[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

[rp]

Hello Marco,

On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote:
>
> hi,
>
> The 
> VoiceMailapplication
>  uses
> */usr/sbin/sendmail* to mail voicemail messages to users. This can be any
> sendmail-compatible MTA. In practice you can use 
> Sendmail,
> Postfix , Exim  or any other
> MTA .
> It is recommended to use the default one that comes with your distribution.


If shall I say I'll use Exim4 here, what do I need to do then?

I would say if you just create your "own" sendmail.sh and place it
> /usr/sbin/sendmail, asterisk will execute it by default, do not forget to
> give permissions for asterisk user to execute it.
>

If I'll create this script, what will be its contents then?  Currently, the
/usr/sbin/sendmail is a symbolic link to /usr/sbin/exim4 for your
information.

Please advice.

GNUbie
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

[rp]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



-- 
Have you noticed that all you need to grow healthy, vigorous grass is a
crack in your sidewalk?

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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

[rp]

Hi guys,

I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better 
for me) and I'm trying to connect them to my provider that provides me 
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I try to connect them they use gsm ... I don't know why ... So, 
that's the reason of my question ...

I just have g.729 on my trunk configuration on sip.conf
I have g.729 licenses from Digium
If I use ulaw on the softphones it connects to the provider through g.729.

Thanks,

Marcelo___
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

[rp]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



-- 
Have you noticed that all you need to grow healthy, vigorous grass is a
crack in your sidewalk?

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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE

[rp]

Hi!

Thank you all for the info!

But I think I haven't explained my scenario well enough.

I am not relaying the calls to SIP.

What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):

1. A CONNECT comes from the PSTN containing a Display IE (which has info
sent by the telco that is used for billing purposes by the PBX) into
span 1 of the asterisk. (The telco is emulating an old billing procedure
based on an impulse count. This was commonly used in analog lines.)

2. The asterisk relays the call to span 2 (which is connected to the PBX).

3. The CONNECT that is sent from span 2 to the PBX does not have the
Display IE. The asterisk strips this IE from the CONNECT message.


This is my problem. Is there a way that i can force the asterisk not to
strip the Display IE?

Thanks and best regards to all,

Óscar Patrício


Anthony Francis wrote:
> Damon Estep wrote:
>   
>> Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is 
>> probably being send before the display IE arrives. The display IE is used 
>> for CNAM delivery, and should not exceed 15 characters.
>>
>> It is very common to put a message in the display IE that indicates that the 
>> CNAM info will be sent in a subsequent Facility IE, and for that you must 
>> wait 1 second.
>>
>> If the ISDN setup actually contains information in the display IE, and that 
>> is not being captured as the CNAM (callerid(name)) you might need to capture 
>> he ISDN messaging to debug it, the telco can usually provide such a trace.
>>
>> I would bet that the display IE contains a information following message, 
>> and what you really want is in the facility IE that follows. Very common, as 
>> is the Wait(1) workaround.
>>
>>
>>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio
>> Sent: Wednesday, July 25, 2007 6:55 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Display IE
>>
>> Hi!
>>
>> I have an Asterisk Box that has 2 E1 connections: one to the PSTN and
>> one to a PBX. It is acting as a telephony gateway. I have a problem: the
>> PSTN sends information in the Display IE (in setup, information ,
>> etc.messages) that the PBX needs por internal processing.
>>
>> The asterisk does not relay the ISDN frames coming from the PSTN. It
>> regenerates them, and when it does, it ommits the Display IE.
>>
>> Is there a way that i can force the asterisk, not to ignore this IE in
>> the ISDN messages?
>>
>> If anyone can shed some light on this issue, i would be very grateful.
>>
>> Thanks in advance,
>>
>> Best regards,
>>
>> Óscar Patrício
>>
>> ___
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> ___
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>> 
> Or trace it yourself with:
> pri intense debug span 1
>
> Make sure you change the 1 to whatever span these calls are coming across.
>
> Anthony
>
> ___
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[asterisk-users] Please unsubscribe or moderate [EMAIL PROTECTED]

[Francesco Peeters (Asterisk)]

All these repeated list replies with "Autoreply: Autoreply: Autoreply:
Autoreply:..." subjects are irritating at best and debilitating at worst!

This makes the list waste bandwidth and my inbox (and the archives too)
unreadable!

Thx!

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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

[rp]

Hi guys,

I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better 
for me) and I'm trying to connect them to my provider that provides me 
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I try to connect them they use gsm ... I don't know why ... So, 
that's the reason of my question ...

I just have g.729 on my trunk configuration on sip.conf
I have g.729 licenses from Digium
If I use ulaw on the softphones it connects to the provider through g.729.

Thanks,

Marcelo___
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

[rp]

Hello Marco,

On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote:
>
> hi,
>
> The 
> VoiceMailapplication
>  uses
> */usr/sbin/sendmail* to mail voicemail messages to users. This can be any
> sendmail-compatible MTA. In practice you can use 
> Sendmail,
> Postfix , Exim  or any other
> MTA .
> It is recommended to use the default one that comes with your distribution.


If shall I say I'll use Exim4 here, what do I need to do then?

I would say if you just create your "own" sendmail.sh and place it
> /usr/sbin/sendmail, asterisk will execute it by default, do not forget to
> give permissions for asterisk user to execute it.
>

If I'll create this script, what will be its contents then?  Currently, the
/usr/sbin/sendmail is a symbolic link to /usr/sbin/exim4 for your
information.

Please advice.

GNUbie
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE

[rp]

Hi!

Thank you all for the info!

But I think I haven't explained my scenario well enough.

I am not relaying the calls to SIP.

What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):

1. A CONNECT comes from the PSTN containing a Display IE (which has info
sent by the telco that is used for billing purposes by the PBX) into
span 1 of the asterisk. (The telco is emulating an old billing procedure
based on an impulse count. This was commonly used in analog lines.)

2. The asterisk relays the call to span 2 (which is connected to the PBX).

3. The CONNECT that is sent from span 2 to the PBX does not have the
Display IE. The asterisk strips this IE from the CONNECT message.


This is my problem. Is there a way that i can force the asterisk not to
strip the Display IE?

Thanks and best regards to all,

Óscar Patrício


Anthony Francis wrote:
> Damon Estep wrote:
>   
>> Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is 
>> probably being send before the display IE arrives. The display IE is used 
>> for CNAM delivery, and should not exceed 15 characters.
>>
>> It is very common to put a message in the display IE that indicates that the 
>> CNAM info will be sent in a subsequent Facility IE, and for that you must 
>> wait 1 second.
>>
>> If the ISDN setup actually contains information in the display IE, and that 
>> is not being captured as the CNAM (callerid(name)) you might need to capture 
>> he ISDN messaging to debug it, the telco can usually provide such a trace.
>>
>> I would bet that the display IE contains a information following message, 
>> and what you really want is in the facility IE that follows. Very common, as 
>> is the Wait(1) workaround.
>>
>>
>>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio
>> Sent: Wednesday, July 25, 2007 6:55 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Display IE
>>
>> Hi!
>>
>> I have an Asterisk Box that has 2 E1 connections: one to the PSTN and
>> one to a PBX. It is acting as a telephony gateway. I have a problem: the
>> PSTN sends information in the Display IE (in setup, information ,
>> etc.messages) that the PBX needs por internal processing.
>>
>> The asterisk does not relay the ISDN frames coming from the PSTN. It
>> regenerates them, and when it does, it ommits the Display IE.
>>
>> Is there a way that i can force the asterisk, not to ignore this IE in
>> the ISDN messages?
>>
>> If anyone can shed some light on this issue, i would be very grateful.
>>
>> Thanks in advance,
>>
>> Best regards,
>>
>> Óscar Patrício
>>
>> ___
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> ___
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>> 
> Or trace it yourself with:
> pri intense debug span 1
>
> Make sure you change the 1 to whatever span these calls are coming across.
>
> Anthony
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   

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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

[rp]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



-- 
Have you noticed that all you need to grow healthy, vigorous grass is a
crack in your sidewalk?

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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

[rp]

Hello Marco,

On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote:
>
> hi,
>
> The 
> VoiceMailapplication
>  uses
> */usr/sbin/sendmail* to mail voicemail messages to users. This can be any
> sendmail-compatible MTA. In practice you can use 
> Sendmail,
> Postfix , Exim  or any other
> MTA .
> It is recommended to use the default one that comes with your distribution.


If shall I say I'll use Exim4 here, what do I need to do then?

I would say if you just create your "own" sendmail.sh and place it
> /usr/sbin/sendmail, asterisk will execute it by default, do not forget to
> give permissions for asterisk user to execute it.
>

If I'll create this script, what will be its contents then?  Currently, the
/usr/sbin/sendmail is a symbolic link to /usr/sbin/exim4 for your
information.

Please advice.

GNUbie
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

[rp]

Hi guys,

I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better 
for me) and I'm trying to connect them to my provider that provides me 
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I try to connect them they use gsm ... I don't know why ... So, 
that's the reason of my question ...

I just have g.729 on my trunk configuration on sip.conf
I have g.729 licenses from Digium
If I use ulaw on the softphones it connects to the provider through g.729.

Thanks,

Marcelo___
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

[rp]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



-- 
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crack in your sidewalk?

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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE

[rp]

Hi!

Thank you all for the info!

But I think I haven't explained my scenario well enough.

I am not relaying the calls to SIP.

What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):

1. A CONNECT comes from the PSTN containing a Display IE (which has info
sent by the telco that is used for billing purposes by the PBX) into
span 1 of the asterisk. (The telco is emulating an old billing procedure
based on an impulse count. This was commonly used in analog lines.)

2. The asterisk relays the call to span 2 (which is connected to the PBX).

3. The CONNECT that is sent from span 2 to the PBX does not have the
Display IE. The asterisk strips this IE from the CONNECT message.


This is my problem. Is there a way that i can force the asterisk not to
strip the Display IE?

Thanks and best regards to all,

Óscar Patrício


Anthony Francis wrote:
> Damon Estep wrote:
>   
>> Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is 
>> probably being send before the display IE arrives. The display IE is used 
>> for CNAM delivery, and should not exceed 15 characters.
>>
>> It is very common to put a message in the display IE that indicates that the 
>> CNAM info will be sent in a subsequent Facility IE, and for that you must 
>> wait 1 second.
>>
>> If the ISDN setup actually contains information in the display IE, and that 
>> is not being captured as the CNAM (callerid(name)) you might need to capture 
>> he ISDN messaging to debug it, the telco can usually provide such a trace.
>>
>> I would bet that the display IE contains a information following message, 
>> and what you really want is in the facility IE that follows. Very common, as 
>> is the Wait(1) workaround.
>>
>>
>>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio
>> Sent: Wednesday, July 25, 2007 6:55 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Display IE
>>
>> Hi!
>>
>> I have an Asterisk Box that has 2 E1 connections: one to the PSTN and
>> one to a PBX. It is acting as a telephony gateway. I have a problem: the
>> PSTN sends information in the Display IE (in setup, information ,
>> etc.messages) that the PBX needs por internal processing.
>>
>> The asterisk does not relay the ISDN frames coming from the PSTN. It
>> regenerates them, and when it does, it ommits the Display IE.
>>
>> Is there a way that i can force the asterisk, not to ignore this IE in
>> the ISDN messages?
>>
>> If anyone can shed some light on this issue, i would be very grateful.
>>
>> Thanks in advance,
>>
>> Best regards,
>>
>> Óscar Patrício
>>
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>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>> 
> Or trace it yourself with:
> pri intense debug span 1
>
> Make sure you change the 1 to whatever span these calls are coming across.
>
> Anthony
>
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

[rp]

Hi guys,

I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better 
for me) and I'm trying to connect them to my provider that provides me 
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I try to connect them they use gsm ... I don't know why ... So, 
that's the reason of my question ...

I just have g.729 on my trunk configuration on sip.conf
I have g.729 licenses from Digium
If I use ulaw on the softphones it connects to the provider through g.729.

Thanks,

Marcelo___
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

[rp]

Hello Marco,

On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote:
>
> hi,
>
> The 
> VoiceMailapplication
>  uses
> */usr/sbin/sendmail* to mail voicemail messages to users. This can be any
> sendmail-compatible MTA. In practice you can use 
> Sendmail,
> Postfix , Exim  or any other
> MTA .
> It is recommended to use the default one that comes with your distribution.


If shall I say I'll use Exim4 here, what do I need to do then?

I would say if you just create your "own" sendmail.sh and place it
> /usr/sbin/sendmail, asterisk will execute it by default, do not forget to
> give permissions for asterisk user to execute it.
>

If I'll create this script, what will be its contents then?  Currently, the
/usr/sbin/sendmail is a symbolic link to /usr/sbin/exim4 for your
information.

Please advice.

GNUbie
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

[rp]

Hi guys,

I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better 
for me) and I'm trying to connect them to my provider that provides me 
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I try to connect them they use gsm ... I don't know why ... So, 
that's the reason of my question ...

I just have g.729 on my trunk configuration on sip.conf
I have g.729 licenses from Digium
If I use ulaw on the softphones it connects to the provider through g.729.

Thanks,

Marcelo___
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

[rp]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



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crack in your sidewalk?

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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

[rp]

Hello Marco,

On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote:
>
> hi,
>
> The 
> VoiceMailapplication
>  uses
> */usr/sbin/sendmail* to mail voicemail messages to users. This can be any
> sendmail-compatible MTA. In practice you can use 
> Sendmail,
> Postfix , Exim  or any other
> MTA .
> It is recommended to use the default one that comes with your distribution.


If shall I say I'll use Exim4 here, what do I need to do then?

I would say if you just create your "own" sendmail.sh and place it
> /usr/sbin/sendmail, asterisk will execute it by default, do not forget to
> give permissions for asterisk user to execute it.
>

If I'll create this script, what will be its contents then?  Currently, the
/usr/sbin/sendmail is a symbolic link to /usr/sbin/exim4 for your
information.

Please advice.

GNUbie
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE

[rp]

Hi!

Thank you all for the info!

But I think I haven't explained my scenario well enough.

I am not relaying the calls to SIP.

What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):

1. A CONNECT comes from the PSTN containing a Display IE (which has info
sent by the telco that is used for billing purposes by the PBX) into
span 1 of the asterisk. (The telco is emulating an old billing procedure
based on an impulse count. This was commonly used in analog lines.)

2. The asterisk relays the call to span 2 (which is connected to the PBX).

3. The CONNECT that is sent from span 2 to the PBX does not have the
Display IE. The asterisk strips this IE from the CONNECT message.


This is my problem. Is there a way that i can force the asterisk not to
strip the Display IE?

Thanks and best regards to all,

Óscar Patrício


Anthony Francis wrote:
> Damon Estep wrote:
>   
>> Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is 
>> probably being send before the display IE arrives. The display IE is used 
>> for CNAM delivery, and should not exceed 15 characters.
>>
>> It is very common to put a message in the display IE that indicates that the 
>> CNAM info will be sent in a subsequent Facility IE, and for that you must 
>> wait 1 second.
>>
>> If the ISDN setup actually contains information in the display IE, and that 
>> is not being captured as the CNAM (callerid(name)) you might need to capture 
>> he ISDN messaging to debug it, the telco can usually provide such a trace.
>>
>> I would bet that the display IE contains a information following message, 
>> and what you really want is in the facility IE that follows. Very common, as 
>> is the Wait(1) workaround.
>>
>>
>>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio
>> Sent: Wednesday, July 25, 2007 6:55 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Display IE
>>
>> Hi!
>>
>> I have an Asterisk Box that has 2 E1 connections: one to the PSTN and
>> one to a PBX. It is acting as a telephony gateway. I have a problem: the
>> PSTN sends information in the Display IE (in setup, information ,
>> etc.messages) that the PBX needs por internal processing.
>>
>> The asterisk does not relay the ISDN frames coming from the PSTN. It
>> regenerates them, and when it does, it ommits the Display IE.
>>
>> Is there a way that i can force the asterisk, not to ignore this IE in
>> the ISDN messages?
>>
>> If anyone can shed some light on this issue, i would be very grateful.
>>
>> Thanks in advance,
>>
>> Best regards,
>>
>> Óscar Patrício
>>
>> ___
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> ___
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>> 
> Or trace it yourself with:
> pri intense debug span 1
>
> Make sure you change the 1 to whatever span these calls are coming across.
>
> Anthony
>
> ___
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Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected

[Erick Perez]

Yes I do. I even did a "pri debug span 1" and when I call the asterisk
box, it sees nothing.


On 7/26/07, Idris AVCI <[EMAIL PROTECTED]> wrote:
> Do you have any extension in default context of your extensions.conf
> file to accept incoming calls ?
> It must be something like;
>
> exten => 12345678,1,Answer()
> exten => 12345678,2,Playback(Welcome)
> ...
>
> 12345678 = The DID number you are calling to reach E1
>
> Idris
>
>
> -Original Message-
> From: Erick Perez [mailto:[EMAIL PROTECTED]
> Sent: Thursday, July 26, 2007 7:03 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming
> calldetected
>
> Hi,
> after many issues we finally managed to make our system do outgoing
> calls with perfect quality.
> However I cannot detect *any* form of incoming call. when I use an
> outside phone to call the E1 connected to the sangoma a102, I
> instantly get a fast busy tone.
>
> My /etc/zaptel.conf is:
> loadzone=us
> defaultzone=us
> #Sangoma A102 port 1 [slot:1 bus:4 span: 1]
> span=1,0,0,ccs,hdb3
> bchan=1-15,17-31
> dchan=16
>
> My /etc/asterisk/zapata.conf is:
> [trunkgroups]
>
> [channels]
> context=default
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> relaxdtmf=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
>
> immediate=no
>
> #include zapata-auto.conf
>
> Zapata-auto.conf has:
> callerid=asreceived
> ;Sangoma A102 port 1 [slot:1 bus:4 span: 1]
> switchtype=euroisdn
> context=from-pstn
> group=0
> signalling=pri_cpe
> channel => 1-15,17-31
>
> Note:
> According to the tech support in the local telco, my E1 should be:
> E1 PRI, CAS, HDB3, NCRC4, DSS1
> However if I configure the card for CAS, it will never connect.
> My card is currently configured (and makes only outgoing calls) as:
> E1 PRI, CCS, HDB3,NCRC4  (i have no idea what dss1 is or where it goes)
>
> My /etc/wanpipe/wanpipe1.conf is:
> [devices]
> wanpipe1 = WAN_AFT_TE1, Comment
>
> [interfaces]
> w1g1 = wanpipe1, , TDM_VOICE, Comment
>
> [wanpipe1]
> CARD_TYPE   = AFT
> S514CPU = A
> CommPort= PRI
> AUTO_PCISLOT= NO
> PCISLOT = 1
> PCIBUS  = 4
> FE_MEDIA= E1
> FE_LCODE= HDB3
> FE_FRAME= NCRC4
> FE_LINE = 1
> TE_CLOCK= NORMAL
> TE_REF_CLOCK= 0
> TE_SIG_MODE = CCS
> TE_HIGHIMPEDANCE= NO
> LBO = 120OH
> FE_TXTRISTATE   = NO
> MTU = 1500
> UDPPORT = 9000
> TTL = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN   = 1
> TDMV_DCHAN  = 16
>
> [w1g1]
> ACTIVE_CH   = ALL
> TDMV_ECHO_OFF   = NO
> TDMV_HWEC   = YES
>
> thanks for your help.
>
>
> --
> 
> Erick Perez
> 
>
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>
>
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-- 

Erick Perez



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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE

[rp]

Hi!

Thank you all for the info!

But I think I haven't explained my scenario well enough.

I am not relaying the calls to SIP.

What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):

1. A CONNECT comes from the PSTN containing a Display IE (which has info
sent by the telco that is used for billing purposes by the PBX) into
span 1 of the asterisk. (The telco is emulating an old billing procedure
based on an impulse count. This was commonly used in analog lines.)

2. The asterisk relays the call to span 2 (which is connected to the PBX).

3. The CONNECT that is sent from span 2 to the PBX does not have the
Display IE. The asterisk strips this IE from the CONNECT message.


This is my problem. Is there a way that i can force the asterisk not to
strip the Display IE?

Thanks and best regards to all,

Óscar Patrício


Anthony Francis wrote:
> Damon Estep wrote:
>   
>> Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is 
>> probably being send before the display IE arrives. The display IE is used 
>> for CNAM delivery, and should not exceed 15 characters.
>>
>> It is very common to put a message in the display IE that indicates that the 
>> CNAM info will be sent in a subsequent Facility IE, and for that you must 
>> wait 1 second.
>>
>> If the ISDN setup actually contains information in the display IE, and that 
>> is not being captured as the CNAM (callerid(name)) you might need to capture 
>> he ISDN messaging to debug it, the telco can usually provide such a trace.
>>
>> I would bet that the display IE contains a information following message, 
>> and what you really want is in the facility IE that follows. Very common, as 
>> is the Wait(1) workaround.
>>
>>
>>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio
>> Sent: Wednesday, July 25, 2007 6:55 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Display IE
>>
>> Hi!
>>
>> I have an Asterisk Box that has 2 E1 connections: one to the PSTN and
>> one to a PBX. It is acting as a telephony gateway. I have a problem: the
>> PSTN sends information in the Display IE (in setup, information ,
>> etc.messages) that the PBX needs por internal processing.
>>
>> The asterisk does not relay the ISDN frames coming from the PSTN. It
>> regenerates them, and when it does, it ommits the Display IE.
>>
>> Is there a way that i can force the asterisk, not to ignore this IE in
>> the ISDN messages?
>>
>> If anyone can shed some light on this issue, i would be very grateful.
>>
>> Thanks in advance,
>>
>> Best regards,
>>
>> Óscar Patrício
>>
>> ___
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>> 
> Or trace it yourself with:
> pri intense debug span 1
>
> Make sure you change the 1 to whatever span these calls are coming across.
>
> Anthony
>
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

[rp]

Hi guys,

I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better 
for me) and I'm trying to connect them to my provider that provides me 
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I try to connect them they use gsm ... I don't know why ... So, 
that's the reason of my question ...

I just have g.729 on my trunk configuration on sip.conf
I have g.729 licenses from Digium
If I use ulaw on the softphones it connects to the provider through g.729.

Thanks,

Marcelo___
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

[rp]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



-- 
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crack in your sidewalk?

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[asterisk-users] Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

[rp]

Hello Marco,

On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote:
>
> hi,
>
> The 
> VoiceMailapplication
>  uses
> */usr/sbin/sendmail* to mail voicemail messages to users. This can be any
> sendmail-compatible MTA. In practice you can use 
> Sendmail,
> Postfix , Exim  or any other
> MTA .
> It is recommended to use the default one that comes with your distribution.


If shall I say I'll use Exim4 here, what do I need to do then?

I would say if you just create your "own" sendmail.sh and place it
> /usr/sbin/sendmail, asterisk will execute it by default, do not forget to
> give permissions for asterisk user to execute it.
>

If I'll create this script, what will be its contents then?  Currently, the
/usr/sbin/sendmail is a symbolic link to /usr/sbin/exim4 for your
information.

Please advice.

GNUbie
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

[rp]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



-- 
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crack in your sidewalk?

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[asterisk-users] Autoreply: Autoreply: Autoreply: Re: Display IE

[rp]

Hi!

Thank you all for the info!

But I think I haven't explained my scenario well enough.

I am not relaying the calls to SIP.

What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):

1. A CONNECT comes from the PSTN containing a Display IE (which has info
sent by the telco that is used for billing purposes by the PBX) into
span 1 of the asterisk. (The telco is emulating an old billing procedure
based on an impulse count. This was commonly used in analog lines.)

2. The asterisk relays the call to span 2 (which is connected to the PBX).

3. The CONNECT that is sent from span 2 to the PBX does not have the
Display IE. The asterisk strips this IE from the CONNECT message.


This is my problem. Is there a way that i can force the asterisk not to
strip the Display IE?

Thanks and best regards to all,

Óscar Patrício


Anthony Francis wrote:
> Damon Estep wrote:
>   
>> Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is 
>> probably being send before the display IE arrives. The display IE is used 
>> for CNAM delivery, and should not exceed 15 characters.
>>
>> It is very common to put a message in the display IE that indicates that the 
>> CNAM info will be sent in a subsequent Facility IE, and for that you must 
>> wait 1 second.
>>
>> If the ISDN setup actually contains information in the display IE, and that 
>> is not being captured as the CNAM (callerid(name)) you might need to capture 
>> he ISDN messaging to debug it, the telco can usually provide such a trace.
>>
>> I would bet that the display IE contains a information following message, 
>> and what you really want is in the facility IE that follows. Very common, as 
>> is the Wait(1) workaround.
>>
>>
>>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio
>> Sent: Wednesday, July 25, 2007 6:55 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Display IE
>>
>> Hi!
>>
>> I have an Asterisk Box that has 2 E1 connections: one to the PSTN and
>> one to a PBX. It is acting as a telephony gateway. I have a problem: the
>> PSTN sends information in the Display IE (in setup, information ,
>> etc.messages) that the PBX needs por internal processing.
>>
>> The asterisk does not relay the ISDN frames coming from the PSTN. It
>> regenerates them, and when it does, it ommits the Display IE.
>>
>> Is there a way that i can force the asterisk, not to ignore this IE in
>> the ISDN messages?
>>
>> If anyone can shed some light on this issue, i would be very grateful.
>>
>> Thanks in advance,
>>
>> Best regards,
>>
>> Óscar Patrício
>>
>> ___
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> ___
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>> 
> Or trace it yourself with:
> pri intense debug span 1
>
> Make sure you change the 1 to whatever span these calls are coming across.
>
> Anthony
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   

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[asterisk-users] Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

[rp]

Hi guys,

I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better 
for me) and I'm trying to connect them to my provider that provides me 
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I try to connect them they use gsm ... I don't know why ... So, 
that's the reason of my question ...

I just have g.729 on my trunk configuration on sip.conf
I have g.729 licenses from Digium
If I use ulaw on the softphones it connects to the provider through g.729.

Thanks,

Marcelo___
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[asterisk-users] Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails

[rp]

Hello Marco,

On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote:
>
> hi,
>
> The 
> VoiceMailapplication
>  uses
> */usr/sbin/sendmail* to mail voicemail messages to users. This can be any
> sendmail-compatible MTA. In practice you can use 
> Sendmail,
> Postfix , Exim  or any other
> MTA .
> It is recommended to use the default one that comes with your distribution.


If shall I say I'll use Exim4 here, what do I need to do then?

I would say if you just create your "own" sendmail.sh and place it
> /usr/sbin/sendmail, asterisk will execute it by default, do not forget to
> give permissions for asterisk user to execute it.
>

If I'll create this script, what will be its contents then?  Currently, the
/usr/sbin/sendmail is a symbolic link to /usr/sbin/exim4 for your
information.

Please advice.

GNUbie
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[asterisk-users] Autoreply: Re: Attaching VoiceMails on E-Mails

[rp]

Hello Marco,

On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote:
>
> hi,
>
> The 
> VoiceMailapplication
>  uses
> */usr/sbin/sendmail* to mail voicemail messages to users. This can be any
> sendmail-compatible MTA. In practice you can use 
> Sendmail,
> Postfix , Exim  or any other
> MTA .
> It is recommended to use the default one that comes with your distribution.


If shall I say I'll use Exim4 here, what do I need to do then?

I would say if you just create your "own" sendmail.sh and place it
> /usr/sbin/sendmail, asterisk will execute it by default, do not forget to
> give permissions for asterisk user to execute it.
>

If I'll create this script, what will be its contents then?  Currently, the
/usr/sbin/sendmail is a symbolic link to /usr/sbin/exim4 for your
information.

Please advice.

GNUbie
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

[rp]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



-- 
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crack in your sidewalk?

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[asterisk-users] Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

[rp]

Hi guys,

I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better 
for me) and I'm trying to connect them to my provider that provides me 
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I try to connect them they use gsm ... I don't know why ... So, 
that's the reason of my question ...

I just have g.729 on my trunk configuration on sip.conf
I have g.729 licenses from Digium
If I use ulaw on the softphones it connects to the provider through g.729.

Thanks,

Marcelo___
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[asterisk-users] Autoreply: Autoreply: Re: Display IE

[rp]

Hi!

Thank you all for the info!

But I think I haven't explained my scenario well enough.

I am not relaying the calls to SIP.

What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):

1. A CONNECT comes from the PSTN containing a Display IE (which has info
sent by the telco that is used for billing purposes by the PBX) into
span 1 of the asterisk. (The telco is emulating an old billing procedure
based on an impulse count. This was commonly used in analog lines.)

2. The asterisk relays the call to span 2 (which is connected to the PBX).

3. The CONNECT that is sent from span 2 to the PBX does not have the
Display IE. The asterisk strips this IE from the CONNECT message.


This is my problem. Is there a way that i can force the asterisk not to
strip the Display IE?

Thanks and best regards to all,

Óscar Patrício


Anthony Francis wrote:
> Damon Estep wrote:
>   
>> Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is 
>> probably being send before the display IE arrives. The display IE is used 
>> for CNAM delivery, and should not exceed 15 characters.
>>
>> It is very common to put a message in the display IE that indicates that the 
>> CNAM info will be sent in a subsequent Facility IE, and for that you must 
>> wait 1 second.
>>
>> If the ISDN setup actually contains information in the display IE, and that 
>> is not being captured as the CNAM (callerid(name)) you might need to capture 
>> he ISDN messaging to debug it, the telco can usually provide such a trace.
>>
>> I would bet that the display IE contains a information following message, 
>> and what you really want is in the facility IE that follows. Very common, as 
>> is the Wait(1) workaround.
>>
>>
>>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio
>> Sent: Wednesday, July 25, 2007 6:55 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Display IE
>>
>> Hi!
>>
>> I have an Asterisk Box that has 2 E1 connections: one to the PSTN and
>> one to a PBX. It is acting as a telephony gateway. I have a problem: the
>> PSTN sends information in the Display IE (in setup, information ,
>> etc.messages) that the PBX needs por internal processing.
>>
>> The asterisk does not relay the ISDN frames coming from the PSTN. It
>> regenerates them, and when it does, it ommits the Display IE.
>>
>> Is there a way that i can force the asterisk, not to ignore this IE in
>> the ISDN messages?
>>
>> If anyone can shed some light on this issue, i would be very grateful.
>>
>> Thanks in advance,
>>
>> Best regards,
>>
>> Óscar Patrício
>>
>> ___
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> ___
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>   
>> 
> Or trace it yourself with:
> pri intense debug span 1
>
> Make sure you change the 1 to whatever span these calls are coming across.
>
> Anthony
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   

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[asterisk-users] Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?

[rp]

Hi guys,

I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better 
for me) and I'm trying to connect them to my provider that provides me 
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I try to connect them they use gsm ... I don't know why ... So, 
that's the reason of my question ...

I just have g.729 on my trunk configuration on sip.conf
I have g.729 licenses from Digium
If I use ulaw on the softphones it connects to the provider through g.729.

Thanks,

Marcelo___
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[asterisk-users] Autoreply: Re: Display IE

[rp]

Hi!

Thank you all for the info!

But I think I haven't explained my scenario well enough.

I am not relaying the calls to SIP.

What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):

1. A CONNECT comes from the PSTN containing a Display IE (which has info
sent by the telco that is used for billing purposes by the PBX) into
span 1 of the asterisk. (The telco is emulating an old billing procedure
based on an impulse count. This was commonly used in analog lines.)

2. The asterisk relays the call to span 2 (which is connected to the PBX).

3. The CONNECT that is sent from span 2 to the PBX does not have the
Display IE. The asterisk strips this IE from the CONNECT message.


This is my problem. Is there a way that i can force the asterisk not to
strip the Display IE?

Thanks and best regards to all,

Óscar Patrício


Anthony Francis wrote:
> Damon Estep wrote:
>   
>> Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is 
>> probably being send before the display IE arrives. The display IE is used 
>> for CNAM delivery, and should not exceed 15 characters.
>>
>> It is very common to put a message in the display IE that indicates that the 
>> CNAM info will be sent in a subsequent Facility IE, and for that you must 
>> wait 1 second.
>>
>> If the ISDN setup actually contains information in the display IE, and that 
>> is not being captured as the CNAM (callerid(name)) you might need to capture 
>> he ISDN messaging to debug it, the telco can usually provide such a trace.
>>
>> I would bet that the display IE contains a information following message, 
>> and what you really want is in the facility IE that follows. Very common, as 
>> is the Wait(1) workaround.
>>
>>
>>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio
>> Sent: Wednesday, July 25, 2007 6:55 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Display IE
>>
>> Hi!
>>
>> I have an Asterisk Box that has 2 E1 connections: one to the PSTN and
>> one to a PBX. It is acting as a telephony gateway. I have a problem: the
>> PSTN sends information in the Display IE (in setup, information ,
>> etc.messages) that the PBX needs por internal processing.
>>
>> The asterisk does not relay the ISDN frames coming from the PSTN. It
>> regenerates them, and when it does, it ommits the Display IE.
>>
>> Is there a way that i can force the asterisk, not to ignore this IE in
>> the ISDN messages?
>>
>> If anyone can shed some light on this issue, i would be very grateful.
>>
>> Thanks in advance,
>>
>> Best regards,
>>
>> Óscar Patrício
>>
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>> 
> Or trace it yourself with:
> pri intense debug span 1
>
> Make sure you change the 1 to whatever span these calls are coming across.
>
> Anthony
>
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[asterisk-users] Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

[rp]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



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[asterisk-users] Query

[sanchal . singh]

Hi,
  Do the following steps are required while configuring D-channel

  1)  In zconfig.h file of zaptel package
  uncomment #define CONFIG_ZAPATA_NET
  make sethdlc-new
  make install
 2)   modprobe wcte12xp
  ztcfg 

 3)   sethdlc hdlc0 cisco   
  Step 3 is giving error hdlc0: Unable to set Cisco HDLC protocol   
information: No such device 
 
Can anybody tell, how to overcome this error.
   Thanx and regards,
   sanchal 
 



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[asterisk-users] Is it possible transcode (ilbc <-> g.729) in software ?

[marcelobiz]

Hi guys,

I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better 
for me) and I'm trying to connect them to my provider that provides me 
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I try to connect them they use gsm ... I don't know why ... So, 
that's the reason of my question ...

I just have g.729 on my trunk configuration on sip.conf
I have g.729 licenses from Digium
If I use ulaw on the softphones it connects to the provider through g.729.

Thanks,

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[asterisk-users] Autoreply: Autoreply: Re: Dialtone when automatically picking up.

[rp]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



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[asterisk-users] Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.

[rp]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



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[asterisk-users] Autoreply: Re: Dialtone when automatically picking up.

[rp]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



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Re: [asterisk-users] Asterisk Conference Call

[Nate]

That's actually a good idea.

- Original Message - 
From: "C F" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Thursday, July 26, 2007 9:23 PM
Subject: Re: [asterisk-users] Asterisk Conference Call


> Yes, have them all meet in the cafeteria for brunch.
>
> On 7/26/07, satish patel <[EMAIL PROTECTED]> wrote:
>> Dear all
>>
>>   I have asterisk with SNOM SIP phone i want to confrance to 
>> my
>> users how to configure confranceing in asterisk meetme.conf is fine but 
>> is
>> there any otherway to confranceing
>>
>>
>>
>>
>>
>>  
>> Got a little couch potato?
>>  Check out fun summer activities for kids.
>>
>>
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>
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[asterisk-users] Query

[sanchal . singh]

Hi,
  I am facing problem in configuring D-channel. I did the following
configuration for TE-120P card
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16

   /etc/asterisk/zaptel.conf
group=1
signalling=pri_cpe
switchtype=euroisdn
context=incoming
channel=1-15,17-31

DIGIUM card is connected through cable to another end.On placing
call from other end to asterisk PBX ( through DIGIUM card ) the
following error messages is coming on console mode of asterisk
(The OTHER END CONNECTED to DIGIUM is E1 CARD RUNNING APPLICATION)
 
Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel:  
PRI Error: We think we're the CPE, but they think they're the   CPE too.

  == Primary D-Channel on span 1 down
Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan:
No D-channels available!  Using Primary channel 16 as   D-channel   
anyway!

NOTE-> The OTHER END CONNECTED to DIGIUM is E1 CARD RUNNING APPLICATION

Can anybody tell me how to overcome this error.
Thanx and Regards
sanchal



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[asterisk-users] Autoreply: Queue stats

[rp]

Greetings, list!

My boss would like some statistics on how many calls are answered out of 
specific queues during a given time period, and I'm not sure how exactly 
to obtain those stats.  Here's how our queue system works.

1) Call comes in and enters our 'ring' queue where the phones ring for 
15 seconds (caller hears the standard ring tone).

2) After 15 seconds, the caller falls into our 'music on hold' queue, a 
message is played and the caller hears our music on hold while the 
phones are rung again.

3) After 30 seconds, if the caller is still in our 'moh' queue, they 
drop out of queue and immediately re-enter the 'moh' queue again until 
the call is answered or the caller hangs up.

How can I find out how many calls are answered out of each queue during 
certain times (1st shift, 2nd shift, etc...)?  Also, I'm curious how I 
can track how many times a call repeats the 'moh' queue.

Thanks in advance,
Jay

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[asterisk-users] Autoreply: Re: Newbie Advice on Asterisk and Linux

[rp]

Mark Burrows wrote:
> HI All,
> 
>  
> 
> I’m new to Asterisk and also to Linux.  I have a large IVR project that 
> I’m about to embark on.  I’m new to programming; new to Linux and new to 
> Asterisk.  I think I’m about to climb a steep learning curve.  I have an 
> existing IVR which is getting on for nine years old and is no longer 
> supported by my vendor.  I intend to replicate the system almost as is 
> and then add additional features and functions. 
> 
>  
> 
> I have been looking for a developer to put together my project and while 
> doing so have done lots of research and spoken to many people.  The 
> people who seem to understand my needs have recommended Asterisk.  For 
> the last couple of days I’ve been trying to look into Asterisk and learn 
> as much as I can; this has got me excited, motivated and a little 
> confused. Asterisk sounds like a great project and a great community.  I 
> think I have as much of an overview as I can.  Now I need to set up a 
> Linux system and get Asterisk running on it.
> 
>  
> 
> I’ve started to read the book Asterisk: The Future Of Telephony and 
> would like to now setup up a hobby computer to do some hands on 
> learning.  The book covers Red Hat Linux so I thought I’d look for a 
> ‘Red Had for Dummies’ book.  Even that got confusing. There’s Linux 
> Fedora, Enterprise Linux 4 and others.
> 
>  
> 
> Can someone suggest a starting point on learning Linux?
> 
>  
> 
> Thanks in advance,
> 
>  
> 
>  
> 
> Mark
> 
>  
> 
>  
> 
>  
> 
> 
> No virus found in this outgoing message.
> Checked by AVG Free Edition.
> Version: 7.5.476 / Virus Database: 269.10.20/919 - Release Date: 
> 26/07/2007 9:56 AM
> 
> 
> 
> 
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http://www.trixbox.org

This is "one" of the many standard configs for aterisk. This uses CentOS 
4, Asterisk 1.2, FreePBX 2.2. You can setup a fully working system in 
about 30 min. Need help you can email me off list.

Jonn Taylor

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Re: [asterisk-users] polycom custom ring tones (slightly OT)

[Matt Gibson]

Hi James,

I have one posting for the Cisco7970 ringtone, which you can adapt for
the Polycom. It's here: http://www.voipphreak.ca/archives/349

I also have another one I posted for the Polycom Ringtones with a
bunch of tunes. It's here:
http://www.voipphreak.ca/archives/78

Hope these help :)

Thanks,
Matt

On 27/07/07, James Andrewartha <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> Has anyone made up custom ring tones for the Polycom SIP phones? We use
> different rings for different lines, but the ones it comes with are all very
> similar. In the interesting of sharing, here's one I made up for paging:
>
>  se.pat.ringer.13.inst.1.type="chord" se.pat.ringer.13.inst.1.value="12"
> se.pat.ringer.13.inst.1.param="200" se.pat.ringer.13.inst.2.type="chord"
> se.pat.ringer.13.inst.2.value="15" se.pat.ringer.13.inst.2.param="600"
> se.pat.ringer.13.inst.3.type="branch" se.pat.ringer.13.inst.3.value="-2"/>
>
> Alternatively, since you can use .wav files for ring tones, do people have
> any recommendations for where to find some distinctive rings?
>
> Thanks,
>
> --
> James Andrewartha
> Systems Administrator
> Data Analysis Australia Pty Ltd
>
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[asterisk-users] polycom custom ring tones (slightly OT)

[James Andrewartha]

Hi all,

Has anyone made up custom ring tones for the Polycom SIP phones? We use
different rings for different lines, but the ones it comes with are all very
similar. In the interesting of sharing, here's one I made up for paging:



Alternatively, since you can use .wav files for ring tones, do people have
any recommendations for where to find some distinctive rings?

Thanks,

-- 
James Andrewartha
Systems Administrator
Data Analysis Australia Pty Ltd 

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Re: [asterisk-users] Asterisk Conference Call

[C F]

Yes, have them all meet in the cafeteria for brunch.

On 7/26/07, satish patel <[EMAIL PROTECTED]> wrote:
> Dear all
>
>   I have asterisk with SNOM SIP phone i want to confrance to my
> users how to configure confranceing in asterisk meetme.conf is fine but is
> there any otherway to confranceing
>
>
>
>
>
>  
> Got a little couch potato?
>  Check out fun summer activities for kids.
>
>
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[asterisk-users] Autoreply: Re: Queue stats

[rp]

On Thu, 2007-07-26 at 09:37 -0500, Jay Moore wrote:
> My boss would like some statistics on how many calls are answered out of 
> specific queues during a given time period, and I'm not sure how exactly 
> to obtain those stats.  

It sounds like you've got quite the queue setup (although I don't quite
see why your calls jump out and back into the moh queue).  All the of
queue statistics you need should be available with careful parsing of
the queue log (usually located in /var/log/asterisk/queue_log).  You can
also trigger custom queue log events from the dialplan by calling the
QueueLog() application.  In your case, you might want to add a custom
queue log entry every time the caller rejoins the moh queue, saying
something to the effect of "Caller XYZ has rejoined the moh queue for
the 10th time" or something like that.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Need help with inbound IAX

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Patrick Buller wrote:
>> What do you get with:
>>
>> iax2 show registry
>>   
> 
> homer*CLI> iax2 show registry
> Host  UsernamePerceived Refresh  State
> 64.85.162.136:456906***   68.XX.XX.XX:4569  300  Registered
> 
> is that bad?

Nope, that's good.  It means you have registered to their server no problem.

Firstly, which version of Asterisk are you using?

If you turn on iax2 debug, and then say call from your cellphone to the
DDI you have registered do you get anything at all?

- --
Cheers,

Matt Riddell
Director
___

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http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
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Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFGqXG1DQNt8rg0Kp4RAn1rAKC/WwYnvDaqQ9FK3YXmiWEkwkiwUwCfWin8
nciXBwS2Ws+lg/6P8gv5XRI=
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Re: [asterisk-users] Default Asterisk Numbers

[Nasir Iqbal]

also have a look on

http://www.voip-info.org/wiki/view/Asterisk+standard+extensions


On Thu, 2007-07-26 at 20:57 -0600, Al lists wrote:
> features.conf
> 
> On 7/26/07, GNUbie <[EMAIL PROTECTED]> wrote:
> Hello all,
> 
> Where can I find the complete list of default Asterisk
> (telephone) numbers and maybe the other special numbers that
> are need to be preserve and not use for setting up own dial
> plan?
> 
> Thank you.
> 
> GNUbie
> 
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Re: [asterisk-users] Asterisk Conference Call

[ram]

On 7/26/07, satish patel <[EMAIL PROTECTED]> wrote:


Dear all

  I have asterisk with SNOM SIP phone i want to confrance to
my users how to configure confranceing in asterisk meetme.conf is fine but
is there any otherway to confranceing




If the End device support conference still you can do that

ram
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Re: [asterisk-users] Need help with inbound IAX

[Patrick Buller]

> What do you get with:
>
> iax2 show registry
>   

homer*CLI> iax2 show registry
Host  UsernamePerceived Refresh  State
64.85.162.136:456906***   68.XX.XX.XX:4569  300  Registered

is that bad?

Patrick

  

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Re: [asterisk-users] Default Asterisk Numbers

[Al lists]

features.conf

On 7/26/07, GNUbie <[EMAIL PROTECTED]> wrote:
>
> Hello all,
>
> Where can I find the complete list of default Asterisk (telephone) numbers
> and maybe the other special numbers that are need to be preserve and not use
> for setting up own dial plan?
>
> Thank you.
>
> GNUbie
>
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Re: [asterisk-users] Need help with inbound IAX

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

What do you get with:

iax2 show registry

- --
Cheers,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFGqVvRDQNt8rg0Kp4RAj89AJ9XzL/EUCBMG3/qfDkhHMSJggoIWQCggAMu
e4tnaTrjg6wRI8kIyOS0Qgo=
=AbcT
-END PGP SIGNATURE-

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[asterisk-users] Need help with inbound IAX

[Patrick Buller]

I have just started working with Asterisk and have run into a road block 
concerning IAX and an inbound DID from callwithus.com. I am getting 
nowhere and I don't really know how to isolate the problem. The asterisk 
version is 1.2.7 on ubuntu, sits behind a firewall with iptables. I can 
connect and make a call to other internal extensions using zoiper and 
iax. When I try and use the number, I do not see any traffic on the 
firewall from the provider, so I think it is config string I have put in 
at callwithus, but I have tried so many things anymore, I am not sure 
anymore. Any help is much appreciated.

Thanks,
Patrick

I have the following rules on the firewall:
-A FORWARD -p udp -m udp --dport 4569 -j ACCEPT
-A PREROUTING -i eth2 -p udp -m udp --dport 4569 -j DNAT 
--to-destination 192.168.1.2

I have similar rules for port 25 and 80 that work.

On the asterisk machine, iax.conf looks like:
[general]
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes

register => 45*:[EMAIL PROTECTED]

[iaxin]
type=friend
context=iax-inbound
username=iaxin
secret=easypass
qualify=no
host=callwithus.com


in extensions.conf:
[iax-inbound]
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,Hangup()

This is what callwithus is supposed to forward the call to:
IAX/iaxin:[EMAIL PROTECTED]/[EMAIL PROTECTED]



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Re: [asterisk-users] Attaching VoiceMails on E-Mails

[Eric \"ManxPower\" Wieling]

GNUbie wrote:
> Hello Tzafrir,
> 
> On 7/27/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> 
>>
>> You need to have a package that provides /usr/sbin/sendmail . While you
>> can get away with using nullmailer or ssmtp (that don't spool mail
>> locally), I would recommend you to install postfix or exim, so a
>> temporary problem won't cause the message to get lost on the way.
>>
> 
> Oh, I mean, I have Exim installed here but it's not running by default.
> Should I run it anyway?

Yes.  You should be able to tell Exim to only listen on the 127.0.0.1 
interface.  I'm sure you can tell Exim to not listen on any interface, 
but don't ask me how 8-)

You might want to tell Exim to send all e-mail thru your main SMTP 
server machine (the correct term in most MTA docs is "smarthost".

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[asterisk-users] Default Asterisk Numbers

[GNUbie]

Hello all,

Where can I find the complete list of default Asterisk (telephone) numbers
and maybe the other special numbers that are need to be preserve and not use
for setting up own dial plan?

Thank you.

GNUbie
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[asterisk-users] Problems with new logic being 'n' option to Queue in 1.4.9

[James FitzGibbon]

I am experiencing a change in behaviour of my Queues in 1.4.9 vs 1.4.8.

I do not pass the 'n' option to any call to Queue() in my dialplan.  Yet
since I upgraded to 1.4.9, I have occasionally seen this on my console:

   -- Nobody picked up in 2 ms
   -- Exiting on time-out cycle

That log message "Exiting on time-out cycle" is exclusive to the logic in
app_queue meant to deal with the 'n' option.  If you don't pass 'n', you
should never see it.

1.4.8 code:

   /* exit after 'timeout' cycle if 'n' option
enabled */
   if (go_on) {
   if (option_verbose > 2)
   ast_verbose(VERBOSE_PREFIX_3
"Exiting on time-out cycle\n");
   ast_queue_log(args.queuename,
chan->uniqueid, "NONE", "EXITWITHTIMEOUT", "%d", qe.pos);
   record_abandoned(&qe);
   reason = QUEUE_TIMEOUT;
   res = 0;
   break;
   }

1.4.9 code:

   /* exit after 'timeout' cycle if 'n' option enabled
*/
   if (go_on >= qe.parent->membercount) {
   if (option_verbose > 2)
   ast_verbose(VERBOSE_PREFIX_3
"Exiting on time-out cycle\n");
   ast_queue_log(args.queuename,
chan->uniqueid, "NONE", "EXITWITHTIMEOUT", "%d", qe.pos);
   record_abandoned(&qe);
   reason = QUEUE_TIMEOUT;
   res = 0;
   break;
   }

In both versions, the variable 'go_on' starts off set to 0, and only gets
set if you pass the 'n' option to Queue().  The manner in which it gets set
differs between 1.4.8 and 1.4.9, but it is only when you pass the 'n'
option, so it shouldn't matter.  In my configuration, go_on should always be
zero.

The logic check around go_on is what's worrying me.  In 1.4.8, go_on had one
of two values - 0 or 1.  If you never passed 'n' to Queue(), it was always
0, so the block of code that takes you back to the dialplan on timeout can
never be executed.

In 1.4.9, if qe.parent->membercount is zero and you didn't pass the 'n'
switch, then you'll exit the queue as if you had timed out, even though you
never passed the 'n' option.  I haven't gone through the entire code of
app_queue to see exactly how membercount gets manipulated, but it seems from
my log that these exitwithtimeouts events seem to occur right after an agent
has let their phone ring without picking it up (see the "nobody picked up in
2ms" message in my example above).

Is it possible for qe.parent->membercount to be set to zero in a queue where
all agents but one are on the phone and that one remaining agent lets their
phone ring without answering it?

--
j.
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Re: [asterisk-users] Attaching VoiceMails on E-Mails

[GNUbie]

Hello Marco,

On 7/27/07, Marco Mouta <[EMAIL PROTECTED]> wrote:


hi,

The 
VoiceMailapplication
 uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use Sendmail,
Postfix , Exim  or any other
MTA .
It is recommended to use the default one that comes with your distribution.



If shall I say I'll use Exim4 here, what do I need to do then?

I would say if you just create your "own" sendmail.sh and place it

/usr/sbin/sendmail, asterisk will execute it by default, do not forget to
give permissions for asterisk user to execute it.



If I'll create this script, what will be its contents then?  Currently, the
/usr/sbin/sendmail is a symbolic link to /usr/sbin/exim4 for your
information.

Please advice.

GNUbie
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Re: [asterisk-users] Attaching VoiceMails on E-Mails

[GNUbie]

Hello Tzafrir,

On 7/27/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:



You need to have a package that provides /usr/sbin/sendmail . While you
can get away with using nullmailer or ssmtp (that don't spool mail
locally), I would recommend you to install postfix or exim, so a
temporary problem won't cause the message to get lost on the way.



Oh, I mean, I have Exim installed here but it's not running by default.
Should I run it anyway?

Please advice.

Thank you once again.

GNUbie
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Re: [asterisk-users] Attaching VoiceMails on E-Mails

[Marco Mouta]

hi,

The 
VoiceMailapplication
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
Sendmail,
Postfix , Exim  or any
other MTA.
It is recommended to use the default one that comes with your distribution.

I would say if you just create your "own" sendmail.sh and place it
/usr/sbin/sendmail, asterisk will execute it by default, do not forget to
give permissions for asterisk user to execute it.


On 7/27/07, GNUbie <[EMAIL PROTECTED]> wrote:


Hello all,

I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to
send the voicemails as attachment to e-mails and delete the voicemails from
my PBX once it has been sent.  But, I don't have a running MTA here even on
the PBX itself.  I just want to send the e-mails to my GMail account from my
PBX.  Can I just use the mail or mailx command to send the e-mail and attach
the voicemail message?

Below are snippets of my voicemail.conf and extensions.conf configuration
files.  Please advice whatever you think I need to change with my current
configurations.

Thank you in advance.

GNUbie

- - - < s n i p > - - -

# cat /etc/asterisk/voicemail.conf

[general]
format=wav49
[EMAIL PROTECTED] ; bogus e-mail address
attach=yes
delete=yes
maxmsg=50
maxmessage=180
minmessage=5
maxgreet=60
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
fromstring=The PBX
usedirectory=yes
emaildateformat=%A, %B %d, %Y at %r
sendvoicemail=yes
emailbody=Hi, ${VM_NAME}!\n\nYou have a new voicemail message from
${VM_CALLERID} attached to this e-mail message.\n\nHave a nice day!\n\nThe
PBX
mailcmd=/usr/bin/exim -t ; not sure about this line

[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp

[default]
101 => 11011,GNUbie,[EMAIL PROTECTED]


# grep 10 /etc/asterisk/extensions.conf

exten => 101,1,Dial(Zap/1,20,rt)
exten => 101,2,VoiceMail(101,u)
exten => 100,1,VoiceMailMain(${CALLERID(num)},s)

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Re: [asterisk-users] Attaching VoiceMails on E-Mails

[Tzafrir Cohen]

On Fri, Jul 27, 2007 at 08:55:01AM +0800, GNUbie wrote:
> Hello all,
> 
> I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to
> send the voicemails as attachment to e-mails and delete the voicemails from
> my PBX once it has been sent.  But, I don't have a running MTA here even on
> the PBX itself.  I just want to send the e-mails to my GMail account from my
> PBX.  Can I just use the mail or mailx command to send the e-mail and attach
> the voicemail message?

You need to have a package that provides /usr/sbin/sendmail . While you
can get away with using nullmailer or ssmtp (that don't spool mail
locally), I would recommend you to install postfix or exim, so a
temporary problem won't cause the message to get lost on the way.

> 
> Below are snippets of my voicemail.conf and extensions.conf configuration

Generally the default voicemail.conf should do.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Attaching VoiceMails on E-Mails

[GNUbie]

Hello all,

I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to
send the voicemails as attachment to e-mails and delete the voicemails from
my PBX once it has been sent.  But, I don't have a running MTA here even on
the PBX itself.  I just want to send the e-mails to my GMail account from my
PBX.  Can I just use the mail or mailx command to send the e-mail and attach
the voicemail message?

Below are snippets of my voicemail.conf and extensions.conf configuration
files.  Please advice whatever you think I need to change with my current
configurations.

Thank you in advance.

GNUbie

- - - < s n i p > - - -

# cat /etc/asterisk/voicemail.conf

[general]
format=wav49
[EMAIL PROTECTED] ; bogus e-mail address
attach=yes
delete=yes
maxmsg=50
maxmessage=180
minmessage=5
maxgreet=60
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
fromstring=The PBX
usedirectory=yes
emaildateformat=%A, %B %d, %Y at %r
sendvoicemail=yes
emailbody=Hi, ${VM_NAME}!\n\nYou have a new voicemail message from
${VM_CALLERID} attached to this e-mail message.\n\nHave a nice day!\n\nThe
PBX
mailcmd=/usr/bin/exim -t ; not sure about this line

[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp

[default]
101 => 11011,GNUbie,[EMAIL PROTECTED]


# grep 10 /etc/asterisk/extensions.conf

exten => 101,1,Dial(Zap/1,20,rt)
exten => 101,2,VoiceMail(101,u)
exten => 100,1,VoiceMailMain(${CALLERID(num)},s)
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Re: [asterisk-users] Newbie Advice on Asterisk and Linux

[GNUbie]

Hello Mark,

On 7/27/07, Mark Burrows <[EMAIL PROTECTED]> wrote:




Can someone suggest a starting point on learning Linux?



First of all, welcome to the community!  =)

I may consider myself as an experienced systems/network administrator but
with Asterisk and telephony, I am still newbie to it.  For me, Asterisk and
telephony in general is a totally different world.  If I may suggest to you,
try to check Trixbox  because it's a pretty easy to
setup VoIP software appliance in one distribution.  Although I must admit
that I didn't tried installing it and I installed Asterisk instead
"the-hard-way" TM on my Debian Etch  so that I would
learn (hopefully) this technology.  Anyway, my setup here is only on my home
PBX so nothing to worry.

If you would like to learn GNU/Linux systems and/or network administration,
you will have to choose a "distribution" to start with.  You can choose
GNU/Linux distributions at the DistroWatch website <
http://www.distrowatch.com/>.  I use Debian GNU/Linux Etch here as my OS.
You can also find useful information on GNU/Linux and F/OSS in general at
The Linux Documentation Project website .  Your
distribution of choice might have a good documentation specially the systems
and network administration.

Good luck!

GNUbie
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[asterisk-users] ISDN: Problems starting off

[Bertram Scharpf]

Hi,

the first thing I did with Asterisk is listening to
`demo-congrats' by Xlite on the same machine. This works
perfectly. The config files are those shipped with the
package.

Now I want to listen to it over ISDN/Capi but I don't
succeed.

My `capi.conf' is like show in many tutorial on the web. In
`extensions.conf' I just added the following lines:

  [capi-in]
  exten => 9876543,1,Goto(demo,1000,1)

where 9876543 is my MSN without the area prefix. `demo' is
the context that plays `demo-congrats'.

The debug output I yield ends with

(after a pause)

  DISCONNECT_IND ID=001 #0x0027 LEN=0014
Controller/PLCI/NCCI= 0x101
Reason  = 0x0

  DISCONNECT_RESP ID=001 #0x0027 LEN=0012
Controller/PLCI/NCCI= 0x101

  -- ISDN1#02: DISCONNECT_IND on incoming without pbx, doing hangup.
 > CAPI/ISDN1/9876543-2: set channel task to 1
== ISDN1#02: CAPI Hangingup for PLCI=0x101 in state 4
== ISDN1#02: Interface cleanup PLCI=0x101
 > CAPI devicestate requested for ISDN1/9876543


Seems that the MSN or even `capi-in' cannot be found at all.

Could anyone give me a hint what is going wrong here or at
least what I have to diagnose next?

Thanks in advance.

Bertram


-- 
Bertram Scharpf
Stuttgart, Deutschland/Germany
http://www.bertram-scharpf.de

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Re: [asterisk-users] Ring forever

[Eric \"ManxPower\" Wieling]

FERNANDO VILLARROEL wrote:
> Hello list, i need help.
> 
> My problem is that when I want to call (using ISDN
> phone or internal SIP client) via the Sip provider a
> real phone number (ISDN phone or internal SIP
> 
> Asterisk >> SIP ), I get a ring tone. When I
> now decide to hang up (e.g. if 
> 
> nobody answers), the called telephone continues to
> ring almost forever.

> -- Call on SIP/nyphone-081a7768 left from hold
> -- SIP/nyphone-081a7768 answered
> SIP/2563105-0819cf80
> -- Packet2Packet bridging SIP/2563105-0819cf80 and
>  SIP/nyphone-081a7768

nyphone is answering your call and then dialing out to the destination.

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Re: [asterisk-users] Grandstream RTP keepalive packets causing Asteriskwarning

[Steve Langstaff]

Grab a network trace (with e.g. Wireshark) and look at the payload type
and lengths of the RTP keepalive messages - if you post this information
to the list I'm sure someone will comment on what's happening.

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Drew Gibson
> Sent: 26 July 2007 19:23
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Grandstream RTP keepalive packets 
> causing Asteriskwarning
> 
> Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an 
> issue where the phone did not send rtp keepalives when on 
> mute (resulting in disconnect from tech support hold and concalls)
> 
> A side effect seems to be that Asterisk pops the following 
> warning on the console...
> 
> Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP 
> Read too short
> 
> Grandstream say they are not sure what it is but "it should 
> not affect anything".
> 
> In other words, "Don't worry, be happy!".
> 
> Any thoughts/experience on this?
> 
> regards,
> 
> Drew
> 
> --
> Drew Gibson
> 
> Systems Administrator
> OANDA Corporation
> 416-593-6767 x322
> www.oanda.com
> 
> 
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[asterisk-users] (Update) Digium FTP server will be replaced with HTTP server

[Kevin P. Fleming]

Some time in the next two weeks, Digium will be shutting down our FTP
server, located at ftp.digium.com, and begin using only the existing
HTTP server on the same system instead.

We have decided to only offer our public downloads over the HTTP
protocol, not the FTP protocol, primarily for reasons related to our
marketing department :-)

The site has a new name, downloads.digium.com, and will also respond to
the old name ftp.digium.com, but will no longer respond to requests made
via the FTP protocol; only the HTTP protocol will be supported. There
should be no other user-visible changes when this change is made to the
server. We will begin using the downloads.digium.com name in all our
announcements, security advisories, documentation and other places
immediately, and encourage the community to do so as well.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)



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Re: [asterisk-users] SetCallerPres and Cisco PRI

[Alex Balashov]

PS.  Check this out:

http://bugs.digium.com/print_bug_page.php?bug_id=2471

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] SetCallerPres and Cisco PRI

[Alex Balashov]

Hi Peder,

You tried blanking the caller ID field and it didn't work?

   i.e., exten => ...,n,Set(CALLERID(all)=)

It worked for me, although my media gateway was not a Cisco one.

Whether SetCallerPres() will work depends entirely on what it 
accomplishes.  Does it just alter the cosmetic From: line, and
does the Cisco gateway take stock in that?  Or does it tack on
the draft privacy headers (Remote-Party-ID) and set privacy to
on/full?

My gut feeling is that SetCallerPres() applies to calls placed
directly out of a PRI interface, not SIP, because "presentation"
is a term typically applied to caller ID in an ISDN, not a SIP
context.

It is hard to tell whether this intuition is correct because
SetCallerPres() is fundamentally implemented in apps/app_setcallerid.c
which calls ast_set_callerid() in main/channel.c and appears to apply
to a variety of channel types variously.

If this doesn't work, try this:

   
http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header

A Cisco MGW should support that just fine.

Good luck,

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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[asterisk-users] SetCallerPres and Cisco PRI

[Peder @ NetworkOblivion]

Does anybody know if SetCallerPres works on calls via SIP through a 
Cisco gateway?  We are trying to mark outbound calls as anonymous and we 
set it to prohib, but calls still show outbound callerid.  We are SIP 
from * to the Cisco gateway and then PRI outbound.  If we strip the 
callerid num, then the first number on the PRI gets added as teh 
callerid, so we can't do that.  We need to make it anonymous so that it 
shows as unknown on the other end.

Peder


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[asterisk-users] Ring forever

[FERNANDO VILLARROEL]

Hello list, i need help.

My problem is that when I want to call (using ISDN
phone or internal SIP client) via the Sip provider a
real phone number (ISDN phone or internal SIP

Asterisk >> SIP ), I get a ring tone. When I
now decide to hang up (e.g. if 

nobody answers), the called telephone continues to
ring almost forever.

the sip.conf:

[2563105] 
accountcode = 2563105
username = 2563105
secret = 135
callerid = 412563105
context = test
canreinvite = no
dtmfmode = rfc2833
host = dynamic
insecure = very
language = es
nat = yes
qualify = yes
type = friend
disallow=all
allow=g729

[nyphone]
accountcode=nyphone
canreinvite=no
reinvite=yes
dtmfmode=rfc2833
host=72.55.143.XXX
insecure=very
language=es
nat=no
qualify=no
type=peer
disallow=all
allow=g729


My extensions.conf

exten => _00X.,1,dial(sip/${EXTEN:[EMAIL PROTECTED],45)
exten => _00X.,2,hangup


Nyphone is my provider for everyone calls
international.

Fernando Villarroel Noriel.
Chillan
Chile

Sorry my English.

This is log:
   
SIP Debugging Enabled for IP: 72.55.143.XXX:5060
-- Executing [EMAIL PROTECTED]:1]
Dial("SIP/2563105-0819cf80",
 "sip/[EMAIL PROTECTED]|45") in new stack
Audio is at 164.77.171.XXX port 16548
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 72.55.143.XXX:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport
From: "2563105"
;tag=as726ac50a
To: 
Contact: 
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 20 Jul 2007 03:38:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 2475 2475 IN IP4 164.77.171.XXX
s=session
c=IN IP4 164.77.171.XXX
t=0 0
m=audio 16548 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called [EMAIL PROTECTED]
vaca*CLI> 
<--- SIP read from 72.55.143.XXX:5060 --->
SIP/2.0 407 Proxy Authentication Required
CSeq: 102 INVITE
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport
From: "2563105"
;tag=as726ac50a
Call-ID:
[EMAIL PROTECTED]
To: 
Contact: 
Proxy-Authenticate: DIGEST realm="VoipSwitch",
 nonce="118490324119231120007472128429"
Content-Length: 0


<->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 72.55.143.XXX:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport
From: "2563105"
;tag=as726ac50a
To: 
Contact: 
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 164.77.171.XXX port 16548
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 72.55.143.XXX:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport
From: "2563105"
;tag=as726ac50a
To: 
Contact: 
Call-ID:
[EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="test770",
realm="VoipSwitch",
 algorithm=MD5, uri="sip:[EMAIL PROTECTED]",
 nonce="118490324119231120007472128429",
response="413be923621811a639c3b0e83d3a2e74", opaque=""
Date: Fri, 20 Jul 2007 03:38:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 2475 2476 IN IP4 164.77.171.XXX
s=session
c=IN IP4 164.77.171.XXX
t=0 0
m=audio 16548 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
vaca*CLI> 
<--- SIP read from 72.55.143.XXX:5060 --->
SIP/2.0 200 OK
CSeq: 103 INVITE
Via: SIP/2.0/UDP
164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport
From: "2563105"
;tag=as726ac50a
Call-ID:
[EMAIL PROTECTED]
To:
;tag=1907470723212675853288937
Contact: 
Content-Type: application/sdp
Content-Length: 215

v=0
o=VoipSwitch 9936 9936 IN IP4 72.55.143.XXX
s=VoipSIP
i=Audio Session
c=IN IP4 72.55.143.XXX
t=0 0
m=audio 8936 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<->
--- (9 headers 10 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 72.55.143.XXX:8936
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100
(g729)/video=0x0
 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1
(telephone-event), peer - 0x1
 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 72.55.143.XXX:8936
list_route: hop:

set_destination: Parsing
 for
 address/port to send to
set_destination: set destination to 72.55.143.XXX,
port 5060
Transmitting (no NAT) to 72.55.143.XXX:5060:
ACK sip:

Re: [asterisk-users] vm-duration announcement missing?

[Kevin P. Fleming]

James FitzGibbon wrote:

> Looking back at my logs, there are semi-regular instances of this error
> message.  In a default setup, it's only used if the message is more than
> 2 minutes long, which I guess most of my user's VMs aren't.

This is my fault; we have a pending list of sounds to be recorded and
included in the core-sounds package and I've neglected it. I'll put it
on my to-do list for tomorrow right now... Sorry :-(

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] Newbie Advice on Asterisk and Linux

[john beaman]

Mark,
  Welcome to the club.  Learning Linux can be a daunting task.  After working 
with it for the last decade, I am still learning.  My best recommendation is to 
"play" with it on a test box, and post questions to a related community forum 
if you get stuck on something.  If you are looking for something more "intense" 
and less time-consuming, check your local colleges.  The colleges in my area 
offer several classes on Linux as part of a degree in Network Administration.

HTH,


John Beaman
Telecom Specialist
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331

>>> [EMAIL PROTECTED] 7/26/2007 3:08:36 PM >>>
HI All,

 

I'm new to Asterisk and also to Linux.  I have a large IVR project that I'm
about to embark on.  I'm new to programming; new to Linux and new to
Asterisk.  I think I'm about to climb a steep learning curve.  I have an
existing IVR which is getting on for nine years old and is no longer
supported by my vendor.  I intend to replicate the system almost as is and
then add additional features and functions.  

 

I have been looking for a developer to put together my project and while
doing so have done lots of research and spoken to many people.  The people
who seem to understand my needs have recommended Asterisk.  For the last
couple of days I've been trying to look into Asterisk and learn as much as I
can; this has got me excited, motivated and a little confused. Asterisk
sounds like a great project and a great community.  I think I have as much
of an overview as I can.  Now I need to set up a Linux system and get
Asterisk running on it. 

 

I've started to read the book Asterisk: The Future Of Telephony and would
like to now setup up a hobby computer to do some hands on learning.  The
book covers Red Hat Linux so I thought I'd look for a 'Red Had for Dummies'
book.  Even that got confusing. There's Linux Fedora, Enterprise Linux 4 and
others.

 

Can someone suggest a starting point on learning Linux?

 

Thanks in advance,

 

 

Mark

 

 

 


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Checked by AVG Free Edition. 
Version: 7.5.476 / Virus Database: 269.10.20/919 - Release Date: 26/07/2007
9:56 AM
 
-

This email transmission and any documents, files or previous

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confidential or legally privileged. If you are not the intended

recipient, you are hereby notified that any disclosure, copying,

printing, distributing or use of this transmission is strictly

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please immediately notify the sender by telephone or return

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[asterisk-users] vm-duration announcement missing?

[James FitzGibbon]

I just saw this on my console:

[Jul 26 11:36:30] WARNING[8667] file.c: File vm-duration does not exist in
any format
[Jul 26 11:36:30] WARNING[8667] file.c: Unable to open vm-duration (format
0x4 (ulaw)): No such file or directory

Thinking I might have lost a file during a fsck or something, I checked -
sure enough, there's no file "vm-duration" in any format.  I downloaded the
current (as of June 14th) core and extra sounds, but it's not in there
either.

1.2.x didn't use this file, but app_voicemail contains reference to it in
1.4.x - as far back as 1.4.0:

   if ((!res) && (durationm >= minduration)) {
   res = wait_file2(chan, vms, "vm-duration");

[snip stuff about polish syntax]

   res = ast_say_number(chan, durationm, AST_DIGIT_ANY,
chan->language, NULL);
   res = wait_file2(chan, vms, "vm-minutes");
   }

Does anyone know where this file can be fetched from, or at least what it's
supposed to say?

Looking back at my logs, there are semi-regular instances of this error
message.  In a default setup, it's only used if the message is more than 2
minutes long, which I guess most of my user's VMs aren't.

Thanks

--
j.
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Re: [asterisk-users] Newbie Advice on Asterisk and Linux

[Jonn Taylor]

Mark Burrows wrote:
> HI All,
> 
>  
> 
> I’m new to Asterisk and also to Linux.  I have a large IVR project that 
> I’m about to embark on.  I’m new to programming; new to Linux and new to 
> Asterisk.  I think I’m about to climb a steep learning curve.  I have an 
> existing IVR which is getting on for nine years old and is no longer 
> supported by my vendor.  I intend to replicate the system almost as is 
> and then add additional features and functions. 
> 
>  
> 
> I have been looking for a developer to put together my project and while 
> doing so have done lots of research and spoken to many people.  The 
> people who seem to understand my needs have recommended Asterisk.  For 
> the last couple of days I’ve been trying to look into Asterisk and learn 
> as much as I can; this has got me excited, motivated and a little 
> confused. Asterisk sounds like a great project and a great community.  I 
> think I have as much of an overview as I can.  Now I need to set up a 
> Linux system and get Asterisk running on it.
> 
>  
> 
> I’ve started to read the book Asterisk: The Future Of Telephony and 
> would like to now setup up a hobby computer to do some hands on 
> learning.  The book covers Red Hat Linux so I thought I’d look for a 
> ‘Red Had for Dummies’ book.  Even that got confusing. There’s Linux 
> Fedora, Enterprise Linux 4 and others.
> 
>  
> 
> Can someone suggest a starting point on learning Linux?
> 
>  
> 
> Thanks in advance,
> 
>  
> 
>  
> 
> Mark
> 
>  
> 
>  
> 
>  
> 
> 
> No virus found in this outgoing message.
> Checked by AVG Free Edition.
> Version: 7.5.476 / Virus Database: 269.10.20/919 - Release Date: 
> 26/07/2007 9:56 AM
> 
> 
> 
> 
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
http://www.trixbox.org

This is "one" of the many standard configs for aterisk. This uses CentOS 
4, Asterisk 1.2, FreePBX 2.2. You can setup a fully working system in 
about 30 min. Need help you can email me off list.

Jonn Taylor

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[asterisk-users] Newbie Advice on Asterisk and Linux

[Mark Burrows]

HI All,

 

I’m new to Asterisk and also to Linux.  I have a large IVR project that I’m
about to embark on.  I’m new to programming; new to Linux and new to
Asterisk.  I think I’m about to climb a steep learning curve.  I have an
existing IVR which is getting on for nine years old and is no longer
supported by my vendor.  I intend to replicate the system almost as is and
then add additional features and functions.  

 

I have been looking for a developer to put together my project and while
doing so have done lots of research and spoken to many people.  The people
who seem to understand my needs have recommended Asterisk.  For the last
couple of days I’ve been trying to look into Asterisk and learn as much as I
can; this has got me excited, motivated and a little confused. Asterisk
sounds like a great project and a great community.  I think I have as much
of an overview as I can.  Now I need to set up a Linux system and get
Asterisk running on it. 

 

I’ve started to read the book Asterisk: The Future Of Telephony and would
like to now setup up a hobby computer to do some hands on learning.  The
book covers Red Hat Linux so I thought I’d look for a ‘Red Had for Dummies’
book.  Even that got confusing. There’s Linux Fedora, Enterprise Linux 4 and
others.

 

Can someone suggest a starting point on learning Linux?

 

Thanks in advance,

 

 

Mark

 

 

 


No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.476 / Virus Database: 269.10.20/919 - Release Date: 26/07/2007
9:56 AM
 
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Re: [asterisk-users] Queue Stats

[Matt King]

Hello Jay,

Sounds like quite a complicated set up.  Most queue statistics packages 
will break your callers down depending on which queue they were actually 
answered in (or hung up on).

If you want your stats listed as if the callers were in a single queue, 
you can sign up for a FREE OrderlyStats account at 
http://www.orderlyq.com/orderlystats.html - once you're all done, let us 
know and we'll show you how OrderlyStats can show these calls as if your 
three queues were just one.

Hope this helps,

Matt.

Jay wrote:

Greetings, list!

My boss would like some statistics on how many calls are answered out of 
specific queues during a given time period, and I'm not sure how exactly 
to obtain those stats.  Here's how our queue system works.

1) Call comes in and enters our 'ring' queue where the phones ring for 
15 seconds (caller hears the standard ring tone).

2) After 15 seconds, the caller falls into our 'music on hold' queue, a 
message is played and the caller hears our music on hold while the 
phones are rung again.

3) After 30 seconds, if the caller is still in our 'moh' queue, they 
drop out of queue and immediately re-enter the 'moh' queue again until 
the call is answered or the caller hangs up.

How can I find out how many calls are answered out of each queue during 
certain times (1st shift, 2nd shift, etc...)?  Also, I'm curious how I 
can track how many times a call repeats the 'moh' queue.

Thanks 





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Re: [asterisk-users] Display IE

[Damon Estep]

Is there more than one display IE in the original ISDN setup message coming 
from the Telco?

 

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio
Sent: Thursday, July 26, 2007 5:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Display IE

 

Hi!

Thank you all for the info!

But I think I haven't explained my scenario well enough. 

I am not relaying the calls to SIP.

What happens is the following (the scenario is: a call started from an ISDN E1 
PBX, through the asterisk, to the PSTN, and was answered by the remote party):

1. A CONNECT comes from the PSTN containing a Display IE (which has info sent 
by the telco that is used for billing purposes by the PBX) into span 1 of the 
asterisk. (The telco is emulating an old billing procedure based on an impulse 
count. This was commonly used in analog lines.)

2. The asterisk relays the call to span 2 (which is connected to the PBX).

3. The CONNECT that is sent from span 2 to the PBX does not have the Display 
IE. The asterisk strips this IE from the CONNECT message.


This is my problem. Is there a way that i can force the asterisk not to strip 
the Display IE?

Thanks and best regards to all,

Óscar Patrício


Anthony Francis wrote: 

Damon Estep wrote:
  

Try putting a 1 second wait as step 1 in the dialplan, the SIP invite 
is probably being send before the display IE arrives. The display IE is used 
for CNAM delivery, and should not exceed 15 characters.
 
It is very common to put a message in the display IE that indicates 
that the CNAM info will be sent in a subsequent Facility IE, and for that you 
must wait 1 second.
 
If the ISDN setup actually contains information in the display IE, and 
that is not being captured as the CNAM (callerid(name)) you might need to 
capture he ISDN messaging to debug it, the telco can usually provide such a 
trace.
 
I would bet that the display IE contains a information following 
message, and what you really want is in the facility IE that follows. Very 
common, as is the Wait(1) workaround.
 
 
 
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar 
Patricio
Sent: Wednesday, July 25, 2007 6:55 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Display IE
 
Hi!
 
I have an Asterisk Box that has 2 E1 connections: one to the PSTN and
one to a PBX. It is acting as a telephony gateway. I have a problem: the
PSTN sends information in the Display IE (in setup, information ,
etc.messages) that the PBX needs por internal processing.
 
The asterisk does not relay the ISDN frames coming from the PSTN. It
regenerates them, and when it does, it ommits the Display IE.
 
Is there a way that i can force the asterisk, not to ignore this IE in
the ISDN messages?
 
If anyone can shed some light on this issue, i would be very grateful.
 
Thanks in advance,
 
Best regards,
 
Óscar Patrício
 
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Or trace it yourself with:
pri intense debug span 1
 
Make sure you change the 1 to whatever span these calls are coming across.
 
Anthony
 
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Re: [asterisk-users] Asterisk Vm functionality question

[Eric \"ManxPower\" Wieling]

Andrew Kohlsmith wrote:
> On Thursday 26 July 2007 12:51:06 pm Eric Chamberlain wrote:
>> Could you elaborate on how you configure the MWI of the mobile device to
>> use asterisk voicemail?
> 
> yes, please explain.  SMSing the phone doesn't light MWI, unless you get 
> access to the "raw" SMSC, as all the email gateways just mangle the message.

I suspect people that light the MWI on their Cell phone do not live in 
the USA.

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Re: [asterisk-users] X100P pass through questions

[Eric \"ManxPower\" Wieling]

That is correct.  The X100P only detects voltage drop, not polarity 
reversal.

Walter Willis wrote:
> i am have x100p and not work fine, no detect polarity, and much problems
> with asterisk 1.2 to up.
> :S
> 
> 
> On 7/25/07, Mike Wright <[EMAIL PROTECTED]> wrote:
>>
>> John Novack wrote:
>> >
>> > Mike Wright wrote:
>> >
>> >>Just purchased a Motorola Wildcard X100P ...
>> >>but the button pressed generates no tone; on button release dialtone
>> returns.
>> >>
>> >
>> > Sure sounds like polarity reversal.
>> >
>> Indeed it was.  Punch block in the basement had tip and ring reversed.
>> Probably been that way for thirty years.  Amazing that the dsl installer
>> that replaced the inside wiring didn't catch it.
>>
>> For years I've had intermittent problems using IVR.  Some phones would
>> work, some would not.  Now I know why.
>>
>> Thanks to John Novack and Eric "ManxPower" Wieling for your help.

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Re: [asterisk-users] IAX2 INBAND DTMF?

[Steven]

Yes, it is a Blue Digium IAXy.

It is on my local LAN , so the Linksys SIP is working fine.

It was just a surprising discovery since Digium's owner defined IAX2, specified 
that there can be no in band DTMF and then Disgium 
left this out of the IAXy.

I believe that they assumed that it would only be used as a station phone.

-- 
-- 
Steven

http://www.glimasoutheast.org



"Jared Smith" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> On Wed, 2007-07-25 at 17:16 -0400, Steven wrote:
>> My biggest issue with this is that the Iaxys will not generate DTMF tones 
>> onto the analog side..
>
> Which type of IAXy do you have? I remember having a problem with this
> over a year ago with one of the older IAXy boxes (the blue ones), but it
> seemed to work fine when I swapped it for one of the grey ones.  It's
> been long enough now that I don't remember if I had to do anything else
> special to get it to work.  But I do know for a fact that this
> elementary school has been using my IAXy to drive their paging system
> for over a year.  (I'm not sure that's much help, but maybe it'll spark
> someone else's memory.)
>
> -- 
> Jared Smith
> Community Relations Manager
> Digium, Inc.
>
>
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Re: [asterisk-users] Lines Not being Hung UP Major

[Jaswinder Singh]

Btw are the phones behind NAT ?? Also you can try some softphone and make
sure that this problem is caused by snom phones or some other factors ..

On 25/07/07, Michael J. Liberatore <[EMAIL PROTECTED]>
wrote:


I thought it was the fios service but now I realize it's the snom 360!
It doesn't hang up random outgoing calls.  It seems like it only happens
on outbound calls from phones that have been updated to 6.5.12 or
6.5.10.  It didn't happen before, but I don't remember what version
firmware it was before, maybe 6.2.3 or so.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron Arts
Sent: Monday, July 16, 2007 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Lines Not being Hung UP Major

Do your SNOM phones sometimes use answer-after:0, and do they have
keyboard LEDs subscribed to their own extensions?
Do those people hangup calls by puttig down the handset instead of
pressing the X key?

We are seeing hanging channels in this particular case.

Ron


Michael J. Liberatore wrote:
> Hi all, i am having a major asterisk problem.  I think it started
> around
> 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360.  basically
> we start getting busy signals, all our 4 line hunt group is busy, i
> then check the channels and there are open calls that were hung up
long ago.
> i thought it was a zap problem but then i saw the same problem with
> iax2 calls.  its becoming a huge issue because if i dont reboot
> asterisk several times a day, all our lines get filled up with dead
> calls.  I am now running 1.2.21.1 asterisk with the same problem.
Please help.
>
> Mike
>
>
> This E-mail, including any attachments, may be intended solely for the

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> information in this e-mail identifying a former, present, or potential

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[asterisk-users] missing digits on PRI

[Jerry Geis]

I seem to be missing digits with a PRI.

I added dtmf logging in logger.conf

This does not happen a-lot but it does happen a number of times over the 
day.

I have watched a few times for calls coming in and the logger
only showed me 09 instead of 209.

I contacted my provider they checked it out and said they see no 
impedance issues
and everything looks fine on their end.

I have installed a handful of T1s at other sites and no issues.
The card is a Te205p. I am running 1.4.7.1 asterisk and 1.4.3 zaptel.

What might I tweek to get 100% on DTMF.

Jerry

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[asterisk-users] Grandstream RTP keepalive packets causing Asterisk warning

[Drew Gibson]

Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where 
the phone did not send rtp keepalives when on mute (resulting in 
disconnect from tech support hold and concalls)

A side effect seems to be that Asterisk pops the following warning on 
the console...

Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short

Grandstream say they are not sure what it is but "it should not affect 
anything".

In other words, "Don't worry, be happy!".

Any thoughts/experience on this?

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com


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Re: [asterisk-users] Dialtone when automatically picking up.

[Peter Hessler]

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: 
:> ;; dialtone in the background isn't there any more
:> ;; dialed '305'
:> ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent... Asterisk is playing
:dialtone *on top* of the background message the first time around?  That
:truly is bizarre.  I have no idea what would cause that.

Correct.

I did some more testing, and found that there is some crazy 
?cross-talk? going on.  If I dial my mobile from that line, two 
caller-id numbers show up, the one that should, and someone else's 
line.  I'm now thinking this is a problem with the installation of the 
phone lines, so I'm now hunting down the installer to have him fix it.

On the bright side, when I was tracking this down I learned Asterisk 
fairly well. ;)



-- 
Have you noticed that all you need to grow healthy, vigorous grass is a
crack in your sidewalk?

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[asterisk-users] Digium FTP server will be replaced with HTTP server

[Kevin P. Fleming]

Some time in the next two weeks, Digium will be shutting down our FTP
server, located at ftp.digium.com, and begin using only the existing
HTTP server on the same system instead.

We have decided to only offer our public downloads over the HTTP
protocol, not the FTP protocol, primarily for reasons related to our
marketing department :-)

The site will still be called ftp.digium.com, but will no longer respond
to requests made via the FTP protocol; only the HTTP protocol will be
supported. There should be no other user-visible changes when this
change is made to the server.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)


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Re: [asterisk-users] Asterisk Vm functionality question

[Andrew Kohlsmith]

On Thursday 26 July 2007 12:51:06 pm Eric Chamberlain wrote:
> Could you elaborate on how you configure the MWI of the mobile device to
> use asterisk voicemail?

yes, please explain.  SMSing the phone doesn't light MWI, unless you get 
access to the "raw" SMSC, as all the email gateways just mangle the message.

-A.

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Re: [asterisk-users] Queue stats

[Jay Moore]

> Hopefully that helps clarify things!

It does immensely.  Thanks a ton!

Jay

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Re: [asterisk-users] Asterisk Vm functionality question

[Eric Chamberlain]

Andrew,



Could you elaborate on how you configure the MWI of the mobile device to
use asterisk voicemail?



--

Eric Chamberlain, CISSP

Chief Technical Officer

Voxilla - http://voxilla.com/



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Re: [asterisk-users] IAX connections broken

[Dave Bour]

Are sites listed by IP or DN.   If IP, dumb question but did it change?  If DN, 
can you resolve it from the respective boxea?

Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]

For those who just want it to work...
Giving you complete IT peace of mind. 

(Sent via Blackberry - hence message may be shorter than my usual verbose 
responses)
PIN 4cc364db (as of March 24, 2007)  

- Original Message -
From: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED] <[EMAIL PROTECTED]>; Asterisk Users Mailing List - 
Non-Commercial Discussion 
Sent: Thu Jul 26 10:17:23 2007
Subject: Re: [asterisk-users] IAX connections broken

Not likely. 
#1, I have a public IP on that firewall. 
#2. If I block 4569 at our firewall, then it goes from closed to
stealth. If I forward the port, it goes from stealth to closed.

The iaxping tool (http://www.bpvn.com/asterisk/iaxping.zip) has no
problems pinging the box from the lan, and our test machine can make an
IAX connection to the box. From outside the network, however, it times
out.

It has to be a NAT problem, but forwarding doesn't appear to be working.

Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Baji
Panchumarti
Sent: Thursday, July 26, 2007 10:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX connections broken

what if your internet provider is blocking inbound 4569 ?

--

  On 7/26/07, Michael Munger wrote:

> Dear All:
>
> I have several boxes that up and running just great, then we changed
> internet equipment due to a lightning strike, now all my inbound IAX
> connections (iax2 show peers) have unknown status. If I log into the
> remote boxes, it says "Request sent."
>
> The authentications haven't changed at all, and all the iax.conf
> settings are correct. It looks like a firewall issue, but we've got
4569
> TCP & UDP forwarded to our Asterisk box. When I use Shields up from
> GRC.com to test the port, it is showing up as "closed" rather than
open,
> which normally means the port is open, but the service is not running,
> yet Asterisk is up and running just fine, and my outbound connections
to
> Voicepulse work fine. I see voicepulse, voicepulse sees me.
>
> There is something I am not seeing here. Any thoughts?
>
> -Michael
>
> ___

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Re: [asterisk-users] Queue stats

[Jared Smith]

On Thu, 2007-07-26 at 11:06 -0500, Jay Moore wrote:
> So here is my question:
> 
> In this format: "1|2|3|4|5|6",
> 1 - ?
> 2 - ?
> 3 - queue in question?
> 4 - agent answering the queue?
> 5 - queue event?
> 6 - queue event info?
> 
> Is that correct?  What are options 1 and 2?  Times of some sort I'm 
> guessing, but I'm not entirely sure.

The first column is the time, in Unix epoch format (number of seconds
since January 1, 1970).  This allows you to tell *when* each event
happened.

The second column is the unique call id of the call in question.  This
tells you *what call* the event happened on.

The third column is the queue name.  

The fourth column is the queue member or agent.

The fifth is the queue event (as described in queuelog.txt).

The sixth (and seventh, eighth, etc.) are the event info.  Each type of
queue event sets different kinds of event info, as described in
queuelog.txt.

Hopefully that helps clarify things!

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Queue stats

[Jay Moore]

Jared Smith wrote:
> On Thu, 2007-07-26 at 09:37 -0500, Jay Moore wrote:
>> My boss would like some statistics on how many calls are answered out of 
>> specific queues during a given time period, and I'm not sure how exactly 
>> to obtain those stats.  
> 
> It sounds like you've got quite the queue setup (although I don't quite
> see why your calls jump out and back into the moh queue).  All the of
> queue statistics you need should be available with careful parsing of
> the queue log (usually located in /var/log/asterisk/queue_log).  You can
> also trigger custom queue log events from the dialplan by calling the
> QueueLog() application.  In your case, you might want to add a custom
> queue log entry every time the caller rejoins the moh queue, saying
> something to the effect of "Caller XYZ has rejoined the moh queue for
> the 10th time" or something like that.
> 
> 

We had some issues with the announcement message not playing reliably. 
My fix was to just have them drop out and re-enter the queue.  It 
doesn't seem to have any adverse effects, but if you have any 
alternative suggestions, I'm more than willing to try them.

I've checked my queue log (38megs, yikes) and looked at the queuelog.txt 
info file for how to parse the lines, but I still have a question.  For 
example, a snippet of my log looks like (line numbers mine):

1) 1185460404|1185460400.334916|queue-ring|NONE|ENTERQUEUE||732
2) 1185460420|1185460400.334916|queue-ring|NONE|EXITWITHTIMEOUT|1
3) 1185460427|1185460400.334916|queue-answer|NONE|ENTERQUEUE||732
4) 1185460448|1185460400.334916|queue-answer|NONE|EXITWITHTIMEOUT|1
5) 1185460454|1185460400.334916|queue-answer|NONE|ENTERQUEUE||732
6) 1185460456|1185460400.334916|queue-answer|SIP/agent3-0a5bc480|CONNECT|2
7)
1185460496|1185460400.334916|queue-answer|SIP/agent3-0a5bc480|COMPLETECALLER|2|40

Here's how I interpret this:

1) Call comes into my ring queue
2) Call exits ring queue due to timeout
3) Call enters answer (moh) queue
4) Call exits answer queue due to timeout
5) Call enters answer queue again
6) Agent 3 picks up the call out of the queue
7) Call ends; caller hangs up

So here is my question:

In this format: "1|2|3|4|5|6",
1 - ?
2 - ?
3 - queue in question?
4 - agent answering the queue?
5 - queue event?
6 - queue event info?

Is that correct?  What are options 1 and 2?  Times of some sort I'm 
guessing, but I'm not entirely sure.

Thanks for your help,
Jay

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Re: [asterisk-users] SunRocket / ALLO / etc special offer

[Matt]

I'll take either

Actually now that I have had a chance to think about what I did (sorry
bad week here).  Yes, I will admit I did patrionize the users list...
sorry if I offended anyone.   I just figured I'd try to help any
SunRocket users out that may not be on the biz list.If you review
my history, you'll see I only post business stuff to the biz list.
This is an exception.

On 7/26/07, Baji Panchumarti <[EMAIL PROTECTED]> wrote:
>   On 7/26/07, Matt Hoppes wrote:
>
> > I would agree... intended to send that to biz, sorry.
>
>  I see that you also sent it to the biz-list.
>
>  And if you fail the lie detector test how about agreeing to a
>  full boycott of your service or at least a M.L.D.P. (mailing
>  list death penalty :-) ?
>
>  --
>
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Re: [asterisk-users] X100P pass through questions

[Walter Willis]

i am have x100p and not work fine, no detect polarity, and much problems
with asterisk 1.2 to up.
:S


On 7/25/07, Mike Wright <[EMAIL PROTECTED]> wrote:


John Novack wrote:
>
> Mike Wright wrote:
>
>>Just purchased a Motorola Wildcard X100P ...
>>but the button pressed generates no tone; on button release dialtone
returns.
>>
>
> Sure sounds like polarity reversal.
>
Indeed it was.  Punch block in the basement had tip and ring reversed.
Probably been that way for thirty years.  Amazing that the dsl installer
that replaced the inside wiring didn't catch it.

For years I've had intermittent problems using IVR.  Some phones would
work, some would not.  Now I know why.

Thanks to John Novack and Eric "ManxPower" Wieling for your help.

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Re: [asterisk-users] Queue stats

[Scott Wolfe]

Jay,
  You could try ASTassistant. It has Queue information at a glance.
http://www.astassistant.com


- Original Message - 
From: "Jay Moore" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Thursday, July 26, 2007 7:37 AM
Subject: [asterisk-users] Queue stats


> Greetings, list!
>
> My boss would like some statistics on how many calls are answered out of
> specific queues during a given time period, and I'm not sure how exactly
> to obtain those stats.  Here's how our queue system works.
>
> 1) Call comes in and enters our 'ring' queue where the phones ring for
> 15 seconds (caller hears the standard ring tone).
>
> 2) After 15 seconds, the caller falls into our 'music on hold' queue, a
> message is played and the caller hears our music on hold while the
> phones are rung again.
>
> 3) After 30 seconds, if the caller is still in our 'moh' queue, they
> drop out of queue and immediately re-enter the 'moh' queue again until
> the call is answered or the caller hangs up.
>
> How can I find out how many calls are answered out of each queue during
> certain times (1st shift, 2nd shift, etc...)?  Also, I'm curious how I
> can track how many times a call repeats the 'moh' queue.
>
> Thanks in advance,
> Jay
>
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Re: [asterisk-users] SunRocket / ALLO / etc special offer

[Baji Panchumarti]

  On 7/26/07, Matt Hoppes wrote:

> I would agree... intended to send that to biz, sorry.

 I see that you also sent it to the biz-list.

 And if you fail the lie detector test how about agreeing to a
 full boycott of your service or at least a M.L.D.P. (mailing
 list death penalty :-) ?

 --

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Re: [asterisk-users] Queue stats

[Matt]

This may not be the best solution for you, but it's the only one I can
speak for.   We use QueueMetrics for our queue information and
reporting.   There is a small cost for it, but it is worth every
penny.

On 7/26/07, Jay Moore <[EMAIL PROTECTED]> wrote:
> Greetings, list!
>
> My boss would like some statistics on how many calls are answered out of
> specific queues during a given time period, and I'm not sure how exactly
> to obtain those stats.  Here's how our queue system works.
>
> 1) Call comes in and enters our 'ring' queue where the phones ring for
> 15 seconds (caller hears the standard ring tone).
>
> 2) After 15 seconds, the caller falls into our 'music on hold' queue, a
> message is played and the caller hears our music on hold while the
> phones are rung again.
>
> 3) After 30 seconds, if the caller is still in our 'moh' queue, they
> drop out of queue and immediately re-enter the 'moh' queue again until
> the call is answered or the caller hangs up.
>
> How can I find out how many calls are answered out of each queue during
> certain times (1st shift, 2nd shift, etc...)?  Also, I'm curious how I
> can track how many times a call repeats the 'moh' queue.
>
> Thanks in advance,
> Jay
>
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Re: [asterisk-users] IAX2 INBAND DTMF?

[Matt]

> mailing list archives is all that is needed to know that. 8-)

We've  never had any issues with L3 and are very happy.

>
> I didn't think that Level3 supported IAX connections.  If you are using
> an ITSP that uses Level3, I would hope the ITSP would be using inband
> DTMF on SIP for their connection to Level3.

L3 doesn't support IAX.. but if you peer with a L3 provider using IAX
you are still affected.

Why would the ITSP need to use inband?  They can use RFC to L3.

>
> In any case, that might be a reason, but I don't know if it is a *good*
> reason. 8-)
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