Re: [asterisk-users] saved.gsm -> Voicemail greeting ??

[Andrew Joakimsen]

If you do VoiceMail(bEXTEN) then all you get is "Please leave your
message after the tone" so  you can do the greeting function  in the
dialplan.

If you do VoiceMail(bEXTEN) then you get " is on the phone.
Please leave..."

(uEXTEN) gets you " is unavailable. Please  leave..."

On 9/15/06, Nick Ellson <[EMAIL PROTECTED]> wrote:

Trying that now... umm, anyone know what condition makes use of just the
"name" in voicemail, is that part of the directory or something?

--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.


On Fri, 15 Sep 2006, Bill Gibbs wrote:

> I assume it will use the files .gsm too?
>
> Bill
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs
> Sent: Friday, September 15, 2006 10:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] saved.gsm -> Voicemail greeting ??
>
> Example for mailbox 100 under context default
>
> /var/spool/asterisk/voicemail/default/100
>
> -rwx-w 1 asterisk asterisk 442604 Aug 22 16:44 busy.wav
> -rwx-w 1 asterisk asterisk  44976 Aug 22 16:44 busy.WAV
> -rwx-w 1 asterisk asterisk  25964 Aug  8 02:17 greet.wav
> -rwx-w 1 asterisk asterisk   2660 Aug  8 02:17 greet.WAV
> drwx-w 2 asterisk asterisk   4096 Sep 13 13:45 INBOX
> drwx-w 2 asterisk asterisk   4096 Sep 13 13:37 tmp
> -rwx-w 1 asterisk asterisk 168044 Aug  9 17:00 unavail.wav
> -rwx-w 1 asterisk asterisk  17090 Aug  9 17:00 unavail.WAV
> drwx-w 2 asterisk asterisk   4096 Sep  1 11:31 Work
>
> Those are the files (wav format) that it expects for the voicemail
> greetings/name announcement.  Greet.wav is the name.
>
> Bill
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Nick
> Ellson
> Sent: Friday, September 15, 2006 10:26 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] saved.gsm -> Voicemail greeting ??
>
>
> Hi John,
>
> Yes, I followed an example that put all my family sound files in
> /var/lib/asterisk/sounds/local, which is also where this file is. Now I
> am
> trying to figure out how to get the unavailable|name|Busy .gsm's I made
> loaded into a mailbox without playing my sounds back into a phone ;)
>
> Nick
>
>
> --
> Nick Ellson
> CCDA, CCNP, CCSP, CCAI,
> MCSE 2000, Security+, Network+
> Network Hobbyist, VFR Private Pilot.
>
>
> On Fri, 15 Sep 2006, John covici wrote:
>
>> Check in /var/spool/asterisk/voicemail/default/  for
>> a particular extension, don't know how you want to differentiate after
>> hours, etc.  Also, you can put files in
>> /var/lib/asterisk/sounds/custom and do with them what you want.
>>
>> on Friday 09/15/2006 Nick Ellson([EMAIL PROTECTED]) wrote
>>>
>>> I seem to have stumped myself on this one. I had my son rattle off
> some
>>> really great sound bytes for his own extension (busy, after hours,
> etc)
>>> and that was easy to set up with the dial plan. Now I have his
> actual VM
>>> greeting in a .gsm  and no idea how to get it into his VM Greeting,
> I am
>>> guessing that these are not stored where the other sounds are, maybe
> in
>>> the database? I looked through the .pdf book, not many helpful hits
> on
>>> google. Help? :)
>>>
>>> Nick
>>>
>>> --
>>> Nick Ellson
>>> CCDA, CCNP, CCSP, CCAI,
>>> MCSE 2000, Security+, Network+
>>> Network Hobbyist, VFR Private Pilot.
>>>
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>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> Your life is like a penny.  You're going to lose it.  The question is:
>> How do
>> you spend it?
>>
>> John Covici
>> [EMAIL PROTECTED]
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Re: [asterisk-users] How to send DTMF down a channel

[Frank Church]

I followed my intuition and sent a DTMF tone via the D() option in the
dial command to the destination and that causes the voice to be
transmitted after the call is answered.

I also realised that the ringing tone to the destination does not come
up until I execute a SendDTMF command before dialling the destination.


From that it appears that a command to issue a tone either before

dialling out or after the call is picked up is necessary for the
exchange of sound to take place, and I don't know whether that is a
bug, or a feature of the Asterisk design.

In verbose mode the call is interspersed with the output from the
complicated AGI script and that will make things difficult, I have the
problem (hopefully) solved now, but I will try to make the time to
send you the output of the script after I remove the AGI code from it.

In dial plan terms the idea is to generate a call file that contains
both the caller id of the caller and the destination, and when the
caller answers the call is routed to an AGI that extracts the dialled
number and dials it as though they were entered via DTMF. And it
appears that because no sound stream is generated through entry of
DTMF, no sound is transferred after the call is answered. Perhaps
trying something like that in a dial plan you create yourself will
reproduce the problem.

Thanks

Frank

On 9/15/06, Moises Silva <[EMAIL PROTECTED]> wrote:

could you post the output of the asterisk console in verbose mode?

In logger.conf

[logfiles]
console => notice,warning,error,verbose,debug

Regards

On 9/15/06, Frank Church <[EMAIL PROTECTED]> wrote:
> The program in question is an adaptation an AGI calling card program.
> It is adapted for callback by setting by channelling the callback call
> into the context used for the normal inbound leg.
>
> When used that way entering the PIN and destination number via DTMF
> works normally.
>
> I tried to implement sms/web callback simply by passing the
> destination number and the callerid in call file's parameters, so that
> the script can go straight to dialling the destination number without
> prompting the caller for it.
>
> This method results in dead air when the call is answered at the
> destination and it seems that the absence of DTMF tones for generating
> the call is the cause.
>
> Logically it should work, why the call should result in dead air
> because the destination number was not obtained through DTMF is what I
> can't understand.
>
> It appears that the absence of DTMF stops the RTP from being
> established to the destination number  or prevents the calls from
> being joined. When I dial back to the calling  number I get the busy
> voice prompt okay.
>
> Injecting the DTMF tones on the call channel is a really a kludge to
> simulate actual key presses in the hope that the dead air problem will
> go away and it .
>
> On 9/15/06, Moises Silva <[EMAIL PROTECTED]> wrote:
> > Frank. PlayDTMF and SendDTMF is the same as pressing keys at the
> > phone. Im not understanding well, can you please explain a practical
> > scenario of how do you expect it to work, and how actually works? :)
> > Thanks
> >
> > Regards
> >
> > On 9/14/06, Frank Church <[EMAIL PROTECTED]> wrote:
> > > How do you specify the channel the SendDTMF is sent on, as it has to
> > > be variable? I am able to use PlayDTMF  to send tones using the
> > > manager interface - the tones come up okay but it doesn't work.
> > >
> > > My problem comes from a working callback script that I am trying to
> > > adapt to web/SMS callback.
> > >
> > > The script is unchanged only that in the web/sms callback the
> > > destination number is passed in the call file variables to allow it be
> > > dialled after the callback is answered rather than being obtained via
> > > DTMF.
> > >
> > > The destination call is made alright, but the called party only gets
> > > dead air after the call is answered. It seems that the fact of
> > > pressing the DTMF keys activates the sound channel in way that the
> > > direct call back does not, which is why I am going through all these
> > > hoops. I just hoped that sending some DTMF tones programmatically
> > > would simulate the actual key presses.
> > >
> > > Do you have any ideas of what the problem might be?
> > >
> > >
> > >
> > > On 9/14/06, Moises Silva <[EMAIL PROTECTED]> wrote:
> > > > http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF
> > > >
> > > > Regards
> > > >
> > > > On 9/14/06, Frank Church <[EMAIL PROTECTED]> wrote:
> > > > > How can DTMF  be sent down a channel?
> > > > >
> > > > > I am thinking of method where say a channel id can be grabbed from
> > > > > Asterisk Manager events and a DTMF signal sent down that channel,
> > > > > through AGI, Asterisk Manager Interface or whatever?
> > > > >
> > > > > Is it possible to have a command in extensions.conf which can take
> > > > > both the channel and the dtmf numbers as parameters and send the DTMF
> > > > > signals?
> > > > >
> > > > > Frank Church
> > > > > 

[asterisk-users] read variable from shell script

[Christophorus Laube]

Hi list,

is it possible to call a shell script from * which returns a number or a 
string which can be read to an asterisk variable? Something like 
'Set(VAR(System(/opt/scripts/something.script)))?
Does anyone have an idea?

Regards, Christophorus
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Re: [asterisk-users] amr codec

[Steve Kennedy]

On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote:

> I have been searching, but I have not found the answer.. How might I add
> the amr codec to my asterisk server?
> I believe I found the amr source from
> http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip
> I compiled it but did not end up with any .so files like I thought I
> would need to put it into asterisk.
> Any pointers on how to get an amr codec into asterisk would be most
> helpful..

AMR is patent encumbered, just because the source is available doesn't
mean you can use it without a license.

Voiceage (at least) run licensing for AMR. It's about $1 per license
(simulataneous encode or decode) with a minimum of something like $50K.


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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[asterisk-users] "Ping" a phone

[Julian Lyndon-Smith]

Currently, I send a popup to an agent asking them to accept the call or 
not. However, they are complaining that they do not *see* the popup 
(it's onscreen for 7 seconds) and I was wanting (have been asked to 
provide) a facility where I can "ping" an agents phone before sending 
the popup.


This would make them aware that there is a call about to come in.

The best I could come up with so far is a Dial(SIP/Exten,1,g). This 
seems to work, but I was wondering if there was another more elegant 
mechanism. We currently use Cisco 7960 phones.


Julian.
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Re: [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy

[Steve Totaro]

Matt,

I am sure this is a RTFM and I am pretty sure you are using meetme 
rooms.  Just not too sure how you do the magic.


28 T1s with NFAS so 95 channels per trunk group, seven trunk groups = 
665 lines.  My client's call volume has shot from 5,000 to about 10,000 
calls a day.  Due to recent product offerings/advertising, I expect to 
be eating up 6 T1 (peak) by the end  of October.  They will eventually 
have every channel in use during peaks, whether that is in November or 
December, I am not sure.  I just know it can't break at that point due 
the the sheer expense of revenue lost for downtime.


Thanks,
Steve


Matt Florell wrote:

How many lines and agents are you looking at?

What kind of call volume?

Average expected hold time?

VICIDIAL could be an option for you since it does not use Asterisk
Queues and can already easily scale across many servers.

MATT---


On 9/15/06, Steve Totaro <[EMAIL PROTECTED]> wrote:

I have been tossing around some ideas about scaling a call center with
load balancing and redundancy and would like the comunities input,
thoughts, criticism and anything anyone wants to toss in.

The most evident thing is to start with beefy servers and only run procs
that are required.  All of the TDM boxes run stripped down versions of
Linux and Asterisk, they just take the call from the PRIs and convert
them to SIP, everything stays ulaw end to end.

*Shared queues across multiple servers would be ideal*.  I don't think
it is possible in asterisk, as is.  Maybe DUNDI could be useful but I am
not up to speed on it enough to really know.

I was toying with a concept of a DB server tracking the number of calls
to queue(s), number of agents logged into the queue(s).  Some agents
will be logged into multiple queues and providing the logic to a series
of Asterisk servers.   Calls could be made to the db to determine which
queue/server to route the call to.  In this situation, duplicate queues
would exist on several servers, so balancing would work somewhat if the
DB made the selection on which box to route the call to and which box an
agent should log into.  FastAGI and the manager interface will provide
the routing and DB updates.

Another thought was to have one central server with all of the queues
and agents, then somehow the central server would cause a "recording/CDR
server" to send re-invites to the two SIP endpoints so that the call/RTP
stream is moved to another asterisk server which would record the call
and keep the CDR info.  Again, this would be done with a DB to decide
which asterisk (recording/CDR) box has the lightest load.  It would take
the burden of maintaining the call from the "Queue" server.  I/O is the
first bottleneck in scaling when you record each and every call.

Would it be difficult to have asterisk send two SIP endpoints re-invites
and then bridge the call?  Then it is just a matter of the "Queue"
server checking the DB which recording/CDR server the call should go to
and send it a message to re-invite and bridge the endpoints.  A transfer
to a meetme is another possiblility but I want the "Queue" server out of
the stream.

Has anybody else thought through the best way to scale something like
this.  I have a DS3 and will be using all of the channels in the
semi-near future.  I need to come up with a workable plan before then.

Thanks,
Steve
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RE: [asterisk-users] "Ping" a phone

[Dean Collins]

How large is the popup?

I cant imagine when my outlook 2003 email popup appears (5 seconds I
guess) that I don't see it, though I think the movement of it appearing
on my display then receeding away has a lot to do with it.

 

Cheers,

Dean

 


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith
> Sent: Saturday, 16 September 2006 7:44 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] "Ping" a phone
> 
> Currently, I send a popup to an agent asking them to accept the call
or
> not. However, they are complaining that they do not *see* the popup
> (it's onscreen for 7 seconds) and I was wanting (have been asked to
> provide) a facility where I can "ping" an agents phone before sending
> the popup.
> 
> This would make them aware that there is a call about to come in.
> 
> The best I could come up with so far is a Dial(SIP/Exten,1,g). This
> seems to work, but I was wondering if there was another more elegant
> mechanism. We currently use Cisco 7960 phones.
> 
> Julian.
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Re: [asterisk-users] "Ping" a phone

[Julian Lyndon-Smith]

Um, it's a 1024x768 bright yellow box ;)

I know, unbelievable.

Julian.

Dean Collins wrote:

How large is the popup?

I cant imagine when my outlook 2003 email popup appears (5 seconds I
guess) that I don't see it, though I think the movement of it appearing
on my display then receeding away has a lot to do with it.

 


Cheers,

Dean

 




-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith
Sent: Saturday, 16 September 2006 7:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] "Ping" a phone

Currently, I send a popup to an agent asking them to accept the call

or

not. However, they are complaining that they do not *see* the popup
(it's onscreen for 7 seconds) and I was wanting (have been asked to
provide) a facility where I can "ping" an agents phone before sending
the popup.

This would make them aware that there is a call about to come in.

The best I could come up with so far is a Dial(SIP/Exten,1,g). This
seems to work, but I was wondering if there was another more elegant
mechanism. We currently use Cisco 7960 phones.

Julian.
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Re: [asterisk-users] "Ping" a phone

[Steve Totaro]

Julian Lyndon-Smith wrote:
Currently, I send a popup to an agent asking them to accept the call 
or not. However, they are complaining that they do not *see* the popup 
(it's onscreen for 7 seconds) and I was wanting (have been asked to 
provide) a facility where I can "ping" an agents phone before sending 
the popup.


This would make them aware that there is a call about to come in.

The best I could come up with so far is a Dial(SIP/Exten,1,g). This 
seems to work, but I was wondering if there was another more elegant 
mechanism. We currently use Cisco 7960 phones.


Julian.

You allow your agents to decline calls?!?
I think your solution is the simplest way of doing it.

Thanks,
Steve
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[asterisk-users] SHSU asterisk installation?

[Roy Sigurd Karlsbakk]

hi all

Linked from /. today, http://www.networkworld.com/news/2006/091206- 
von-sam-houston.html talks about  the Sam Houston State University  
(SHSU) migrating a rather large amount of users to asterisk. The  
article describes the installation in rather vague terms, so I was  
wondering if someone know how they plan to do this, in detail.


thanks

roy
---
"Humans mostly aren't particularly evil. They just get carried away  
by new ideas, like dressing up in jackboots and shooting people, or  
dressing up in white sheets and lynching people, or dressing up in  
tie-dye jeans and playing guitars at people"

 - Terry Pratchett
---
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]



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Re: [asterisk-users] SHSU asterisk installation?

[Steve Totaro]

Roy Sigurd Karlsbakk wrote:

hi all

Linked from /. today, 
http://www.networkworld.com/news/2006/091206-von-sam-houston.html 
talks about  the Sam Houston State University (SHSU) migrating a 
rather large amount of users to asterisk. The article describes the 
installation in rather vague terms, so I was wondering if someone know 
how they plan to do this, in detail.


thanks

roy
---
"Humans mostly aren't particularly evil. They just get carried away by 
new ideas, like dressing up in jackboots and shooting people, or 
dressing up in white sheets and lynching people, or dressing up in 
tie-dye jeans and playing guitars at people"

 - Terry Pratchett
---
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]


Yes, there have been several threads about this.


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Re: [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy

[Matt Florell]

Hello,

At this point in time VICIDIAL is more focused on outbound features,
but inbound and blended capcbilities have been part of VICIDIAL for
about two years. The most I have done inbound-only with it is 3 T1s
with 60 agents. But for outbound and inbound agents together we have
had upto 120 agents on one setup with 20 T1s(spread across 8 Asterisk
servers, 2 web servers and 1 MySQL server) handling over 200,000 calls
a day(mostly outbound of course).

The inbound portion of VICIDIAL does not have customized hold music or
periodic announcements yet, but we plan on adding those features in a
future version as we begin to focus more on inbound in the project.

We do use meetme rooms in VICIDIAL. This allows for easy third-party
calls and multiple monitoring/manager intrusion into an agent session.

The load balancing works by having the Database keep track of agents
and calls for all of the servers and then use IAX to move the calls
from where they originated to whoever the next available agent is, no
matter what server they are on. So the calls can come from anywhere
and the agents can be logged in anywhere.

If you receive inbound callerID on these calls you will also be able
to have caller information appear on the Agent's web interface. And if
they are a customer that is already in the system it will bring up
their existing record.

As for reliability, we have not had a total system failure in the last
2 years(aside from long power outages and hurricane interruptions).
MySQL can handle a tremendous volume and it is the only
total-system-single-point-of-failure in VICIDIAL, ours never crashes.
The web servers can be load balanced(no need for session-awareness)
and you can use any Apache/PHP webserver that may be on your system to
serve the AJAX-based agent web interface. As for Asterisk, we have had
servers crash periodically(a couple crashes a month across 8 servers),
but that is to be expected when you push tens of thousands of calls
through each one per day.

Will you be recording all calls in this setup?

MATT---

On 9/16/06, Steve Totaro <[EMAIL PROTECTED]> wrote:

Matt,

I am sure this is a RTFM and I am pretty sure you are using meetme
rooms.  Just not too sure how you do the magic.

28 T1s with NFAS so 95 channels per trunk group, seven trunk groups =
665 lines.  My client's call volume has shot from 5,000 to about 10,000
calls a day.  Due to recent product offerings/advertising, I expect to
be eating up 6 T1 (peak) by the end  of October.  They will eventually
have every channel in use during peaks, whether that is in November or
December, I am not sure.  I just know it can't break at that point due
the the sheer expense of revenue lost for downtime.

Thanks,
Steve


Matt Florell wrote:
> How many lines and agents are you looking at?
>
> What kind of call volume?
>
> Average expected hold time?
>
> VICIDIAL could be an option for you since it does not use Asterisk
> Queues and can already easily scale across many servers.
>
> MATT---
>
>
> On 9/15/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
>> I have been tossing around some ideas about scaling a call center with
>> load balancing and redundancy and would like the comunities input,
>> thoughts, criticism and anything anyone wants to toss in.
>>
>> The most evident thing is to start with beefy servers and only run procs
>> that are required.  All of the TDM boxes run stripped down versions of
>> Linux and Asterisk, they just take the call from the PRIs and convert
>> them to SIP, everything stays ulaw end to end.
>>
>> *Shared queues across multiple servers would be ideal*.  I don't think
>> it is possible in asterisk, as is.  Maybe DUNDI could be useful but I am
>> not up to speed on it enough to really know.
>>
>> I was toying with a concept of a DB server tracking the number of calls
>> to queue(s), number of agents logged into the queue(s).  Some agents
>> will be logged into multiple queues and providing the logic to a series
>> of Asterisk servers.   Calls could be made to the db to determine which
>> queue/server to route the call to.  In this situation, duplicate queues
>> would exist on several servers, so balancing would work somewhat if the
>> DB made the selection on which box to route the call to and which box an
>> agent should log into.  FastAGI and the manager interface will provide
>> the routing and DB updates.
>>
>> Another thought was to have one central server with all of the queues
>> and agents, then somehow the central server would cause a "recording/CDR
>> server" to send re-invites to the two SIP endpoints so that the call/RTP
>> stream is moved to another asterisk server which would record the call
>> and keep the CDR info.  Again, this would be done with a DB to decide
>> which asterisk (recording/CDR) box has the lightest load.  It would take
>> the burden of maintaining the call from the "Queue" server.  I/O is the
>> first bottleneck in scaling when you record each and every call.
>>
>> Would it be difficult to hav

Re: [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy

[Steve Totaro]

Right now we are all inbound and every call is recorded.

Matt Florell wrote:

Hello,

At this point in time VICIDIAL is more focused on outbound features,
but inbound and blended capcbilities have been part of VICIDIAL for
about two years. The most I have done inbound-only with it is 3 T1s
with 60 agents. But for outbound and inbound agents together we have
had upto 120 agents on one setup with 20 T1s(spread across 8 Asterisk
servers, 2 web servers and 1 MySQL server) handling over 200,000 calls
a day(mostly outbound of course).

The inbound portion of VICIDIAL does not have customized hold music or
periodic announcements yet, but we plan on adding those features in a
future version as we begin to focus more on inbound in the project.

We do use meetme rooms in VICIDIAL. This allows for easy third-party
calls and multiple monitoring/manager intrusion into an agent session.

The load balancing works by having the Database keep track of agents
and calls for all of the servers and then use IAX to move the calls
from where they originated to whoever the next available agent is, no
matter what server they are on. So the calls can come from anywhere
and the agents can be logged in anywhere.

If you receive inbound callerID on these calls you will also be able
to have caller information appear on the Agent's web interface. And if
they are a customer that is already in the system it will bring up
their existing record.

As for reliability, we have not had a total system failure in the last
2 years(aside from long power outages and hurricane interruptions).
MySQL can handle a tremendous volume and it is the only
total-system-single-point-of-failure in VICIDIAL, ours never crashes.
The web servers can be load balanced(no need for session-awareness)
and you can use any Apache/PHP webserver that may be on your system to
serve the AJAX-based agent web interface. As for Asterisk, we have had
servers crash periodically(a couple crashes a month across 8 servers),
but that is to be expected when you push tens of thousands of calls
through each one per day.

Will you be recording all calls in this setup?

MATT---

On 9/16/06, Steve Totaro <[EMAIL PROTECTED]> wrote:

Matt,

I am sure this is a RTFM and I am pretty sure you are using meetme
rooms.  Just not too sure how you do the magic.

28 T1s with NFAS so 95 channels per trunk group, seven trunk groups =
665 lines.  My client's call volume has shot from 5,000 to about 10,000
calls a day.  Due to recent product offerings/advertising, I expect to
be eating up 6 T1 (peak) by the end  of October.  They will eventually
have every channel in use during peaks, whether that is in November or
December, I am not sure.  I just know it can't break at that point due
the the sheer expense of revenue lost for downtime.

Thanks,
Steve


Matt Florell wrote:
> How many lines and agents are you looking at?
>
> What kind of call volume?
>
> Average expected hold time?
>
> VICIDIAL could be an option for you since it does not use Asterisk
> Queues and can already easily scale across many servers.
>
> MATT---
>
>
> On 9/15/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
>> I have been tossing around some ideas about scaling a call center 
with

>> load balancing and redundancy and would like the comunities input,
>> thoughts, criticism and anything anyone wants to toss in.
>>
>> The most evident thing is to start with beefy servers and only run 
procs
>> that are required.  All of the TDM boxes run stripped down 
versions of

>> Linux and Asterisk, they just take the call from the PRIs and convert
>> them to SIP, everything stays ulaw end to end.
>>
>> *Shared queues across multiple servers would be ideal*.  I don't 
think
>> it is possible in asterisk, as is.  Maybe DUNDI could be useful 
but I am

>> not up to speed on it enough to really know.
>>
>> I was toying with a concept of a DB server tracking the number of 
calls

>> to queue(s), number of agents logged into the queue(s).  Some agents
>> will be logged into multiple queues and providing the logic to a 
series
>> of Asterisk servers.   Calls could be made to the db to determine 
which
>> queue/server to route the call to.  In this situation, duplicate 
queues
>> would exist on several servers, so balancing would work somewhat 
if the
>> DB made the selection on which box to route the call to and which 
box an
>> agent should log into.  FastAGI and the manager interface will 
provide

>> the routing and DB updates.
>>
>> Another thought was to have one central server with all of the queues
>> and agents, then somehow the central server would cause a 
"recording/CDR
>> server" to send re-invites to the two SIP endpoints so that the 
call/RTP
>> stream is moved to another asterisk server which would record the 
call

>> and keep the CDR info.  Again, this would be done with a DB to decide
>> which asterisk (recording/CDR) box has the lightest load.  It 
would take
>> the burden of maintaining the call from the "Queue" server.  I/O 
is the

Re: [asterisk-users] SHSU asterisk installation?

[Roy Sigurd Karlsbakk]

Linked from /. today, http://www.networkworld.com/news/2006/091206- 
von-sam-houston.html talks about  the Sam Houston State University  
(SHSU) migrating a rather large amount of users to asterisk. The  
article describes the installation in rather vague terms, so I was  
wondering if someone know how they plan to do this, in detail.


Yes, there have been several threads about this.


Obviously I _have_ tried to google about this, so if you could point  
me to one of those threads, I'd be grateful


roy
---
"Humans mostly aren't particularly evil. They just get carried away  
by new ideas, like dressing up in jackboots and shooting people, or  
dressing up in white sheets and lynching people, or dressing up in  
tie-dye jeans and playing guitars at people"

 - Terry Pratchett
---
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]



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Re: [asterisk-users] SHSU asterisk installation?

[Matt Riddell (IT)]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Roy Sigurd Karlsbakk wrote:
>>> Linked from /. today,
>>> http://www.networkworld.com/news/2006/091206-von-sam-houston.html
>>> talks about  the Sam Houston State University (SHSU) migrating a
>>> rather large amount of users to asterisk. The article describes the
>>> installation in rather vague terms, so I was wondering if someone
>>> know how they plan to do this, in detail.
>>
>> Yes, there have been several threads about this.
> 
> Obviously I _have_ tried to google about this, so if you could point me
> to one of those threads, I'd be grateful

- From 3 days ago:

http://www.sineapps.com/news.php?rssid=1509

Thread here:

[asterisk-users] University switches to Asterisk

- --
Cheers,

Matt Riddell
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Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFFDCScS6d5vy0jeVcRAmQAAKCOcWU04FK6txwH1NfduA0QsFYaogCfb3a4
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Re: [asterisk-users] SHSU asterisk installation?

[Aaron Daniel]

Maybe I should just hold a conference call about all this stuff.

On Sat, 2006-09-16 at 18:21 +0200, Matt Riddell (IT) wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Roy Sigurd Karlsbakk wrote:
> >>> Linked from /. today,
> >>> http://www.networkworld.com/news/2006/091206-von-sam-houston.html
> >>> talks about  the Sam Houston State University (SHSU) migrating a
> >>> rather large amount of users to asterisk. The article describes the
> >>> installation in rather vague terms, so I was wondering if someone
> >>> know how they plan to do this, in detail.
> >>
> >> Yes, there have been several threads about this.
> > 
> > Obviously I _have_ tried to google about this, so if you could point me
> > to one of those threads, I'd be grateful
> 
> - From 3 days ago:
> 
> http://www.sineapps.com/news.php?rssid=1509
> 
> Thread here:
> 
> [asterisk-users] University switches to Asterisk
> 
> - --
> Cheers,
> 
> Matt Riddell
> ___
> 
> http://www.sineapps.com/news.php (Daily Asterisk News - html)
> http://wap.sineapps.com (Daily Asterisk News for your cellphone)
> http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.2 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
> 
> iD8DBQFFDCScS6d5vy0jeVcRAmQAAKCOcWU04FK6txwH1NfduA0QsFYaogCfb3a4
> KojTXtiqQLK7YNtXp+4Qh+I=
> =T9d+
> -END PGP SIGNATURE-
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[asterisk-users] USA Regulatons

[Marnus van Niekerk]

Guys,

this may be a off question but where do I start to find all the
regulations to comply with to set up a residential voip phone service
in the US?

The tech stuff I am fine with but the legal is where I need some
pointers.

Also I am not looking for a flame war just pointers on the legal issues.

Thanx


M

-- 

"Opportunity is missed by most people because it is
dressed in overalls and looks like work."

Thomas Alva Edison - Inventor of 1093 patents,
including the light bulb, phonogram and motion pictures.



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Re: [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy

[Matt Florell]

To maintain high recording quality with no audio skips we have found
that you should not go over 50 conversations being recorded on a
single server. What have you found is your limit while maintaining
very good audio quality?

MATT---

On 9/16/06, Steve Totaro <[EMAIL PROTECTED]> wrote:

Right now we are all inbound and every call is recorded.

Matt Florell wrote:
> Hello,
>
> At this point in time VICIDIAL is more focused on outbound features,
> but inbound and blended capcbilities have been part of VICIDIAL for
> about two years. The most I have done inbound-only with it is 3 T1s
> with 60 agents. But for outbound and inbound agents together we have
> had upto 120 agents on one setup with 20 T1s(spread across 8 Asterisk
> servers, 2 web servers and 1 MySQL server) handling over 200,000 calls
> a day(mostly outbound of course).
>
> The inbound portion of VICIDIAL does not have customized hold music or
> periodic announcements yet, but we plan on adding those features in a
> future version as we begin to focus more on inbound in the project.
>
> We do use meetme rooms in VICIDIAL. This allows for easy third-party
> calls and multiple monitoring/manager intrusion into an agent session.
>
> The load balancing works by having the Database keep track of agents
> and calls for all of the servers and then use IAX to move the calls
> from where they originated to whoever the next available agent is, no
> matter what server they are on. So the calls can come from anywhere
> and the agents can be logged in anywhere.
>
> If you receive inbound callerID on these calls you will also be able
> to have caller information appear on the Agent's web interface. And if
> they are a customer that is already in the system it will bring up
> their existing record.
>
> As for reliability, we have not had a total system failure in the last
> 2 years(aside from long power outages and hurricane interruptions).
> MySQL can handle a tremendous volume and it is the only
> total-system-single-point-of-failure in VICIDIAL, ours never crashes.
> The web servers can be load balanced(no need for session-awareness)
> and you can use any Apache/PHP webserver that may be on your system to
> serve the AJAX-based agent web interface. As for Asterisk, we have had
> servers crash periodically(a couple crashes a month across 8 servers),
> but that is to be expected when you push tens of thousands of calls
> through each one per day.
>
> Will you be recording all calls in this setup?
>
> MATT---
>
> On 9/16/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
>> Matt,
>>
>> I am sure this is a RTFM and I am pretty sure you are using meetme
>> rooms.  Just not too sure how you do the magic.
>>
>> 28 T1s with NFAS so 95 channels per trunk group, seven trunk groups =
>> 665 lines.  My client's call volume has shot from 5,000 to about 10,000
>> calls a day.  Due to recent product offerings/advertising, I expect to
>> be eating up 6 T1 (peak) by the end  of October.  They will eventually
>> have every channel in use during peaks, whether that is in November or
>> December, I am not sure.  I just know it can't break at that point due
>> the the sheer expense of revenue lost for downtime.
>>
>> Thanks,
>> Steve
>>
>>
>> Matt Florell wrote:
>> > How many lines and agents are you looking at?
>> >
>> > What kind of call volume?
>> >
>> > Average expected hold time?
>> >
>> > VICIDIAL could be an option for you since it does not use Asterisk
>> > Queues and can already easily scale across many servers.
>> >
>> > MATT---
>> >
>> >
>> > On 9/15/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
>> >> I have been tossing around some ideas about scaling a call center
>> with
>> >> load balancing and redundancy and would like the comunities input,
>> >> thoughts, criticism and anything anyone wants to toss in.
>> >>
>> >> The most evident thing is to start with beefy servers and only run
>> procs
>> >> that are required.  All of the TDM boxes run stripped down
>> versions of
>> >> Linux and Asterisk, they just take the call from the PRIs and convert
>> >> them to SIP, everything stays ulaw end to end.
>> >>
>> >> *Shared queues across multiple servers would be ideal*.  I don't
>> think
>> >> it is possible in asterisk, as is.  Maybe DUNDI could be useful
>> but I am
>> >> not up to speed on it enough to really know.
>> >>
>> >> I was toying with a concept of a DB server tracking the number of
>> calls
>> >> to queue(s), number of agents logged into the queue(s).  Some agents
>> >> will be logged into multiple queues and providing the logic to a
>> series
>> >> of Asterisk servers.   Calls could be made to the db to determine
>> which
>> >> queue/server to route the call to.  In this situation, duplicate
>> queues
>> >> would exist on several servers, so balancing would work somewhat
>> if the
>> >> DB made the selection on which box to route the call to and which
>> box an
>> >> agent should log into.  FastAGI and the manager interface will
>> provide
>> >> the routing and DB up

Re[2]: [asterisk-users] SHSU asterisk installation?

[Melcon Moraes]

That would be superb!

 -Original Message-
From:   Aaron Daniel <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Cc: 
Sent:  Sat, 16 Sep 2006 11:32:00 -0500
Delivered:  Sat,  16 Sep 2006 13:21:48 
Subject:[asterisk-users] SHSU asterisk installation?

Maybe I should just hold a conference call about all this stuff.

On Sat, 2006-09-16 at 18:21 +0200, Matt Riddell (IT) wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Roy Sigurd Karlsbakk wrote:
> >>> Linked from /. today,
> >>> http://www.networkworld.com/news/2006/091206-von-sam-houston.html
> >>> talks about  the Sam Houston State University (SHSU) migrating a
> >>> rather large amount of users to asterisk. The article describes the
> >>> installation in rather vague terms, so I was wondering if someone
> >>> know how they plan to do this, in detail.
> >>
> >> Yes, there have been several threads about this.
> > 
> > Obviously I _have_ tried to google about this, so if you could point me
> > to one of those threads, I'd be grateful
> 
> - From 3 days ago:
> 
> http://www.sineapps.com/news.php?rssid=1509
> 
> Thread here:
> 
> [asterisk-users] University switches to Asterisk
> 
> - --
> Cheers,
> 
> Matt Riddell
> ___
> 
> http://www.sineapps.com/news.php (Daily Asterisk News - html)
> http://wap.sineapps.com (Daily Asterisk News for your cellphone)
> http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.2 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
> 
> iD8DBQFFDCScS6d5vy0jeVcRAmQAAKCOcWU04FK6txwH1NfduA0QsFYaogCfb3a4
> KojTXtiqQLK7YNtXp+4Qh+I=
> =T9d+
> -END PGP SIGNATURE-
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E-mail classificado pelo Identificador de Spam Inteligente Terra.
Para alterar a categoria classificada, visite
http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&_l=1,1158424231.497848.25931.almora.hst.terra.com.br,5369,Des15,Des15

 --Original Message Ends--

-- 
Melcon Moraes <[EMAIL PROTECTED]>

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Re: [asterisk-users] Asterisk as a gateway to SER

[Jeremy McNamara]

Siqhamo Sifo wrote:

My asterisk is giving me problems when I use it as a pstn gateway to SER ,
basically what happens is that its either I get one way audio or no audio
at all when I make pstn calls via asterisk from sip clients registered
with SER.




SER itself is just a SIP Proxy. So your issue may be the fact that you 
are not re-writing the SIP headers, if your endpoints are behind NAT.


Diagnose the situation then provide detailed information, if you expect 
any assistance - We cannot read minds, yet.




Jeremy McNamara
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RE: [asterisk-users] Dell hardware ...

[Ed Greenberg]

Precisely our configuration. Dell 1850 with 4 port PRI digium cards.

No issues on my last two consulting jobs.



--On Tuesday, September 12, 2006 4:14 PM +0200 Arjan Kroon 
<[EMAIL PROTECTED]> wrote:



Hi, Alan,

We use Dell 1850 (about 20 server) and we have 4 ports PRI Digium cards
in it and it works perfect.
It is almost Plug&Play.

greetings

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RE: [asterisk-users] Dell hardware ...

[Kevin Kiely]

This information seems to indicate there is a problem with the 1850 and the
onboard nic.

http://connection-telecom.com/support.html




-Original Message-
From: Ed Greenberg [mailto:[EMAIL PROTECTED] 
Sent: Saturday, September 16, 2006 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Dell hardware ...

Precisely our configuration. Dell 1850 with 4 port PRI digium cards.

No issues on my last two consulting jobs.



--On Tuesday, September 12, 2006 4:14 PM +0200 Arjan Kroon 
<[EMAIL PROTECTED]> wrote:

> Hi, Alan,
>
> We use Dell 1850 (about 20 server) and we have 4 ports PRI Digium cards
> in it and it works perfect.
> It is almost Plug&Play.
>
> greetings
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-- 
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.12.4/449 - Release Date: 9/15/2006
 

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Re: [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problem with module versionmagic

[Tzafrir Cohen]

On Fri, Sep 15, 2006 at 02:58:02PM +0200, Robert Rozman wrote:
> I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel 
> :
> 
> Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64
> x86_64 x86_64 GNU/Linux
> 
> 
> and Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s.
> 
> I get this :
> 
> laps1:~/Voipy/Bristuff/bristuff-0.3.0-PRE-1s/zaptel # modprobe zaphfc
> FATAL: Error inserting zaphfc
> (/lib/modules/2.6.13-15.11-smp/misc/zaphfc.ko): Invalid module format
> 
> and this in dmesg :
> 
> zaphfc: version magic '2.6.13-15.11-smp gcc-4.0' should be '2.6.13-15.11-smp
> SMP gcc-4.0'

This means you built zaptel with the wrong kernel headers. Is there a
SUSE power user in the crowd?

> 
> 
> What am I doing wrong? Anyone sucessfully using latest Bristuff under Suse ?

A bit of trivia: kapejod (the developer of bristuff) is known to be 
using SUSE as the main system. Not sure of the version. 

However this is basically a kernel module building question. As such you
can also ask in in more specific SUSE places.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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RE: [asterisk-users] Reliability of the newer IAXy's

[Lists]

> -Original Message-
> From: Andrew Joakimsen [mailto:[EMAIL PROTECTED]
> Sent: Friday, September 15, 2006 4:37 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-
> Commercial Discussion
> Subject: Re: [asterisk-users] Reliability of the newer IAXy's
> 
> This goes for everyone;
> 
> With an Asterisk server directly assigned a public static IP and with
> the client SIP devices behind a NAT (usually Linksys routers) I have
> yet to have a NAT issue, even when the client is behind dual NATs.
> 



Of which MY issue is that both the client and asterisk would be behind NAT 
boxes. And I was looking for something a little easier than trying to tyrick 
SIP into working.

Robert
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Re: [asterisk-users] ZT_SPANCONFIG failed on span 1: No such device or address (6)

[Tzafrir Cohen]

On Fri, Sep 15, 2006 at 11:09:51AM -0500, Juan Miguel Yamakawa wrote:
> Help me please..
> 
> ZT_SPANCONFIG failed on span 1: No such device or address (6)
> 
> how can i fixed this problem.

This means that a span linein /etc/zaptel.conf did not fit the spans
that exist on your system now (see /proc/zaptel/* . Each file under
/proc/zaptel represents a span).

What card(s) do you have?

What's the output of:   cat /proc/zaptel/*

To generate a configuration that matches your currently-loaded modules,
use xpp/utils/genzaptelconf . Use the option -d to try to detect the
cards on your system. Note that those are not necessarily the same cards
you'll have next time system boots.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
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Re: Re[2]: [asterisk-users] SHSU asterisk installation?

[Marco Mouta]

yes in did, this could be one excellent case study for all asterisk Community! Please let me know when it will be held!
 
By the way, keep asterisk community following your steps would be great for knowledge of everyone and to solve possible problems you could find! 
On 9/16/06, Melcon Moraes <[EMAIL PROTECTED]> wrote:
That would be superb!-Original Message-From:   Aaron Daniel <
[EMAIL PROTECTED]>To: Asterisk Users Mailing List - Non-Commercial Discussion Cc:Sent:  Sat, 16 Sep 2006 11:32:00 -0500
Delivered:  Sat,  16 Sep 2006 13:21:48Subject:[asterisk-users] SHSU asterisk installation?Maybe I should just hold a conference call about all this stuff.On Sat, 2006-09-16 at 18:21 +0200, Matt Riddell (IT) wrote:
> -BEGIN PGP SIGNED MESSAGE-> Hash: SHA1>> Roy Sigurd Karlsbakk wrote:> >>> Linked from /. today,> >>> 
http://www.networkworld.com/news/2006/091206-von-sam-houston.html> >>> talks about  the Sam Houston State University (SHSU) migrating a> >>> rather large amount of users to asterisk. The article describes the
> >>> installation in rather vague terms, so I was wondering if someone> >>> know how they plan to do this, in detail.> >>> >> Yes, there have been several threads about this.
> >> > Obviously I _have_ tried to google about this, so if you could point me> > to one of those threads, I'd be grateful>> - From 3 days ago:>> 
http://www.sineapps.com/news.php?rssid=1509>> Thread here:>> [asterisk-users] University switches to Asterisk>> - --> Cheers,>> Matt Riddell> ___
>> http://www.sineapps.com/news.php (Daily Asterisk News - html)> http://wap.sineapps.com (Daily Asterisk News for your cellphone)
> http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)> -BEGIN PGP SIGNATURE-> Version: GnuPG v1.4.2 (MingW32)> Comment: Using GnuPG with Mozilla - 
http://enigmail.mozdev.org>> iD8DBQFFDCScS6d5vy0jeVcRAmQAAKCOcWU04FK6txwH1NfduA0QsFYaogCfb3a4> KojTXtiqQLK7YNtXp+4Qh+I=> =T9d+> -END PGP SIGNATURE-
> ___> --Bandwidth and Colocation provided by Easynews.com -->> asterisk-users mailing list> To UNSUBSCRIBE or update options visit:
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E-mail classificado pelo Identificador de Spam Inteligente Terra.Para alterar a categoria classificada, visite
http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&_l=1,1158424231.497848.25931.almora.hst.terra.com.br,5369,Des15,Des15--Original Message EndsMelcon Moraes <
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  http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,
Marco Mouta 
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Re: [asterisk-users] read variable from shell script

[Tzafrir Cohen]

On Sat, Sep 16, 2006 at 01:15:01PM +0200, Christophorus Laube wrote:
> Hi list,
> 
> is it possible to call a shell script from * which returns a number or a 
> string which can be read to an asterisk variable? Something like 
> 'Set(VAR(System(/opt/scripts/something.script)))?
> Does anyone have an idea?

http://pbxfreeware.org/app_backticks.c

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Re: [asterisk-users] amr codec

[Net Nut]

So with that said, can anyone recommend a way that I can get a sip
client on a cell phone that uses H.263 and amr to talk to an asterisk
system?
Is it just not possible because of licensing? It sounds kind of lame to
have a sip client that can't talk to anything else because of codecs..


Steve Kennedy wrote:
> On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote:
>
>   
>> I have been searching, but I have not found the answer.. How might I add
>> the amr codec to my asterisk server?
>> I believe I found the amr source from
>> http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip
>> I compiled it but did not end up with any .so files like I thought I
>> would need to put it into asterisk.
>> Any pointers on how to get an amr codec into asterisk would be most
>> helpful..
>> 
>
> AMR is patent encumbered, just because the source is available doesn't
> mean you can use it without a license.
>
> Voiceage (at least) run licensing for AMR. It's about $1 per license
> (simulataneous encode or decode) with a minimum of something like $50K.
>
>
> Steve
>
>   
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[asterisk-users] RE: [Asterisk-video] VXIasterisk is available !

[Dean Collins]

Yeh well I don't and I have just as much right to this list as you do.

I think it's fine for a company, commercial or otherwise to post product
announcements in the non -biz list.

As long as they only post it once, and it's not an announcement about a
price change or something equally lame (like this Friday 20% off
polycoms etc).

But if a company has gone to all of the effort to develop and asterisk
add-on and they announce the launch in the users list once, then I'm
grateful for the heads up.

You guys act like this is hallowed turf or something.

It's not, it's a user list and this product certainly interested me so
thanks for the heads up. Anyone else, feel free to delete.

 

Cheers,

Dean

 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-video-
> [EMAIL PROTECTED] On Behalf Of Michael.Benarouch
> Sent: Tuesday, 15 July 2003 11:34 PM
> To: Development discussion of video media support in Asterisk
> Subject: Re: [Asterisk-video] VXIasterisk is available !
> 
> ya fully agree
> Micke
> - Original Message -
> From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
> To: "Development discussion of video media support in Asterisk"
> <[EMAIL PROTECTED]>
> Sent: Saturday, September 16, 2006 5:08 PM
> Subject: Re: [Asterisk-video] VXIasterisk is available !
> 
> 
> > better post this sort of spam in -biz
> > commercial products belong there, not here, or perhaps nowhere
> >
> > roy
> >
> > On 14. sep. 2006, at 11.59, [EMAIL PROTECTED] wrote:
> >
> >> After launching the FFasterisk project, "i6net" is launching
VXIasterisk
> >> : the
> >> VoiceXML browser for Asterisk.
> >> The VoiceXML browser supports Voice and Video with Asterisk.
> >>
> >> VXIasterisk allows to create easily  powerful video portals for
> >> Asterisk.
> >>
> >> For more informations,  http://products.i6net.com
> >>
> >>
> >> i6net
> >> ___
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> >>
> >> asterisk-video mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-video
> >>
> >
> > ---
> > "Humans mostly aren't particularly evil. They just get carried away
by
> > new ideas, like dressing up in jackboots and shooting people, or
dressing
> > up in white sheets and lynching people, or dressing up in  tie-dye
jeans
> > and playing guitars at people"
> >  - Terry Pratchett
> > ---
> > Roy Sigurd Karlsbakk
> > [EMAIL PROTECTED]
> >
> >
> >
> > ___
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> > asterisk-video mailing list
> > To UNSUBSCRIBE or update options visit:
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> >
> >
> > --
> > No virus found in this incoming message.
> > Checked by AVG Free Edition.
> > Version: 7.1.405 / Virus Database: 268.12.4/448 - Release Date:
14/09/2006
> >
> >
> 
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[asterisk-users] Mediatrix 1204

[Bill Michaelson]

Would anyone be kind enough to post a sip.conf fragment as a sample for 
use with a Mediatrix 1204?


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Re: [asterisk-users] Mediatrix 1204

[Florian Overkamp]

Bill Michaelson wrote:
Would anyone be kind enough to post a sip.conf fragment as a sample for 
use with a Mediatrix 1204?


Ours works with:

[mtrix1]
type=peer
host=172.28.4.46
mask=255.255.255.255
context=in-mtrix1
qualify=no
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw


Best regards,
Florian
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[asterisk-users] Calling to PSTN newbie question

[Pablo Almido]

Hi list, I have a asterisk and sopthones working well, and I make a configuration for it work with a voice gateway Addpac 2120 (4port FXO y 4 ports FXS), I have connected my gateway to my PBX, when I try to call to PSTN from my softphone, I have a trouble that Asterisk add the number 9 over the number that I dial. How can ignore that number 9 when I dial from my softphone?  

 
 
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Re: [asterisk-users] Calling to PSTN newbie question

[Andrew Joakimsen]

In your extensions.conf you probably have:

exten => _9nxxnxx,1,Dial(ZAP/g1/${EXTEN},180,r)

Change that to:

exten => _9nxxnxx,1,Dial(ZAP/g1/${EXTEN:1},180,r)

Of couse use the values that you need, not that exact example...

On 9/16/06, Pablo Almido <[EMAIL PROTECTED]> wrote:


Hi list, I have a asterisk and sopthones working well, and I make a
configuration for it work with a voice gateway Addpac 2120 (4port FXO y 4
ports FXS), I have connected my gateway to my PBX, when I try to call to
PSTN from my softphone, I have a trouble that Asterisk add the number 9 over
the number that I dial. How can ignore that number 9 when I dial from my
softphone?


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Re: [asterisk-users] amr codec

[Tim Panton]

On 16 Sep 2006, at 20:38, Net Nut wrote:


So with that said, can anyone recommend a way that I can get a sip
client on a cell phone that uses H.263 and amr to talk to an asterisk
system?
Is it just not possible because of licensing? It sounds kind of  
lame to

have a sip client that can't talk to anything else because of codecs..


Well Asterisk does not _have_ to have an amr codec for you
to be able to use your handset. If you have several of these
handsets or other devices that support amr, then asterisk can
route calls between them, just passing the stream through.

If you want any of the interesting asterisk features, then
it will need to transcode, and then Steve's right, not only do you have
to add codec code to Asterisk (which is almost certainly a GPL  
violation)
you also have to pay the patent holder for any commercial use of the  
codec.


Your best hope is if a few of us can persuade Digium to support amr
in the same way that they license g729 .

Tim.
Tim Panton

www.mexuar.com



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[asterisk-users] Polycom programmable buttons

[Ron McCarthy]

Hello list!Is there a way to program one of the buttons on the 501 (Like the services button) to do on the fly call recording? So in the middle of the phonecall you can record the call without have to do a transfer type of setup. Ive looked at the manual but cant seem how to do that, I only see how you can "remap" a key function.
Any help would be great!TIARon
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Re: [asterisk-users] call across 2 asterisks

[Andrew Joakimsen]

So the config is stored in one db shared between two asterisk? What
you need to do is setup an IAX trunk between the two machines and then
tell it to dial UA2 on the local machine, and then the extension for
UA2 on the 2nd asterisk. Setup both the same on the other machine.

Now do you have the users always fixed to one machine? You would need
to figure a solution for CDR and voicemail of not... CDR and voicemail
you can do in a database as well..

On 9/15/06, unplug <[EMAIL PROTECTED]> wrote:

How can I make this work (UA1 makes call to UA2)?
If I issue a command Dial(SIP/UA2), the call will fail.  Someone in
the forum said that Asterisk doesn't support the configuration below.
But I am thinking about the possibility to make it works.  Anyone can
share if it really works?

UA1UA2
--NAT1NAT2
  |  |
  |  |
Asterisk1   Asterisk2
  \ /
   \ /
 - (ARA) DB
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[asterisk-users] Re: help connecting cell phone, chan_bluetooth

[Todd]

Mauricio Mantilla  gmail.com> writes:

> 
> 
> Hi,I'm trying to connect my cell phone (motorola V3) to asterisk, using this
guide: 
> http://www.thetechguide.com/howto/asterisk/chanbluetooth.html
> Everything has worked ok, but when I actually want to start
> asterisk, my phone doesn't connect all the way. All I'm getting in the
> asterisk CLI is this:[AG] Motorola < AT+BRSF=23and my phone keeps asking the
pin number in order to get paired with asterisk, but apparently it never gets
validated.
> I changed the hcid.conf file so it wouldn't ask for a pin
> number, and now the phone doesn't ask for it anymore, but I think it
> still can't connect, cause I'm still getting the same message in the
> asterisk CLI, every few seconds.Does anyone have an idea of what could be
wrong? can it be something wrong with my phone or my bluetooth dongle? or
something about configuring asterisk?Any help would be really appreciated,
> 
> 
> Mauricio Mantilla
> 
> 
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> 

I am following the same guide using trixbox, but after I 'tar xzf
bluetoothfiles.tar.gz' and then 'cd /usr/src/asterisk' and perform the 'make
clean' I get the following error:

make: *** No rule to make target `clean'.  Stop.

Any thoughts I why that may be? Can I use these same instructions for trixbox or
do I have to go back to Asterisk version pre-trixbox?

Thanks---Todd




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[asterisk-users] Re: Reliability of the newer IAXy's

[Martin Joseph]

On 2006-09-15 13:42:21 -0700, "Lists" <[EMAIL PROTECTED]> said:

Not sure what to tell you. But for the price, I might have to try one 
of these instead:
 
http://cgi.ebay.com/IAX-Native-FXS-for-Digium-Asterisk-VoIP-PBX-Beats-IAXy_W0QQitemZ130026839028QQihZ003QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem


There 


is also the AG168V which is a relatively inexpensive ATA that can be 
configured for SIP or for IAX depending on firmware flash.


I have a couple of these and they work well.

I never got the IAXy because it is so limited with respect to codecs 
and kind of expensive also...




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Re: [asterisk-users] SHSU asterisk installation?

[Paul Hales]

And I for one would like a transcript or recording of it!

PaulH

Melcon Moraes wrote:


That would be superb!

-Original Message-
From:   Aaron Daniel <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Cc: 
Sent:  Sat, 16 Sep 2006 11:32:00 -0500
Delivered:  Sat,  16 Sep 2006 13:21:48 
Subject:[asterisk-users] SHSU asterisk installation?


Maybe I should just hold a conference call about all this stuff.

On Sat, 2006-09-16 at 18:21 +0200, Matt Riddell (IT) wrote:
 


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Roy Sigurd Karlsbakk wrote:
   


Linked from /. today,
http://www.networkworld.com/news/2006/091206-von-sam-houston.html
talks about  the Sam Houston State University (SHSU) migrating a
rather large amount of users to asterisk. The article describes the
installation in rather vague terms, so I was wondering if someone
know how they plan to do this, in detail.
 


Yes, there have been several threads about this.
   


Obviously I _have_ tried to google about this, so if you could point me
to one of those threads, I'd be grateful
 


- From 3 days ago:

http://www.sineapps.com/news.php?rssid=1509

Thread here:

[asterisk-users] University switches to Asterisk

- --
Cheers,

Matt Riddell
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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E-mail classificado pelo Identificador de Spam Inteligente Terra.
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Re: [asterisk-users] How to send DTMF down a channel

[Moises Silva]

Hi, Actually I have several AGI scripts, manager "Originates" and Call
files doing different stuff applying the same procedure, thats why I
asked about console messages. It seems very weird to me.

Regards

On 9/16/06, Frank Church <[EMAIL PROTECTED]> wrote:

I followed my intuition and sent a DTMF tone via the D() option in the
dial command to the destination and that causes the voice to be
transmitted after the call is answered.

I also realised that the ringing tone to the destination does not come
up until I execute a SendDTMF command before dialling the destination.

>From that it appears that a command to issue a tone either before
dialling out or after the call is picked up is necessary for the
exchange of sound to take place, and I don't know whether that is a
bug, or a feature of the Asterisk design.

In verbose mode the call is interspersed with the output from the
complicated AGI script and that will make things difficult, I have the
problem (hopefully) solved now, but I will try to make the time to
send you the output of the script after I remove the AGI code from it.

In dial plan terms the idea is to generate a call file that contains
both the caller id of the caller and the destination, and when the
caller answers the call is routed to an AGI that extracts the dialled
number and dials it as though they were entered via DTMF. And it
appears that because no sound stream is generated through entry of
DTMF, no sound is transferred after the call is answered. Perhaps
trying something like that in a dial plan you create yourself will
reproduce the problem.

Thanks

Frank

On 9/15/06, Moises Silva <[EMAIL PROTECTED]> wrote:
> could you post the output of the asterisk console in verbose mode?
>
> In logger.conf
>
> [logfiles]
> console => notice,warning,error,verbose,debug
>
> Regards
>
> On 9/15/06, Frank Church <[EMAIL PROTECTED]> wrote:
> > The program in question is an adaptation an AGI calling card program.
> > It is adapted for callback by setting by channelling the callback call
> > into the context used for the normal inbound leg.
> >
> > When used that way entering the PIN and destination number via DTMF
> > works normally.
> >
> > I tried to implement sms/web callback simply by passing the
> > destination number and the callerid in call file's parameters, so that
> > the script can go straight to dialling the destination number without
> > prompting the caller for it.
> >
> > This method results in dead air when the call is answered at the
> > destination and it seems that the absence of DTMF tones for generating
> > the call is the cause.
> >
> > Logically it should work, why the call should result in dead air
> > because the destination number was not obtained through DTMF is what I
> > can't understand.
> >
> > It appears that the absence of DTMF stops the RTP from being
> > established to the destination number  or prevents the calls from
> > being joined. When I dial back to the calling  number I get the busy
> > voice prompt okay.
> >
> > Injecting the DTMF tones on the call channel is a really a kludge to
> > simulate actual key presses in the hope that the dead air problem will
> > go away and it .
> >
> > On 9/15/06, Moises Silva <[EMAIL PROTECTED]> wrote:
> > > Frank. PlayDTMF and SendDTMF is the same as pressing keys at the
> > > phone. Im not understanding well, can you please explain a practical
> > > scenario of how do you expect it to work, and how actually works? :)
> > > Thanks
> > >
> > > Regards
> > >
> > > On 9/14/06, Frank Church <[EMAIL PROTECTED]> wrote:
> > > > How do you specify the channel the SendDTMF is sent on, as it has to
> > > > be variable? I am able to use PlayDTMF  to send tones using the
> > > > manager interface - the tones come up okay but it doesn't work.
> > > >
> > > > My problem comes from a working callback script that I am trying to
> > > > adapt to web/SMS callback.
> > > >
> > > > The script is unchanged only that in the web/sms callback the
> > > > destination number is passed in the call file variables to allow it be
> > > > dialled after the callback is answered rather than being obtained via
> > > > DTMF.
> > > >
> > > > The destination call is made alright, but the called party only gets
> > > > dead air after the call is answered. It seems that the fact of
> > > > pressing the DTMF keys activates the sound channel in way that the
> > > > direct call back does not, which is why I am going through all these
> > > > hoops. I just hoped that sending some DTMF tones programmatically
> > > > would simulate the actual key presses.
> > > >
> > > > Do you have any ideas of what the problem might be?
> > > >
> > > >
> > > >
> > > > On 9/14/06, Moises Silva <[EMAIL PROTECTED]> wrote:
> > > > > http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF
> > > > >
> > > > > Regards
> > > > >
> > > > > On 9/14/06, Frank Church <[EMAIL PROTECTED]> wrote:
> > > > > > How can DTMF  be sent down a channel?
> > > > > >
> > > > > > I am thinking of method where say

[asterisk-users] Wrong outgoing port

[Master_PE]

Hi,

I have just compiled
trion*CLI> show version
Asterisk 1.2.12.1 built by root @ trion on a i686 running Linux on  
2006-09-16 16:39:13 UTC


I have copyed the old configuration files from the stable debian  
version 1:1.0.7.df

I have changed in sip.conf the bindport (port=) to (bindport=) .

When i try to connect to my provider it doesn't work becourse it  
try's to connect to port 5060. sip debug says

Retransmitting #5 (NAT) to 62.177.135.42:5060:
REGISTER sip:62.177.135.42 SIP/2.0

trion*CLI> sip show registry
62.177.135.41:38383 31##120 Request Sent

What's wrong. Somehow it looks like a bug.

Regards,

M. Piscaer

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Re: [asterisk-users] Mediatrix 1204

[C F]

I have the same setup as Florian, however I have dtmfmode set to rfc
instead of inband

On 9/16/06, Florian Overkamp <[EMAIL PROTECTED]> wrote:

Bill Michaelson wrote:
> Would anyone be kind enough to post a sip.conf fragment as a sample for
> use with a Mediatrix 1204?

Ours works with:

[mtrix1]
type=peer
host=172.28.4.46
mask=255.255.255.255
context=in-mtrix1
qualify=no
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw


Best regards,
Florian
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Re: [asterisk-users] amr codec

[Steve Underwood]

Steve Kennedy wrote:


On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote:

 


I have been searching, but I have not found the answer.. How might I add
the amr codec to my asterisk server?
I believe I found the amr source from
http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip
I compiled it but did not end up with any .so files like I thought I
would need to put it into asterisk.
Any pointers on how to get an amr codec into asterisk would be most
helpful..
   



AMR is patent encumbered, just because the source is available doesn't
mean you can use it without a license.

Voiceage (at least) run licensing for AMR. It's about $1 per license
(simulataneous encode or decode) with a minimum of something like $50K.
 

While the Voiceage licence for G.729 seems to cover you for patent 
issues on that codec, their licence for AMR does not. Their patent pool 
does not include all those who claim a patent on AMR. So, you need their 
licence plus you need to search out and licence other people's patents 
too. I assume this is why the AMR licence appears cheaper than the G.729 
one.


Steve'
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Re: [asterisk-users] call across 2 asterisks

[unplug]

On 9/17/06, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:

So the config is stored in one db shared between two asterisk?

Yes, the configuration files are shared between them.

What you need to do is setup an IAX trunk between the two machines and then

tell it to dial UA2 on the local machine, and then the extension for
UA2 on the 2nd asterisk. Setup both the same on the other machine.

Do you mean my configuration works?  Could you express it a little bit more?



On 9/15/06, unplug <[EMAIL PROTECTED]> wrote:
> How can I make this work (UA1 makes call to UA2)?
> If I issue a command Dial(SIP/UA2), the call will fail.  Someone in
> the forum said that Asterisk doesn't support the configuration below.
> But I am thinking about the possibility to make it works.  Anyone can
> share if it really works?
>
> UA1UA2
> --NAT1NAT2
>   |  |
>   |  |
> Asterisk1   Asterisk2
>   \ /
>\ /
>  - (ARA) DB
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[asterisk-users] system cmd

[unplug]

How can I use a system cmd to get back the return value in dial plan?
Say, I want to run a script using system cmd to get the hostname.
System(hostname)
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