid=no
 
 It didn't really occur to me as was looking at configuring the SIP side.

Then you need to do the tricks in SIP to not send more than one call at
a time to the endpoint. The changes in zaptel.conf won't help here.
-- 
Steven Critchfield [EMAIL PROTECTED]

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[Asterisk-Users] Call Waiting Audio Prompt

[Alex Barnes]

Hi all,
 
Hopefully you can help me.
 
I want to turn off the audio Call Waiting beep that plays during a call.
 
I have found the line in the indications.conf for Call Waiting but apart from 
setting the frequency to zero or the length to zero is there a proper way to 
disable this functionality.
 
thanks very much
 
alex


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Re: [Asterisk-Users] Call Waiting Audio Prompt

[Steven Critchfield]

On Sat, 2005-01-29 at 00:00 +, Alex Barnes wrote:
 Hi all,
  
 Hopefully you can help me.
  
 I want to turn off the audio Call Waiting beep that plays during a
 call.
  
 I have found the line in the indications.conf for Call Waiting but
 apart from setting the frequency to zero or the length to zero is
 there a proper way to disable this functionality.
 
how do you expect to get the indication that you have a callwaiting
call? The beep is a guard that hides the caller-id fsk spill also. So
you can't get callwaiting-callerid and not have a beep. 

 This email and any attached files are confidential and copyright
 protected.  If you are not the addressee, any dissemination,
 distribution or copying of this communication is strictly prohibited.
 Unless otherwise expressly agreed in writing, nothing stated in this
 communication shall be legally binding.

Since you haven't given me written authorization, does that mean your
disclaimer isn't legally binding either? Stupid disclaimer people.

-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] Call Waiting Audio Prompt

[Jon Gabrielson]

assuming a ZAP interface, just set

callwaiting=no
callwaitingcallerid=no

in zapata.conf


Cheers,


Jon.


On Friday 28 January 2005 06:00 pm, Alex Barnes wrote:
 Hi all,

 Hopefully you can help me.

 I want to turn off the audio Call Waiting beep that plays during a call.

 I have found the line in the indications.conf for Call Waiting but apart
 from setting the frequency to zero or the length to zero is there a proper
 way to disable this functionality.

 thanks very much

 alex


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 of this communication is strictly prohibited.  Unless otherwise expressly
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Re: [Asterisk-Users] Call Waiting Audio Prompt

[Jay Wilton]

 assuming a ZAP interface, just set
 
 callwaiting=no
 callwaitingcallerid=no
 
 in zapata.conf

callwaiting options in zapata.conf are ignored with my tdm
cards and the ol x100p.  I set
callwaiting,callwaitingcallerid =no, stop asterisk,unload
modules, reload tdm and zaptel, ztcfg, restart, and I still
get call waiting beeps.
Either I am daft, or its seems to be a Setgroup-Checkgroup
thing from HEAD.

JJay 

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[Asterisk-Users] Call Waiting + Call Transfer Problem

[Miguel]

I have a problem:

When I'm in a call and a second call arrive (call waiting) I can't transfer
the first call. If I press flash the line change to the second call, if I
press flash again the line change to the first call.

How I can transfer a call in this kind of situation ? 

Kind regards,

Miguel


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Re: [Asterisk-Users] Call Waiting + Call Transfer Problem

[Listas]

Miguel,
you can try using # as a way of transfering the call, but that's a blind
transfer meaning that you will be prompted an extension number and the call
will be transfered and that's it, on the other hand pressing flash put's the
call on hold and then let's you dial another call and if you press flash
again you'll conference (the three of you) and if you hang the two calls get
bridged...

bye,
Matt
- Original Message - 
From: Miguel [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, January 10, 2005 7:10 PM
Subject: [Asterisk-Users] Call Waiting + Call Transfer Problem


 I have a problem:

 When I'm in a call and a second call arrive (call waiting) I can't
transfer
 the first call. If I press flash the line change to the second call, if I
 press flash again the line change to the first call.

 How I can transfer a call in this kind of situation ?

 Kind regards,

 Miguel


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RE: [Asterisk-Users] Call Waiting + Call Transfer Problem

[Paul Rodan]

Doesn't that depend on the converter? With my Cisco ATA186, when I call a
party, then flash, call another party, flash back so we're in a 3-way call,
when I hangup, everybody gets dropped.

With my Sipura SPA-2000, when I do the same trick, the other 2 stay
connected. I found the configuration option in the Sipura to enable/disable
this feature, and it's on by default. But Cisco's configuration page is
cryptic at best, and I couldn't find this feature, but I didn't look too
hard. I just like Sipura's.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Listas
Sent: Monday, January 10, 2005 5:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Waiting + Call Transfer Problem

Miguel,
you can try using # as a way of transfering the call, but that's a blind
transfer meaning that you will be prompted an extension number and the call
will be transfered and that's it, on the other hand pressing flash put's the
call on hold and then let's you dial another call and if you press flash
again you'll conference (the three of you) and if you hang the two calls get
bridged...

bye,
Matt
- Original Message - 
From: Miguel [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, January 10, 2005 7:10 PM
Subject: [Asterisk-Users] Call Waiting + Call Transfer Problem


 I have a problem:

 When I'm in a call and a second call arrive (call waiting) I can't
transfer
 the first call. If I press flash the line change to the second call, if I
 press flash again the line change to the first call.

 How I can transfer a call in this kind of situation ?

 Kind regards,

 Miguel


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[Asterisk-Users] call waiting/ 3 way calling

[mohammad]

HI;



I have an Asterisk with 10 "SIP" ip-phones, our pbx 
features are now: Voicemail and Call Transfer.
How can I serve both "Call Waiting / 3 way calling" 
for our SIP Phones.?/


Appreciate Any Help
Mohammad




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Re: [Asterisk-Users] call waiting/ 3 way calling

[Eric Wieling aka ManxPower]

mohammad wrote:
I have an Asterisk with 10 SIP ip-phones, our pbx features are now: Voicemail 
and Call Transfer.
How can I serve both Call Waiting / 3 way calling for our SIP Phones.?/
This is what I call one of the dirty little secrets of SIP.  On SIP 
phones (and H323) all the call control is done by the PHONE itself, not 
by the PBX.  Some SIP phones do not even support 3-way calling or 
supervised transfers (the BT101 comes to mind).  There really isn't 
anything Asterisk can do to make it work if the phone does not support 
the feature.

--Eric
--
I am seeking part or full time employment in the Greater Toronto Area, 
My preference is part time employment with some telecommuting, but all 
offers will be considered. Contact eric at fnords.org.
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[Asterisk-Users] Call Waiting FXS and *

[Philippe Daoust]

I have a small home setup with one Cisco 7960 SIP phone, one WISIP, one 
FXO connected to Bell Canada PSTN, and four FXS connected to POTS phones 
throughout the house.  I also have an account to a SIP based DID provider.

My problem is when I'm on a call on one of the FXS connected phones and 
receive another call either via the PSTN line (assuming the call I'm on 
is using my SIP account) or SIP account I can hear my other phones ring 
but don't get the call waiting tone on the phone I am (it works great on 
the Cisco though).  I can run to another phone and answer the call or 
let it go to VM but I would really like to be able to pick it up using 
the FXS connected phone.

I did a bit of searching and it doesn't seem like it's possible...  Is 
there a way or is it on the roadmap for an eventual feature?  Is this a 
software of hardware limitation?

This is really important, it's seriously affecting the WAF for this 
project...  ;-)  I'm starting to think I should have gone with SPA's to 
support my POTS phones...  :-(

Thanks!
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[Asterisk-Users] Call waiting does not work with g729 codec

[John Lange]

I have purchased the g729 codec from Digium and I'm using it in
conjunction with Cisco 7905 phones.

When a call is placed to an extension which is in use, instead of
sounding the call waiting tone, it causes an error:

-- Executing Dial(SIP/206.XX.XXX.XXX-08f899d8, SIP/204X83|10) in new stack
-- Called 2044X83
-- Got SIP response 488 Not Acceptable Here back from 192.168.1.112

If I change the codec to ulaw the problem goes away and the tone is
played normally.

There are two issues here:

1) The call waiting codec problem.

2) The error causes asterisk to return a NOANSWER status when it
should return something else. More sensible would be BUSY or
UNAVAILABLE.

This is significantly important for me because in this case, when
someone is on the phone the dialplan is supposed to roll-over to the
next available extension. Only when it exhausts all extensions or when
it gets a NOANSWER should it go to voice mail.

In short, BUSY should roll-over. NOANSWER should go to voice mail.

Since the error is returning a NOANSWER status code I can't setup my
dialplan properly.

And one final question, is it possible to disable call waiting?

-- 
John Lange


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[Asterisk-Users] Call Waiting Via Sipura to X100P

[Greg Boehnlein]

Hello,
I am having a hard time sending a Flash Hook via my Analog - 
SPA-2000 - X100P - POTS connection. Anyone have any suggestions?

When I hit flash-hook on the Analog phone the Sipura intercepts it and 
puts the caller on hold. I have no way of sending a flash-hook out the 
X100P to pick up the other caller.

Anyone have any suggestions?

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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[Asterisk-Users] Call Waiting

[Nikhil Jogia]

Hi,

I have just set up an Asterisk box.it sure is a big job to get
everything perfect, especially when you have picky users.

Anyway, the box has 2 X100P's and a couple of sipura spa-2000's
connected to the LAN.

1 of the lines connected to the X100P's goes straight to extension 1000
after a short greeting.

Anyway, say ext 1000 is talking to ext 1001, and the line rings. Ext
1000 hears a small beep every few seconds. This is obviously call
waiting.

My question is how do I answer that incoming call whilst on a call? I
have looked around, tried *0 and even 0*, the flash key, but to no avail
:(



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Re: [Asterisk-Users] Call Waiting

[Steve Totaro]

You are supposed to be able to either press flash or quickly push the actual
hook switch.


- Original Message - 
From: Nikhil Jogia [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 7:54 AM
Subject: [Asterisk-Users] Call Waiting


 Hi,

 I have just set up an Asterisk box.it sure is a big job to get
 everything perfect, especially when you have picky users.

 Anyway, the box has 2 X100P's and a couple of sipura spa-2000's
 connected to the LAN.

 1 of the lines connected to the X100P's goes straight to extension 1000
 after a short greeting.

 Anyway, say ext 1000 is talking to ext 1001, and the line rings. Ext
 1000 hears a small beep every few seconds. This is obviously call
 waiting.

 My question is how do I answer that incoming call whilst on a call? I
 have looked around, tried *0 and even 0*, the flash key, but to no avail
 :(



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RE: [Asterisk-Users] Call Waiting

[Henry Devito]

If this is call waiting on the CO line, I found to flash the CO line you
have to (flash *0) to answer it.  If it is another station calling your
phone while you are on , a normal flash will do. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nikhil Jogia
Sent: Sunday, October 24, 2004 6:54 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Call Waiting

Hi,

I have just set up an Asterisk box.it sure is a big job to get
everything perfect, especially when you have picky users.

Anyway, the box has 2 X100P's and a couple of sipura spa-2000's
connected to the LAN.

1 of the lines connected to the X100P's goes straight to extension 1000
after a short greeting.

Anyway, say ext 1000 is talking to ext 1001, and the line rings. Ext
1000 hears a small beep every few seconds. This is obviously call
waiting.

My question is how do I answer that incoming call whilst on a call? I
have looked around, tried *0 and even 0*, the flash key, but to no avail
:(

If this is call waiting on the CO line, I found to flash the CO line you
have to (flash *0) to answer it.  If it is another station calling your
phone while you are on , a normal flash will do. 


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Re: [Asterisk-Users] Call Waiting

[Steve Totaro]

ah good thinking, i didnt even factor CO call waiting into the equation


- Original Message - 
From: Henry Devito [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 1:19 PM
Subject: RE: [Asterisk-Users] Call Waiting


 If this is call waiting on the CO line, I found to flash the CO line you
 have to (flash *0) to answer it.  If it is another station calling your
 phone while you are on , a normal flash will do.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Nikhil Jogia
 Sent: Sunday, October 24, 2004 6:54 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Call Waiting

 Hi,
 
 I have just set up an Asterisk box.it sure is a big job to get
 everything perfect, especially when you have picky users.
 
 Anyway, the box has 2 X100P's and a couple of sipura spa-2000's
 connected to the LAN.

 1 of the lines connected to the X100P's goes straight to extension 1000
 after a short greeting.

 Anyway, say ext 1000 is talking to ext 1001, and the line rings. Ext
 1000 hears a small beep every few seconds. This is obviously call
 waiting.

 My question is how do I answer that incoming call whilst on a call? I
 have looked around, tried *0 and even 0*, the flash key, but to no avail
 :(

 If this is call waiting on the CO line, I found to flash the CO line you
 have to (flash *0) to answer it.  If it is another station calling your
 phone while you are on , a normal flash will do.


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[Asterisk-Users] Call Waiting Beep

[Kenneth E. Lussier]

All,

I was just informed by someone that they can hear the call waiting beep
on my phone when I get a second call. Has anyone seen this before, and
is there a way to prevent it? I'm using a Cisco 7940 connected to an *
box, which is connected to a channel bank that terminates my POTS lines
(incase that makes any difference).

TIA,
Kenny



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[Asterisk-Users] Call waiting trouble with 7912 cisco phones

[Ludovic Drolez]

Hello !
We have 7912G SIP phones with the 1.02.00 firmware.
*Sometimes* when you call someone who is already on the phone, our PBX receives
immediatly a 302 Moved Temporarily SIP message, so that the 2nd caller is
forwarded to the voicemail instead of waiting 20s (Allow Call Waiting is set to
1, and Forward to VMail Delay to 20).
Since I know that Cisco, won't fix the bug before 20 years, I wondered if I can 
find a work-around thanks to asterisk.

As you see in the following trace:
1- The incoming call is 'broadcast' (SIP/8791SIP/8792SIP/8793)
2- Phones are ringing except one, which sends a 302 message
3- Asterisk immediatly redirects the incoming call to the voicemail
= TRACE 
-- Executing Dial(CAPI[contr2/387508790]/116, 
SIP/8791SIP/8792SIP/8793SIP/8794SIP/8795SIP/8797SIP/8798|20|t) in new stack
-- Called 8791
-- Called 8792
-- Called 8793
-- Called 8794

-- Called 8797
-- Called 8798
-- Got SIP response 302 Moved Temporarily back from 192.168.0.202
-- Now forwarding CAPI[contr2/387508790]/116 to '[EMAIL PROTECTED]' 
(thanks to SIP/8792-b8bc)
-- Executing Wait(Local/[EMAIL PROTECTED],2, 1) in new stack
-- SIP/8797-b246 is ringing
-- SIP/8791-2583 is ringing
-- SIP/8793-e1eb is ringing
-- SIP/8798-cce6 is ringing
-- SIP/8794-3c01 is ringing
-- Executing VoiceMailMain2(Local/[EMAIL PROTECTED],2, 06XXX) 
in new stack


Is there a way to tell asterisk to ignore 302 messages when a call is broadcast 
(A nice Dail option) ?

TIA,
--
Ludovic DROLEZ  Linbox / FreeALter Soft
152 rue de Grigy - Technopole Metz 2000   57070 METZ
tel : 03 87 50 87 90fax : 03 87 75 19 26
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Re: [Asterisk-Users] Call waiting trouble with 7912 cisco phones

[Ludovic Drolez]

Philipp von Klitzing wrote:
 How about this pseudo code:

 [default]
 1,Dial(Sip/1Sip/2)
 2,SetVar(foo=x)
 3,Goto(international,8500,1)
 102,SetVar(foo=x)
 103,Goto(international,8500,1)

[international]
8500,1,GotoIf(foo=x THEN voicemail ELSE callotherphones)
Many thanks for the reply, but with 'callotherphones' I think that there would 
be a loop or a '302 message storm'...

Do you know if I can replace a 'callotherphones' by a 'do nothing, continue 
ringing other phones' ? How could I code that ?

Cheers,
--
Ludovic DROLEZ  Linbox / FreeALter Soft
152 rue de Grigy - Technopole Metz 2000   57070 METZ
tel : 03 87 50 87 90fax : 03 87 75 19 26
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Re: [Asterisk-Users] call waiting, * and FXO

[Ben Wern]

Not in any way a good solution, but what I've done is create an extension that flashs the line, and then returns the call to my sip phone. For example:

[app-flash]
exten = _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM})

[macro-test]
exten = s,1,Answer
exten = s,3,Flash
exten = s,3,Dial(SIP/${ARG2},30,t)
exten = s,4,Dial(SIP/${ARG1},30,t)
exten = s,t,Hangup
exten = s,i,Hangup
exten = s,h,Hangup

Then if you're on a call through the Zap line, and transfer the call to *4, it will flash the line and return it to  SIP extension. I've been trying to get it to auto-detect the SIP extension to return it to, but callerid is different depending on if the call is incoming or outgoing through the Zap. 

Again, not good.. but works in a home environment. I think we'll need in-call triggers to do anything better.

Ben Wern




-- Original Message ---
From: mike jennings [EMAIL PROTECTED] 
To: [EMAIL PROTECTED] 
Sent: Wed, 28 Jul 2004 22:38:41 -0500 
Subject: [Asterisk-Users] call waiting, * and FXO 

 I have been told that the combination of call waiting, * and FXO does and will not work because “Asterisk is a PBX”.  I guess I’d like to hear if this is a hard and fast “no this will not work and here’s why”, or that this currently doesn’t work but with some coding might work. 

 
  

 I’d like to have the option to be able to continue using call waiting with an FXO line (and I know I’m not alone).  I know if I switched to a SIP based connection instead of the FXO this would work, but I currently like my unlimited plan with Vonage.  

 
  

 Would anyone like to enlighten me?
 
 
  

 I have done numerous searches and I’ve included a few postings that were mostly not answered.
 
 
  

 http://lists.digium.com/pipermail/asterisk-users/2004-May/046855.html
 
 http://www.vovida.org/pipermail/mgcp/2001-May/000571.html
 
 
  

 Thanks
 
--- End of Original Message ---

[Asterisk-Users] call waiting, * and FXO

[mike jennings]

I have been told that the combination of call waiting, * and
FXO does and will not work because Asterisk is a PBX. I
guess Id like to hear if this is a hard and fast no this will not
work and heres why, or that this currently doesnt work but
with some coding might work. 



Id like to have the option to be able to continue
using call waiting with an FXO line (and I know Im not alone). I
know if I switched to a SIP based connection instead of the FXO this would
work, but I currently like my unlimited plan with Vonage. 



Would anyone like to enlighten me?



I have done numerous searches and Ive included a few postings
that were mostly not answered.



http://lists.digium.com/pipermail/asterisk-users/2004-May/046855.html

http://www.vovida.org/pipermail/mgcp/2001-May/000571.html



Thanks