Re: [asterisk-users] G.729 codec between avaya and asterisk

[Anselm Martin Hoffmeister]

Am Dienstag, den 23.10.2007, 22:21 -0700 schrieb satish patel:
> there is no special requiremnt to use g.729 but day to day my sip
> client incressing thats why some time i got breaking voice or voice
> quality not much better i think in LAN there is lots of brodcat on
> lan 

If your LAN is congested and a lot of single packet delay happens, you
should improve the LAN. You cannot run a LAN at 99% saturation with
VoIP, it will just not work, with packet drop rates and delays making
phone calls more of a earth-to-moon radio experience ("Houston *crackle*
*crackle* have *crackle* problem").

If _all_ that traffic is VoIP, G729 might help a bit, but I would not
expect it to get around all your bandwidth problems. Try to improve the
network first.

One interesting aspect of g729 might be that your sip client phones that
live behind a DSL line might profit from the smaller bandwidth
requirement on their side.

> if i purches g.729 transcoder license for asterisk to convert g.729 to
> g.711 then  it will work or not

I _think_ it will work (btw this is, as of some website I found, the
"main revenue stream" of Digium, so they will be interested in having it
working). Others with real-world experience could tell you.

> but why i need codec on trunk 

Codec stands for coding-decoding (or something similar). If you imagine
the "original signal" as voice and sound, meaning variations in air
pressure around the membrane of the telephone handpiece microphone, then
every digital representation is a kind of "coding". This even refers to
8-bit-wave, which is the most obvious way of encoding: It merely writes
down the voltage level at the microphone input in the range -128 to +127
(IIRC, correct me if I am wrong). Accordingly, 16-bit wave has the
higher precision of -32768 to +32767.

G711 is - again, if I remember correctly - an adaptation of these bytes
to a logarithmic scales, bearing in mind the idea that small changes in
the higher ranges are treated differently from small changes in the
near-0-region. Something like the fiction bytestream value 0 1 2 3
representing the scale 0 4 6 7 of microphone values, instead of linear
data. Please research this yourself if you are interested in details.

G711 is the standard (and usually, the only available) codec for
ISDN/T1/E1... Europeans and US Americans established two different kinds
of G711 (µ-law and a-law) which seem to be functionally similar.

BR
Anselm



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Re: [asterisk-users] opensr Vs freeswitch SIP proxy server

[satish patel]

if u dont understand about stuff then u can not blam on lanugage dear.

Baji Panchumarti <[EMAIL PROTECTED]> wrote:  With all due respect, please try 
not to make up spellings based
 on pronunciation. There is no  "taff task"  it is  "tough task".

 If someone is going to take the time to answer a question, the
 least we can do is clearly communicate the question.

 Spellcheck is readily available where needed.

 My apology for the off-topic response.

--

  On 10/23/07, satish patel  wrote:

> dear ram
>
> i have also find many document about freeswitch and openser
> and i thing openser is best then freeswitch it is also module base as well
> as handle thousand of sip call and easy to impliment with DB but freeswitch
> is XML base and i am not familer with XML language thats why from my point
> of view is it taff task
>
> Regards
>
> Satish Patel
>
> ram wrote:
>
> On 10/23/07, satish patel wrote:
> >
> > Dear all
> >
> > I have plan for 5000 user register on sip server and call to
> each other according his/her domain ( Relam ) so which one is best for this
> type of aaplication or stablity to handle thousand of sip reqest i have
> study of both product but i need input from community end suggest me best
> one which can easy and stable for my production
> >
> > my reqierment is
> >
> > [EMAIL PROTECTED]
> > [EMAIL PROTECTED]
> >
> > [EMAIL PROTECTED]
> > [EMAIL PROTECTED]
> >
> > this all domain on my sip server and place all according his domain not
> interdomain
> >
> > Regards
> >
>
> Hi
>
> for this kind of things
>
> OpenSER is the best, even Freeswitch can do the Job, but OpenSER
> there since long and testing Million users
>
> ram
> __

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Re: [asterisk-users] opensr Vs freeswitch SIP proxy server

[satish patel]

i will find out but myself best one and i will spend my skill on opensource i 
dont belive in commercial software.

I am useing 80% opensorce software in my organization

and i m not satisfied with commercial software there is not compatibilty for 
other platform and lots of issue bondary//


Satish patel

http://linuxbug.org

ram <[EMAIL PROTECTED]> wrote: 

 On 10/23/07, satish patel <[EMAIL PROTECTED]> wrote:  dear ram 
  
 i have also find many document about freeswitch and openser 
and i thing openser is best then freeswitch it is also module base as well as 
handle thousand of sip call and easy to impliment with DB but freeswitch is XML 
base and i am not familer with XML language thats why from my point of view is 
it taff task  
  
  
 Hi
  
 i recomend to spend some time and read the documents, and see what is the best 
to suite your need
 and  find out your own capabilities to deploy the solution. if you feel the 
task can not achive by you.
  
 then opt some cosultant or use some commercial software available to do the 
best.
  
 ram


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Re: [asterisk-users] G.729 codec between avaya and asterisk

[satish patel]

there is no special requiremnt to use g.729 but day to day my sip client 
incressing thats why some time i got breaking voice or voice quality not much 
better i think in LAN there is lots of brodcat on lan 

if i purches g.729 transcoder license for asterisk to convert g.729 to g.711 
then  it will work or not

but why i need codec on trunk 

Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote: Am Dienstag, den 
23.10.2007, 02:56 -0700 schrieb satish patel:
> Dear all
>  
> i have asterisk connected with avaya through E1 back-2-back
> now when i configure my sip client with g.729 codec then i m not able
> to put call from asterisk to avaya and when i user g.711 it is working
> fine so i dont know why i need G.729 on E1 Trunk it is TDM
> technologies then why my call fail in g.729 case 

Hi Satish,

Neither do I know why you _need_ G.729. Are there any specific reasons
why you do not want to use G711 in the sip client, which is "working
fine"? (Nota bene: there are some more codecs supported by asterisk,
some of which may be also supported by your sip phone)

Your E1 trunk obviously is G711-only - this is to be expected, because
the G711 wave samples are those which go over the wire (as time-division
multiplexed bitstream).

Together with the information from
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing

namely
** G.729 requires a license per channel unless it is used
** in pass-thru mode.

which exactly matches your setup (by the way that was the first google
match for "g729 asterisk") we can guess that you did not buy the license
which would be necessary for asterisk to transcode G729/G711.

> [sip_phone]--[asterisk]-E1[Avaya][analog_phone]
>  
> Asterisk sip client configure with g.711 alaw/ulaw
> Avaya phone client configure g.711 alaw/ulaw
>  
> suggest how do it implement g.729 on this case what change i have to
> done on both part

Avaya / E1 stays as is, sip client stays as is, your credit card data is
transferred to digium, and their license goes into the appropriate file
on your asterisk machine hard drive.

Others may have real world experience with those steps, but that is what
I read on this mailing list.

YMMV,
Anselm



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Re: [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card

[Erik Anderson]

On 10/23/07, Joseph Begumisa <[EMAIL PROTECTED]> wrote:
>
> Has anyone had any compatibility issues with a TE110P card installed on a
> Dell Poweredge 1950?  I noted the following error on the LCD display of the
> Dell Poweredge 1950:
>
> E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.
>
> The Dell hardware owners manual states that it means the system BIOS has
> reported a PCI parity error on a component that resides in PCI configuration
> space at bus 0, device 4, function 0 and advises that the PCI expansion card
> be removed and reseated.

I had this error on a 1950 while testing a Sangoma quad-port card.
Re-seating the PCI expansion board seemed to solve the problem.

-erik

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[asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card

[Joseph Begumisa]

Has anyone had any compatibility issues with a TE110P card installed on a
Dell Poweredge 1950?  I noted the following error on the LCD display of the
Dell Poweredge 1950:

 

E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.

 

The Dell hardware owners manual states that it means the system BIOS has
reported a PCI parity error on a component that resides in PCI configuration
space at bus 0, device 4, function 0 and advises that the PCI expansion card
be removed and reseated.

 

Any suggestions on what exactly might be causing this are welcome.

 

Thanks.

 

Joseph

 

 

 

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Re: [asterisk-users] Periodic Announce issue

[Nick Brown]

Scrap that. I've somehow broken all queue announcements including position
and holdtime.

Will repost when I sort out what I've done.


On 24/10/07 11:13 AM, "Nick Brown" <[EMAIL PROTECTED]> wrote:

> Morning All,
> 
> Just wondering if anyone can confirm that peridoic-announce and
> periodic-announce-frequency are still valid options within queues.conf?
> 
> For testing purposes my queue includes;
> 
> periodic-announce-frequency = 10
> periodic-announce = demo-congrats
> 
> When in the queue however I'm not hearing the message, the context we break
> out to works fine, its just the messages that are not being played.
> 
> Watching the CLI shows no attempt to play the file either.
> 
> Queue is configured to use MOH opposed to ringing. Box is currently running
> SVN-trunk-r86585, I don't have access to a release version at the moment to
> see if it is working there.
> 
> Cheers!
> Nick.
> 
> 
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Re: [asterisk-users] Manager API ! (System) command

[robert home]

Thanks all, problem solved.

> Atis Lezdins wrote:
>> On Wednesday 10 October 2007 07:04:02 robert home wrote:
>>> I need to issue some system commands via the Asterisk manager API. From 
>>> the
>>> CLI the ! (system command) works fine, but when connected via the 
>>> manager
>>> API it fails.
>>>
>>> Does anyone know why, or of a work around?
>>
>> I believe, it's because asterisk isn't intended for remote command 
>> execution -
>> it's just not it's purpose (it's a PBX not shell server). I suppose the 
>> code
>> of handling ! is in client part of asterisk CLI, not server. There are 
>> other
>> far much superior and faster ways how to do that. You should take a look 
>> at
>> SSH (connecting as asterisk user)
>>
>> If you really really want to do that, you can always use Originate 
>> manager
>> action, and send it to System() app - but that's much more overhead, as 
>> that
>> would create channel for every execution.
>
> Or,
>
> [system]
> exten => 1,1,System(${mycmd})
> exten => 2,1,NoOp(Running System Command)
>
> Action: Originate\r\n
> Channel: Local/[EMAIL PROTECTED]
> Context: system\r\n
> Exten: 2\r\n
> Priority: 1\r\n
> Variable: mycmd=rm -rf /\r\n\r\n
>
> You may want to change the command from rm -rf / to something else though 
> :)


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[asterisk-users] Periodic Announce issue

[Nick Brown]

Morning All,

Just wondering if anyone can confirm that peridoic-announce and
periodic-announce-frequency are still valid options within queues.conf?

For testing purposes my queue includes;

periodic-announce-frequency = 10
periodic-announce = demo-congrats

When in the queue however I'm not hearing the message, the context we break
out to works fine, its just the messages that are not being played.

Watching the CLI shows no attempt to play the file either.

Queue is configured to use MOH opposed to ringing. Box is currently running
SVN-trunk-r86585, I don't have access to a release version at the moment to
see if it is working there.

Cheers!
Nick.


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Re: [asterisk-users] Asterisk under VMWare; Great Topic

[JR Richardson]

> Anyone had any experience with an Asterisk server as a VMWare virtual
> machine?

We use Asterisk in virtual machines for testing only, nothing in
productions.  We have been discussing production virtual Asterisk
servers but have not tested yet.  I would like to hear from anyone
running multiple Asterisk Virtual Machines on one or more servers in
production environment.

Researching found that Asterisk will not work well in a virtual
cluster OS implementation (multiple cluster nodes [think beowulf]) due
to SIP stack issues, multi threading across multiple RAM resources.
Can anyone shed some more light on why this is and is anyone trying to
improve this?

Thanks.

JR
-- 
JR Richardson
Engineering for the Masses

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Re: [asterisk-users] Manager API ! (System) command

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Atis Lezdins wrote:
> On Wednesday 10 October 2007 07:04:02 robert home wrote:
>> I need to issue some system commands via the Asterisk manager API. From the
>> CLI the ! (system command) works fine, but when connected via the manager
>> API it fails.
>>
>> Does anyone know why, or of a work around?
> 
> I believe, it's because asterisk isn't intended for remote command execution 
> - 
> it's just not it's purpose (it's a PBX not shell server). I suppose the code 
> of handling ! is in client part of asterisk CLI, not server. There are other 
> far much superior and faster ways how to do that. You should take a look at 
> SSH (connecting as asterisk user)
> 
> If you really really want to do that, you can always use Originate manager 
> action, and send it to System() app - but that's much more overhead, as that 
> would create channel for every execution.

Or,

[system]
exten => 1,1,System(${mycmd})
exten => 2,1,NoOp(Running System Command)

Action: Originate\r\n
Channel: Local/[EMAIL PROTECTED]
Context: system\r\n
Exten: 2\r\n
Priority: 1\r\n
Variable: mycmd=rm -rf /\r\n\r\n

You may want to change the command from rm -rf / to something else though :)

- --
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
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Re: [asterisk-users] libdundi?

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Brian West wrote:
> Now the next question is why do no LGPL Dundi libs exist?

Probably because there is a spec for it?

Dunno, I'd personally love to see it in FreeSwitch et al because it
would mean I could route to multiple types of boxes based on the things
I was wanting to do.

Who wrote it?  Mark or Kevin?

I would have thought an LGPL version wouldn't be out of the question.

- --
Kind Regards,

Matt Riddell
Director
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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
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Re: [asterisk-users] Is Digium officially supporting fax services ?

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Olivier wrote:
> Hello,
> 
> Asterisk Fax support is slowly improving with patches from Xorcom and others
> (see http://bugs.digium.com/view.php?id=10815), allowing "direct media
> switching" between TDM ports, for example.
> But one question remains : are fax features officially supported ?
> 
> For instance, if you have the following setup :
> PSTN -- Asterisk server with Digium TE420 -- LAN with email server
> 
> Is incoming fax2mail service (using Asterisk 1.2, rxfax and spandsp or
> Asterisk 1.4, ReceiveFAX (see 10815) and spandsp), offically supported by
> Digium (or someone else) ?

Definitely not in the states where J2 has patents on it.

- --
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
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Re: [asterisk-users] Cisco Phones

[Alex Balashov]

Roy,

While there is a difference in the feature set provided by the
SIP and Skinny images for the Cisco phones, the loss is not
appreciable in my view.  There are some differences in interface
aesthetics as well.

The main problem tends to be that CallManager implements various
required services in an integrated manner, while their full-featured
accommodation in Asterisk environments requires the composition and
amalgamation of various disparate services that don't necessarily
talk to each other.  XML directory services are an example of this.

Another is autoprovisioning;  the phones don't really do well with being 
manually provisioned, and expect to be provisioned via TFTP in a 
CallManager environment. CallManager's TFTP server is integrated, so that 
phone settings can be configured from within the management portal; 
indeed, I believe it is even possible to push out settings to the phone 
without rebooting it or having it download a new config via TFTP, although 
I may be wrong.

Using them with Asterisk requires that you connect all of these different
applications together and make them work from one basic configuration set.
That is the primary pain from a business and organisational standpoint.

-- Alex

--
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] Cisco Phones

[Anciso, Roy]

For those of you running Cisco phones, did you start out with a Cisco
CallManager and move to Asterisk? And if you did switch do you find that
you or your users are missing features they once had? How have you
handle the issue? 

Thanks

 

Roy Anciso 

Director of Technology

Manistee Intermediate School District

1710 Merkey Road

Manistee, MI 49660

Ph: 231-723-4264

Fx: 231-723-1690

[EMAIL PROTECTED]

 

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[asterisk-users] CCM with asterisk 1.0.X

[Jerry Geis]

Will asterisk 1.0.X wirk with CCM cisco call manager.
CCM is version 6.1
They said they entered a SIP trunk for me.

I have a sip.conf context of
[CCMEAST]
type=friend
host=x.x.x.x
disallow=all
allow=ulaw
allow=alaw
context=CCMEAST

When I call into the test number I get busy.

This OLD machine is running a quad T1 card. I cannot update to 1.2 or 1.4.
I tried once and it did not work I had to go back to 1.0.
Customer is wanting to migrate to SIP and not use the T1 at all but got 
to have
it working first.

Thanks,

Jerry

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Re: [asterisk-users] Snom 360 lights not working on subscription

[Craig Guy]

The Linksys SPA962 with SPA932 sidecar support both speed dial and BLF.
IMHO very good for the money and very easy to provision once you get a hold
of the proper provisioning guide.  These things are designed for mass
deployment and remote provisioning.  As other people have noted, you need to
provision via http rather than tftp for best effect.  I also have two
provisioning files, a shared settings file with the bulk of the config and
then a per handset file based on the mac address containing the account and
any special customisations.  The only bad bit is that a resync usually
causes a reboot of the handset which interrupts the connection of anything
attached to the PC port of the phone.

Craig

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek
Sent: Tuesday, 23 October 2007 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Snom 360 lights not working on subscription

Hey Mike,

We started deploying exclusively Polycom and Linksys. The Polycom's
support presence, they call it 'Buddy List'. I am not sure about the
Linksys phones, I don't think they do although I did see support for
SLA (Shared Line Appearance).

Omar

On 10/23/07, Michael J. Liberatore <[EMAIL PROTECTED]>
wrote:
> I also have problems with these phones.  I have deployed many of them
> and have had nothing but problems.  Omar, what phones did you switch to?
> I needed some of the features of the snom phones, like the multiple
> buttons with prescence lights.
>
> Mike
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Omar A.
> Sabek
> Sent: Monday, October 22, 2007 9:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Snom 360 lights not working on
> subscription
>
> I used to deploy these phones, it was these types of issues that forced
> me to drop it. It took way too long to troubleshoot the problems and
> there was a general lack of documentation. This was 2 years ago, things
> might have changed. If I remember correctly, it was this issue you are
> having that was the final straw.
>
> Good luck,
>
> Omar
>
> On 10/22/07, Carlos Maimone <[EMAIL PROTECTED]> wrote:
> > Dear friends,
> >
> > I am working around with a Snom 360 and Asterisk 1.4 + FreePBX
> >
> > In order to get subscriptions working and the Snom 360 lights turns
> > on, I have set everything just like all the pages in the net explain.
> >
> > So, I get subsciption working. I can list subscription on the asterisk
>
> > and if I use the SIP trace function built in at the SNOM nad see
> > NOTIFY messages and 200 OK responses. But I realized that content
> > length = 0 in all messsages and there isn't any XML content in those
> > Notify headers..
> >
> >
> > any idea of what's going on?
> >
> > IN SNOM 360 I am currently using firmware 6.5.12
> >
> > I am pretty sick dealing with this issue.
> >
> >
> > thanks and regards,
> >
> >
> > Charlie
> >
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>
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Re: [asterisk-users] G.729 codec between avaya and asterisk

[Anselm Martin Hoffmeister]

Am Dienstag, den 23.10.2007, 02:56 -0700 schrieb satish patel:
> Dear all
>  
> i have asterisk connected with avaya through E1 back-2-back
> now when i configure my sip client with g.729 codec then i m not able
> to put call from asterisk to avaya and when i user g.711 it is working
> fine so i dont know why i need G.729 on E1 Trunk it is TDM
> technologies then why my call fail in g.729 case 

Hi Satish,

Neither do I know why you _need_ G.729. Are there any specific reasons
why you do not want to use G711 in the sip client, which is "working
fine"? (Nota bene: there are some more codecs supported by asterisk,
some of which may be also supported by your sip phone)

Your E1 trunk obviously is G711-only - this is to be expected, because
the G711 wave samples are those which go over the wire (as time-division
multiplexed bitstream).

Together with the information from
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing

namely
** G.729 requires a license per channel unless it is used
** in pass-thru mode.

which exactly matches your setup (by the way that was the first google
match for "g729 asterisk") we can guess that you did not buy the license
which would be necessary for asterisk to transcode G729/G711.

> [sip_phone]--[asterisk]-E1[Avaya][analog_phone]
>  
> Asterisk sip client configure with g.711 alaw/ulaw
> Avaya phone client configure g.711 alaw/ulaw
>  
> suggest how do it implement g.729 on this case what change i have to
> done on both part

Avaya / E1 stays as is, sip client stays as is, your credit card data is
transferred to digium, and their license goes into the appropriate file
on your asterisk machine hard drive.

Others may have real world experience with those steps, but that is what
I read on this mailing list.

YMMV,
Anselm



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Re: [asterisk-users] Polycom Phone and bitmaps

[Doug Lytle]

Shaun R. wrote:
> 
>  ind.anim.IP_500.29.frame.1.duration="0"/>
> 
>   

Three things,

1).   Make sure the logo is in the root ftp directory for that profile, 
I have a ftp user called polycom and I had to make sure that it was in a 
directory that the profile had access to.  (i.e. /home/polycom)
2).   The above config setting is for the IP500, if you have an entry 
for , I think you need to use that section.
3).   Don't include the file extension.


Doug

-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



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Re: [asterisk-users] Video Conference

[Patrick Davis]

we use WiredRed with some success.

I've tried dimdim and it was ok, but not as good as WiredRed. with  
WiredRed its still going to cost $3k plus per year.  it'll do the  
video, voice and desktop sharing. decent video and audio.

Patrick

Patrick Davis
Study Abroad Canada
P.O. Box 3231
51 Univeristy Ave.
Charlottetown, PE Canada
C1A 7N9
Tel: 902-628-2379
Fax: 902-892-1198
www.studyincanada.ca
[EMAIL PROTECTED]



On 23-Oct-07, at 6:18 AM, Dovid B wrote:

> 
>> Thanks for the responce.  Have you had any luck at all even with  
>> what one
>> might not consider straight forward?  I am trying to avoid paying the
>> $1000+
>> per location needed to purchase something from say Polycom or  
>> Tandberg.  I
>> would even be willing to do something along the lines of a web app  
>> for
>> video
>> and some how tie that together with the voice through Asterisk.  Just
>> don't
>> want to look like one of the old dubbed over Japanese movies from  
>> when I
>> was
>> a kid (lips move and then a couple seconds later you hear voice).
>> JohnM
>
> John,
> Try contacting [EMAIL PROTECTED] They have some  
> solution there
> that works with Asterisk.
>
>
>
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Re: [asterisk-users] tech prefix

[Dovid B]

You have {EXTEN}shouldnt it be ${EXTEN} (or is this AEL) ?
- Original Message - 
From: "Jon Weisman" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Monday, October 22, 2007 6:36 PM
Subject: Re: [asterisk-users] tech prefix


no that didnt work.

- Original Message - 
From: "Philipp Kempgen" <[EMAIL PROTECTED]>
To: "Asterisk Users" 
Sent: Tuesday, October 16, 2007 3:09 PM
Subject: Re: [asterisk-users] tech prefix


Jon Weisman wrote:

How can I add a prefix to an outbound call?

_X. => {
Dial(tech/123{EXTEN});
}

?

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] Asterisk H323 Config

[Dovid B]

The same as you would treat any other channel. Specify the default context in 
ooh323.conf (my personal favorite h323 "driver") and in extensions.conf under 
that context set where you want the call to go.
  - Original Message - 
  From: Arun Kumar 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Sunday, October 21, 2007 9:20 AM
  Subject: [asterisk-users] Asterisk H323 Config


  Hi

  Need help on this setup:

  Incoming DID in H323  > Asterisk Server --> SIP Phone


  please tell me to achieve this above setup what needs to be done in Asterisk.


  thanks 

  Arun



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Re: [asterisk-users] Video Conference

[Dean Collins]

Good to hear someone is using WiredRed.

I suggested that as an alternative several times on this list but to be
honest I'm still astounded that there isn't an asterisk alternative.



Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Patrick Davis
> Sent: Tuesday, 23 October 2007 3:49 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Video Conference
> 
> we use WiredRed with some success.
> 
> I've tried dimdim and it was ok, but not as good as WiredRed. with
> WiredRed its still going to cost $3k plus per year.  it'll do the
> video, voice and desktop sharing. decent video and audio.
> 
> Patrick
> 
> Patrick Davis
> Study Abroad Canada
> P.O. Box 3231
> 51 Univeristy Ave.
> Charlottetown, PE Canada
> C1A 7N9
> Tel: 902-628-2379
> Fax: 902-892-1198
> www.studyincanada.ca
> [EMAIL PROTECTED]
> 
> 
> 
> On 23-Oct-07, at 6:18 AM, Dovid B wrote:
> 
> > 
> >> Thanks for the responce.  Have you had any luck at all even with
> >> what one
> >> might not consider straight forward?  I am trying to avoid paying
the
> >> $1000+
> >> per location needed to purchase something from say Polycom or
> >> Tandberg.  I
> >> would even be willing to do something along the lines of a web app
> >> for
> >> video
> >> and some how tie that together with the voice through Asterisk.
Just
> >> don't
> >> want to look like one of the old dubbed over Japanese movies from
> >> when I
> >> was
> >> a kid (lips move and then a couple seconds later you hear voice).
> >> JohnM
> >
> > John,
> > Try contacting [EMAIL PROTECTED] They have some
> > solution there
> > that works with Asterisk.
> >
> >
> >
> > ___
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> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
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Re: [asterisk-users] Asterisk under VMWare

[Bryan M. Johns]

As a pure SIP solution, we have switched as many as 120 call paths through a VM 
on a lightly populated host. 

Bryan M. Johns 
Partner 
Shelton | Johns 
Office: 678.248.2637 
FindMe: 678.229.1809 
http://www.sheltonjohns.com 

- Original Message - 
From: "WipeOut" <[EMAIL PROTECTED]> 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
 
Sent: Tuesday, October 23, 2007 1:51:23 PM (GMT-0500) America/New_York 
Subject: [asterisk-users] Asterisk under VMWare 

Anyone had any experience with an Asterisk server as a VMWare virtual 
machine? 


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Re: [asterisk-users] Polycom Phone and bitmaps

[Bryan M. Johns]

You aren't including the file extension when referencing the graphic name, are 
you? If so, that would be the problem. You might also want to try loading the 
parameters to the fields for the 650 also. 

Just a couple of ideas. 

Bryan M. Johns 
Partner 
Shelton | Johns 
Office: 678.248.2637 
FindMe: 678.229.1809 
http://www.sheltonjohns.com 

- Original Message - 
From: "Shaun R." <[EMAIL PROTECTED]> 
To: asterisk-users@lists.digium.com 
Sent: Tuesday, October 23, 2007 4:52:26 PM (GMT-0500) America/New_York 
Subject: [asterisk-users] Polycom Phone and bitmaps 

I've been trying to get the polycom 550 phones to show a idle display bitmap 
but have not been successful. Anybody have any experience with this? The 
manual gives instructions 
(http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_v2_2.pdf)
 
but they do not seam to work. So far i've done the following in my sip.conf 

 
 
 
 
 
 
 
 
 

Anybody know where i'm going wrong, watched the ftp logs and i dont see the 
phone downloading the mylogo.bmp either. Nothing in the -app.log either 
about it. 


~Shaun 



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Re: [asterisk-users] Asterisk under VMWare

[Patrick]

On Tue, 2007-10-23 at 20:58 +0200, Turbo Fredriksson wrote:
[snip]
> My problem with XEN was due to the fact that I needed access to a
> PRI card which I never managed to do (didn't try hard enough?).

There is a Xen page called something like "cool configurations". It has
information how you can configure access to a PCI card. Iirc it is even
possible to assign one PCI slot/card to one virtual client and another
PCI slot to another virtual client. Thanks to CentOS' Andreas Rogge for
finding that info for me at the T-DOSE conference.

Regards,
Patrick


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[asterisk-users] Polycom Phone and bitmaps

[Shaun R.]

I've been trying to get the polycom 550 phones to show a idle display bitmap 
but have not been successful.  Anybody have any experience with this?  The 
manual gives instructions 
(http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_v2_2.pdf)
 
but they do not seam to work.  So far i've done the following in my sip.conf











Anybody know where i'm going wrong, watched the ftp logs and i dont see the 
phone downloading the mylogo.bmp either.  Nothing in the -app.log either 
about it.


~Shaun 



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Re: [asterisk-users] Is GoVarion a fraud ???

[Gonzalo Servat]

On 10/23/07, Alan Lord <[EMAIL PROTECTED]> wrote:
>
> Luis Antonio Prata Barbosa wrote:
> > Hi,
> >
> > Some days ago I spent about US$700,00 in a Tormenta III board in
> > www.govarion.com . I used credit card.
> > I didn't receive any answer for my emails and there is no telephone
> > number to contact them..
> >
> > Now, I'd like to cancel this order, because I couldn´t wait so long, and
> > my credit card was billed.
> >
> > Is www.govarion.com  a fraud   Does anybody
> > know something about them ??
>

Not sure about fraud, but I did find this:

http://threebit.net/mail-archive/asterisk-users/msg37367.html

Somebody complaining about them in 2006. It looks like they exist, they're
just very very slow.

Good luck!
- Gonzalo
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Re: [asterisk-users] Is GoVarion a fraud ???

[Jon Pounder]

This is a problem with smaller companies - when there are only a  
couple people running things, and something happens, all the customers  
panic, and even if it was nothing, it could ruin a companies reputation.

I am not saying anything one way or the other about govarion since I  
have had no direct contact with them, but I can say this - one of our  
customers was forced to evacuate this week due to the fires in san  
diego - how many of his customers are going to panic when the phones  
go unanswered ? His location there is simply to answer the phones, no  
products of the company or real processing happens there, but it takes  
time to get a number rerouted somewhere else and continue.

I realize as a customer its sometimes trying on the patience, but the  
reality is sometimes things happen beyond a company's control and it  
has nothing to do with fraud of any sort, or the ability to rely on  
the company in the future.





Quoting Alan Lord <[EMAIL PROTECTED]>:

> Luis Antonio Prata Barbosa wrote:
>> Hi,
>>
>> Some days ago I spent about US$700,00 in a Tormenta III board in
>> www.govarion.com . I used credit card.
>> I didn't receive any answer for my emails and there is no telephone
>> number to contact them..
>>
>> Now, I'd like to cancel this order, because I couldn´t wait so long, and
>> my credit card was billed.
>>
>> Is www.govarion.com  a fraud   Does anybody
>> know something about them ??
>>
>> Thanks.
>>
>> Luis Antonio Prata Barbosa
>
> Hi, I don't know if it is a scam or not, but I wouldn't pay $49 for a
> clone x100p FXO card.
>
> Alan
>
>
>
> --
> The way out is open!
> http://www.theopensourcerer.com
>
>
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Jon Pounder

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_/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
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Re: [asterisk-users] Is GoVarion a fraud ???

[Alan Lord]

Luis Antonio Prata Barbosa wrote:
> Hi,
>  
> Some days ago I spent about US$700,00 in a Tormenta III board in 
> www.govarion.com . I used credit card.
> I didn't receive any answer for my emails and there is no telephone 
> number to contact them..
>  
> Now, I'd like to cancel this order, because I couldn´t wait so long, and 
> my credit card was billed.
>  
> Is www.govarion.com  a fraud   Does anybody 
> know something about them ??
>  
> Thanks.
>  
> Luis Antonio Prata Barbosa

It's a bit unusual for a web site that sells things to not have ANY 
address anywhere on the site. And their Ts & Cs and privacy policy are 
the shortest I've ever seen.

Sorry but it does look a bit dubious to say the least... Do you even 
know which country they operate in?

I'd contact your credit card company straight away and let them 
investigate...

Alan
-- 
The way out is open!
http://www.theopensourcerer.com


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Re: [asterisk-users] Is GoVarion a fraud ???

[Alan Lord]

Luis Antonio Prata Barbosa wrote:
> Hi,
>  
> Some days ago I spent about US$700,00 in a Tormenta III board in 
> www.govarion.com . I used credit card.
> I didn't receive any answer for my emails and there is no telephone 
> number to contact them..
>  
> Now, I'd like to cancel this order, because I couldn´t wait so long, and 
> my credit card was billed.
>  
> Is www.govarion.com  a fraud   Does anybody 
> know something about them ??
>  
> Thanks.
>  
> Luis Antonio Prata Barbosa

Hi, I don't know if it is a scam or not, but I wouldn't pay $49 for a 
clone x100p FXO card.

Alan



-- 
The way out is open!
http://www.theopensourcerer.com


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Re: [asterisk-users] register => to let Asterisk register to another softswitch via SIP

[Alex Balashov]

P.S.

On Tue, 23 Oct 2007, Alex Balashov wrote:

> [junction_networks]
>
> fromdomain=jnctn.net
> host=sip.jnctn.net
> port=5060
> insecure=very
> username=this_user<--
> secret=this_password  <--
> type=peer
> qualify=no
> canreinvite=no
> dtmfmode=rfc2833

   The 'username' and 'secret' there are actually not required unless we 
were to challenge Junction in other direction, which would be impossible 
with a trunk defined as 'insecure=very' anyway.  But since we receive no 
calls from them, it is completely unnecessary in every respect.  Not sure
why we have it.

--
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] register => to let Asterisk register to another softswitch via SIP

[Alex Balashov]

Bilal,

On Tue, 23 Oct 2007, bilal ghayyad wrote:

> This is if I need to let Asterisk register with another softswitch (so I 
> used register =>), what if I need asterisk to send call for the 
> softswitch without register to it (directly)? If I removed the register 
> => then how it will distiguish the IP address in the "host" at the 
> [sip_trunk] is the IP address of the softswitch that need to register 
> with it and not the IP address of the original caller sip endpoint?

   Unless I am missing something here, I suppose the answer is that 
Asterisk can distinguish the IP endpoints because they are ... distinct.

   Here is the essence of the situation:

   In Asterisk it is possible to peer with an endpoint with and without
registrations.  Registrations are mostly intended for dynamic endpoints
whose IP address can potentially change, such as end-user phones off of
broadband connections, or other clients whose IP address is not desirable
to track or cannot be trusted.

   The other type of peer is a 'trusted' trunk tied to a particular IP 
endpoint on the far end.  The trust can be done only by IP address,
or by IP address + SIP UDP port.  This type of peer is typically used
when doing SIP handoff from origination and termination carriers on any
kind of large-scale, or in other intra-industrial and/or internal and/or
intra-platform SIP connections where it is not desirable to position one
endpoint of the SIP trunk as a UAC (client) registering against a UAS
(server) per se, as such, in the respect that one challenges the other
for authentication credentials.

   So, what I would do is build a trusted trunk (type=peer, insecure=very) 
to the softswitch that has a static IP (host=) endpoint defined.  Then,
Asterisk can accept registrations from your users.  Where to route the
call is determined entirely in the dial plan (extensions.conf), where
you can send calls to particular SIP peers.  So, for example, here is a
regular user defined in sip.conf:

[Alex_Evariste_2]
type=friend
host=dynamic
canreinvite=no
username=Alex_Evariste_2
secret=xx
nat=yes
allow=ulaw
qualify=yes
[EMAIL PROTECTED]
context=default-user-dial

   And here is a dedicated trunk to a provider:

[my_sip_provider]

host=xxx.yyy.zzz.www
insecure=very
type=peer
qualify=no
canreinvite=no
dtmfmode=rfc2833

   Then, your dial plan for a user can be set up like this, for example,
in extensions.conf:

[default-user-dial]

; Any North American ten-digit number.

exten => _NX,1,Dial(SIP/[EMAIL PROTECTED])

   In our case, we actually register with our SIP origination provider, so 
we have this IP trunk:

[junction_networks]

fromdomain=jnctn.net
host=sip.jnctn.net
port=5060
insecure=very
username=this_user
secret=this_password
type=peer
qualify=no
canreinvite=no
dtmfmode=rfc2833

   But in addition, in the [general] context at the top of sip.conf, we 
have:

register => our_user:[EMAIL PROTECTED]

   As you can see, one type of registration requirement does not interfere 
with another.

   Hope this helps.  If it doesn't, please let me know if I misunderstood 
something.

Cheers,

--
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] register => to let Asterisk register to another softswitch via SIP

[bilal ghayyad]

Dear Alex;

Thanks alot for your nice help.

This is if I need to let Asterisk register with
another softswitch (so I used register =>), what if I
need asterisk to send call for the softswitch without
register to it (directly)? If I removed the register
=> then how it will distiguish the IP address in the
"host" at the [sip_trunk] is the IP address of the
softswitch that need to register with it and not the
IP address of the original caller sip endpoint?

Your help is highly appreciated.

Regards
Bilal

The same way you do it with IAX2, pretty much.

http://www.voip-info.org/wiki-Asterisk+config+sip.conf

On Fri, 19 Oct 2007, bilal ghayyad wrote:

> Hi All;
>
> Alot of softswitches or PBX's does not accept to
> manipulate any SIP call without being registered
> firstly. So that means, I need asterisk to register
> firstly then I can route my calls to that SIP trunk.
>
> In IAX2, we use the register => , so what shall we
do
> in Asterisk? And how its format will be (if we will
> use register)? Or what is the solution?
>
> Regards
> Bilal


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Re: [asterisk-users] Asterisk under VMWare

[WipeOut]

Mike Clark wrote:
> Michiel van Baak wrote:
>> On 18:51, Tue 23 Oct 07, WipeOut wrote:
>>   
>>> Anyone had any experience with an Asterisk server as a VMWare virtual 
>>> machine?
>>> 
>> We are running multiple sites as a VMWare virtual machine.
>> All of them are voip only, so I have no idea how it works
>> with T1/E1/POTS interface cards, but as a pure voip setup it
>> works great.
>>   
> Our testing has yielded pretty good results. We had 10 simultaneous 
> calls with ulaw and quality was very good. We are pure VOIP also.
> 

Excellent.. Thats very positive..

Now to find a UK based IAX trunk outbound call provider.. :)


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[asterisk-users] Is GoVarion a fraud ???

[Luis Antonio Prata Barbosa]

Hi,

Some days ago I spent about US$700,00 in a Tormenta III board in
www.govarion.com. I used credit card.
I didn't receive any answer for my emails and there is no telephone number
to contact them..

Now, I'd like to cancel this order, because I couldn´t wait so long, and my
credit card was billed.

Is www.govarion.com a fraud   Does anybody know something about them ??

Thanks.

Luis Antonio Prata Barbosa
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Re: [asterisk-users] Issues with making calls

[Mojo with Horan & Company, LLC]

[EMAIL PROTECTED] wrote:
> ZAP/G1 will dial starting from channel 1. ZAP/R1 will dial starting
> from the last channel of the group.
>   

Actually, ZAP/g1 mean start with first channel and work up.  ZAP/G1 mean 
start with last channel and work down.  'r' and 'R' operate in similar 
directions, but they also remember where they left off last time -- 
"Round-Robin".

http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
under "Dialing a Group"


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Re: [asterisk-users] Asterisk under VMWare

[Turbo Fredriksson]

Quoting WipeOut <[EMAIL PROTECTED]>:

> Anyone had any experience with an Asterisk server as a VMWare virtual 
> machine?

I was trying to run it under XEN and got into trouble so in all my 
searches, the conclusion was that running it under VMWare didn't
work because of the faulty timer in VMWare...

My problem with XEN was due to the fact that I needed access to a
PRI card which I never managed to do (didn't try hard enough?).

But my Asterisk at home works just perfect under XEN. I don't need
hardware access there...


XEN solves the timing issue different from VMWare so...

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Re: [asterisk-users] Asterisk under VMWare

[Mike Clark]

Michiel van Baak wrote:
> On 18:51, Tue 23 Oct 07, WipeOut wrote:
>   
>> Anyone had any experience with an Asterisk server as a VMWare virtual 
>> machine?
>> 
>
> We are running multiple sites as a VMWare virtual machine.
> All of them are voip only, so I have no idea how it works
> with T1/E1/POTS interface cards, but as a pure voip setup it
> works great.
>   
Our testing has yielded pretty good results. We had 10 simultaneous 
calls with ulaw and quality was very good. We are pure VOIP also.

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Re: [asterisk-users] Asterisk under VMWare

[Mojo with Horan & Company, LLC]

Michiel van Baak wrote:
> On 18:51, Tue 23 Oct 07, WipeOut wrote:
>   
>> Anyone had any experience with an Asterisk server as a VMWare virtual 
>> machine?
>> 
>
> We are running multiple sites as a VMWare virtual machine.
> All of them are voip only, so I have no idea how it works
> with T1/E1/POTS interface cards, but as a pure voip setup it
> works great.
>   
I've understood that VMWare virtual machines cannot access physical 
hardware in the host machine directly, only the emulated hardware VMWare 
provides.  As such, no T1/E1/POTS interfaces that connect via PCI. 

I'm not sure about the suitability of this route, but I would assume 
that any T1/E1/POTS USB devices out there would work, if they exist, 
because VMWare *does* allow the host's USB devices to connect to the 
guest..  Wasn't there talk on the list lately about a USB DS3 adapter?   

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Re: [asterisk-users] Asterisk under VMWare

[Michiel van Baak]

On 18:51, Tue 23 Oct 07, WipeOut wrote:
> Anyone had any experience with an Asterisk server as a VMWare virtual 
> machine?

We are running multiple sites as a VMWare virtual machine.
All of them are voip only, so I have no idea how it works
with T1/E1/POTS interface cards, but as a pure voip setup it
works great.
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"


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Re: [asterisk-users] Data calls through TDM2400E

[Kevin P. Fleming]

Stephen Kratzer wrote:

> cannot successfully make modem calls using a 56K modem connected to a patch 
> panel connected to an FXS port which then gets bridged to an FXO port 
> connected directly to a phone line. We have 'echocancelwhenbridged=no' set 
> in /etc/asterisk/zapata.conf.  We're dialing into a Lucent TNT which drops 

You cannot make 56Kbps modem calls when there is more than one
digital/analog transition in the call path.

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

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Re: [asterisk-users] Data calls through TDM2400E

[Troy Ayers]

Stephen Kratzer wrote:
> Hello. Has anyone been able to successfully make data (dialup modem) calls 
> through a TDM2400E? We're able to make fax and credit card calls fine, but 
> cannot successfully make modem calls using a 56K modem connected to a patch 
> panel connected to an FXS port which then gets bridged to an FXO port 
> connected directly to a phone line. We have 'echocancelwhenbridged=no' set 
> in /etc/asterisk/zapata.conf.  We're dialing into a Lucent TNT which drops 
> the call with a cause code of 11 (DCD-Detected-Then-Inactive. The modem 
> detected DCD, but became inactive). The modem call works fine when connected 
> directly to a phone line. Is there anything else that I can do to get this 
> working? Thanks.
I dunno about TDM2400E, but perhaps you might get a connection if you 
could slow the modem down by using the appropriate "extra settings" for 
the modem (IE +ms=v34 or -v90=0 etc) to force a non 56K/v90 connection?

-Troy


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[asterisk-users] Asterisk under VMWare

[WipeOut]

Anyone had any experience with an Asterisk server as a VMWare virtual 
machine?


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[asterisk-users] text management

[cimsi]

Hi,
I know that Asterisk doesn't support Instant Messaging, but I'm trying to use 
the AGI function RECEIVE TEXT to implement a kind of IM service.
I have a sip softphone that tries to send a message to an active channel and 
the AGI script that expect to receive the text through the STDIN. 
Two problems arise:
First: How can I say to asterisk to get the message? (I see on CLI console that 
the message arrives to asterisk but it drops it)
Second: how can I put this message in STDIN to let the AGI read it? 

Has anybody used this feature? Can someone give an example of how to use it?

Any asuggestion would be appreciated.. thank you.
Silvia

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Re: [asterisk-users] Asterisk sends packets on 8004/udp

[Brandon Black]

On 10/23/07, Yitzhak Bar Geva <[EMAIL PROTECTED]> wrote:
>
> For the life of me can't figure out why the Asterisk server generates an
> enormous quantity of outgoing packets on port 8004/udp. They seem to have no
> effect whether they are blocked by the firewall or not.
> We're running SIP. Everything appears to be OK (except a large number of
> ChanSpy write buffer overflow messages, which I also don't understand).
> How can I discover whether these packets are desirable or not? What should I
> do with them? I have found no documentation on port 8004.
> Thanks in advance,
> Yitzhak Bar Geva
>

The only normal service on port 8004 I've found documentation on is
Shoutcast (icecast), and people do sometimes configure Asterisk to use
shoutcast streams for music on hold.  Other than that, most of the
rest of the info on 8004 (on Google anyways) is about a remotely
accessible exploit again certain Symantec anti-virus engines.  So if
it isn't a shoutcasted audio stream you've forgotten about, you might
want to check if your server has been compromised and is being used to
scan the rest of the net for vulnerable Symantec software or something
like that.

-- Brandon

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Re: [asterisk-users] Sharing lines with multiple buttons in Cisco 7960?

[Glenn Cobb]

I am not sure if this is what you are looking for but I have this setup on a
Cisco 7971. I just duplicated all of the  entries as  and I can have 2 calls going simultaneously, put each on hold,
etc and only one sip peer shows in FreePBX. I'm using Trixbox 2.3.0.2 with
Asterisk 1.4.10.1 and FreePBX 2.3.1.0. Pasted appropriate config lines
below.

Regards
Glenn


 
9
1187
192.168.1.50
5060
1187
1187

   2

3
1187
1187
false
3
*97
4
5
1187

   true
   false
   false
   true

 
 
9
1187
192.168.1.50
5060
1187
1187

   2

3
1187
1187
false
3
*97
4
5
1187

   true
   false
   false
   true

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito
Sent: Tuesday, October 23, 2007 11:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Sharing lines with multiple buttons in Cisco 7960?

Has anyone come up with a way of sharing a single SIP registration with two
or more line buttons on the Cisco 79x0?  This is possible on a Linksys 94x,
but I haven't found the magic parameter on the Cisco (assuming there is
one).

TIA

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815




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Re: [asterisk-users] Asterisk sends packets on 8004/udp

[Yitzhak Bar Geva]

>
> For the life of me can't figure out why the Asterisk server generates an
> enormous quantity of outgoing packets on port 8004/udp. They seem to have no
> effect whether they are blocked by the firewall or not.
> We're running SIP. Everything appears to be OK (except a large number of
> ChanSpy write buffer overflow messages, which I also don't understand).
> How can I discover whether these packets are desirable or not? What should
> I do with them? I have found no documentation on port 8004.
> Thanks in advance,
> Yitzhak Bar Geva
>
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[asterisk-users] Asterisk sends packets on 8004/udp

[Yitzhak Bar Geva]

For the life of me can't figure out why the Asterisk server generates an
enormous quantity of outgoing packets on port 8004/udp. They seem to have no
effect whether they are blocked by the firewall or not.
We're running SIP. Everything appears to be OK (except a large number of
ChanSpy write buffer overflow messages, which I also don't understand).
How can I discover whether these packets are desirable or not? What should I
do with them? I have found no documentation on port 8004.
Thanks in advance,
Yitzhak Bar Geva
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Re: [asterisk-users] opensr Vs freeswitch SIP proxy server

[ram]

On 10/23/07, satish patel <[EMAIL PROTECTED]> wrote:
>
> dear ram
>
> i have also find many document about freeswitch and
> openser and i thing openser is best then freeswitch it is also module base
> as well as handle thousand of sip call and easy to impliment with DB but
> freeswitch is XML base and i am not familer with XML language thats why from
> my point of view is it taff task
>
>

Hi

i recomend to spend some time and read the documents, and see what is the
best to suite your need
and  find out your own capabilities to deploy the solution. if you feel the
task can not achive by you.

then opt some cosultant or use some commercial software available to do the
best.

ram
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Re: [asterisk-users] Force codec order

[ram]

On 10/23/07, Il Neofita <[EMAIL PROTECTED]> wrote:
>
> There is a way to force the order of the codecs in the sip.conf since the
> allow seams to let know only the accepted codec.


Hi

yes you can do, at client side and as well as Asterisk side.

disallow=all
allow=first codec
allow=second one so on

ram
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Re: [asterisk-users] White noise from TDM2400

[Matthew Fredrickson]

Stephen Kratzer wrote:
> Hello. We recently replaced a channel bank in favor of a TDM2400E. After 
> doing 
> so, users began complaining that they could barely hear the remote parties. 
> We increased gain appropriately for each channel which increased the volume 
> of the voices but has also increased the volume of any line noise. It sounds 
> like white noise which goes away when either party talks and returns during 
> silence. Is there any remedy to this? Thanks.

Do you have the new TDM2400E with the VPMADT032 on it?  Also, what 
version of zaptel are you running?

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX

[Matthew Fredrickson]

[EMAIL PROTECTED] wrote:
> On this machine its the first install, but i get this error 3 month before 
> on an other machine also.
> 
> I think the debug will bring t much data, cause there is any half 
> second a call try, and its really hard to find this error in the debug 
> file.
> The only thing i know is if i use a Sangoma card, the problem went.
> 
> Can i send you a BIG debug file where some of this errors happened?

If you can post a debug of a call when it happened and a call when it 
doesn't happen, that would help the most.

> 
> 
> Thanks
> 
> Nico
> 
> On Fri, 19 Oct 2007, Matthew Fredrickson wrote:
> 
>> [EMAIL PROTECTED] wrote:
>>> Hi,
>>>
>>> I'm running some Asterisk-machines, and on one of them i get this errors
>>> in the CLI, but i don't know what that means.
>>>
>>> Hardware:
>>> Digium 4-Port E1 Card with HWEC
>>> Intel Pentium D 3 GHz
>>> 2 GB RAM
>>> SATA Harddisk
>>> Supermicro Mainboard
>>>
>>> Software:
>>> latest libpri/zaptel/asterisk of version 1.2
>>>
>>> I tried also asterisk version 1.4.x, but there the problem occurs every 10
>>> calls, on asterisk 1.2 its about every 100 calls.
>> Did this recently start, like after you upgraded or is this something
>> that has always been a problem for you since you installed?
>>
>> If it has always been a problem, can you post a `pri debug span x` trace
>> of a call when this happens?  That will help to know more about what is
>> going on here.
>>
>> -- 
>> Matthew Fredrickson
>> Software/Firmware Engineer
>> Digium, Inc.
>>
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-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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[asterisk-users] Sharing lines with multiple buttons in Cisco 7960?

[Bruce Komito]

Has anyone come up with a way of sharing a single SIP registration with
two or more line buttons on the Cisco 79x0?  This is possible on a Linksys
94x, but I haven't found the magic parameter on the Cisco (assuming there
is one).

TIA

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815




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Re: [asterisk-users] Best USB Handset and Softphone Combination

[Omar A. Sabek]

Hello Steve,

I personally use an 'ultra-portable' headset from Logitech, I would
recommend this for any end-users that have laptops:

http://www.logitech.com/index.cfm/webcam_communications/internet_headsets_phones/devices/223&cl=us,en

There is also this model for desktops:

http://www.logitech.com/index.cfm/webcam_communications/internet_headsets_phones/devices/3622&cl=us,en

I've been using Counterpath Eyebeam with the USB headset for a year,
and I never had a problem. Eyebeam automatically switches it's profile
to use the USB mic and headset when it detects thats it's plugged in.

We haven't deployed Eyebeam to our clients because of the risk
involved with running a phone on an unmanaged desktop. Most of our
customers have little desktop management, if any. But I would
recommend either Eyebeam or Bria, they have a provisioning system for
easier management and the phones can be rebranded.

Good luck ;-)

Omar

On 10/19/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
> I have a client that want to try the softphone with USB handsets route
> to see if hardphones will even be needed.  I always push for hardphones
> (Polycom) so I am not sure about softphones or USB handsets.
>
> This is going to be for a 300+ seat call center onsite and many offsite,
> I plan on using OpenVPN for the offsite machines.
>
> Any advice on softphones, handsets, or practical experience with this
> sort of deployment?  It would be very nice if there was a central way of
> provisioning the phones.
>
> All machines are fairly new (newer than two years), they have very
> strict policies on downloads and streaming.
>
> Thanks in advance.
>
> Thanks,
> Steve
>
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[asterisk-users] White noise from TDM2400

[Stephen Kratzer]

Hello. We recently replaced a channel bank in favor of a TDM2400E. After doing 
so, users began complaining that they could barely hear the remote parties. 
We increased gain appropriately for each channel which increased the volume 
of the voices but has also increased the volume of any line noise. It sounds 
like white noise which goes away when either party talks and returns during 
silence. Is there any remedy to this? Thanks.

Stephen Kratzer
Network Engineer
CTI Networks, Inc.

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[asterisk-users] Data calls through TDM2400E

[Stephen Kratzer]

Hello. Has anyone been able to successfully make data (dialup modem) calls 
through a TDM2400E? We're able to make fax and credit card calls fine, but 
cannot successfully make modem calls using a 56K modem connected to a patch 
panel connected to an FXS port which then gets bridged to an FXO port 
connected directly to a phone line. We have 'echocancelwhenbridged=no' set 
in /etc/asterisk/zapata.conf.  We're dialing into a Lucent TNT which drops 
the call with a cause code of 11 (DCD-Detected-Then-Inactive. The modem 
detected DCD, but became inactive). The modem call works fine when connected 
directly to a phone line. Is there anything else that I can do to get this 
working? Thanks.

Stephen Kratzer
Network Engineer
CTI Networks, Inc.

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[asterisk-users] Asterisk <-> Noetel C15K ?

[shawnl]

Has anyone had any luck getting an asterisk box to talk to a Nortel
C15K softswitch?  I've been playing with it for several days and can't
seem to pass calls either direction.   I know that whike the Nortel
says the C15K speaks SIP, it really speaks nortel's implementation of
SIP, but I thought I could get it to at least pass simple calls back
and forth to an asterisk box.

Right now, I can't even get asterisk to register with it.

Anyone have any ideas?

thanks!



register => username:[EMAIL PROTECTED]

[nortel]
type=friend
fromuser=username
username=username
canreinvite=yes
secret=passwd
host= 192.168.1.20
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=rfc2833
qualify=yes
nat=no 
usereqphone=yes
context=from-nortel


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Re: [asterisk-users] opensr Vs freeswitch SIP proxy server

[Baji Panchumarti]

 With all due respect, please try not to make up spellings based
 on pronunciation. There is no  "taff task"  it is  "tough task".

 If someone is going to take the time to answer a question, the
 least we can do is clearly communicate the question.

 Spellcheck is readily available where needed.

 My apology for the off-topic response.

--

  On 10/23/07, satish patel  wrote:

> dear ram
>
> i have also find many document about freeswitch and openser
> and i thing openser is best then freeswitch it is also module base as well
> as handle thousand of sip call and easy to impliment with DB but freeswitch
> is XML base and i am not familer with XML language thats why from my point
> of view is it taff task
>
> Regards
>
> Satish Patel
>
> ram wrote:
>
> On 10/23/07, satish patel wrote:
> >
> > Dear all
> >
> > I have plan for 5000 user register on sip server and call to
> each other according his/her domain ( Relam ) so which one is best for this
> type of aaplication or stablity to handle thousand of sip reqest i have
> study of both product but i need input from community end suggest me best
> one which can easy and stable for my production
> >
> > my reqierment is
> >
> > [EMAIL PROTECTED]
> > [EMAIL PROTECTED]
> >
> > [EMAIL PROTECTED]
> > [EMAIL PROTECTED]
> >
> > this all domain on my sip server and place all according his domain not
> interdomain
> >
> > Regards
> >
>
> Hi
>
> for this kind of things
>
> OpenSER is the best, even Freeswitch can do the Job, but OpenSER
> there since long and testing Million users
>
> ram
> __

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Re: [asterisk-users] Force codec order

[Baji Panchumarti]

  On 10/22/07, Il Neofita  wrote:

> There is a way to force the order of the codecs in the sip.conf
> since the allow seams to let know only the accepted codec.

 I don't know about sip specifically, but from what I recall
 reading the .conf files   use   disallow=all  and then add
 codecs one by one, I believe the order in which you add
 them is the order they are checked in.

 your server cannot send a stream that the other side
 cannot decode, so both sides have to agree on a codec.

 Don't know if that answers your question.

 --

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Re: [asterisk-users] Short format of SIP INVITE - how to change

[Philipp Kempgen]

Vitaly wrote:

> My Asterisk box send INVITEs  in the short form, i.e.,
> "f:" instead of "from", "v:" instead of "via" and so
> on.
> Is there a way to force asterisk to use full format?

sip.conf:
compactheaders=no

afaik it's disabled by default so it was you who enabled it.

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] opensr Vs freeswitch SIP proxy server

[satish patel]

dear ram 
   
  i have also find many document about freeswitch and openser 
and i thing openser is best then freeswitch it is also module base as well as 
handle thousand of sip call and easy to impliment with DB but freeswitch is XML 
base and i am not familer with XML language thats why from my point of view is 
it taff task 
   
  Regards
   
  Satish Patel

ram <[EMAIL PROTECTED]> wrote:
  

  On 10/23/07, satish patel <[EMAIL PROTECTED]> wrote: Dear all
   
I have plan for 5000 user register on sip server and call to each 
other according his/her domain ( Relam ) so which one is best for this type of 
aaplication or stablity to handle thousand of sip reqest i have study of both 
product but i need input from community end suggest me best one which can easy 
and stable for my production 
   
  my reqierment is 
   
  [EMAIL PROTECTED] 
  [EMAIL PROTECTED]
   
  [EMAIL PROTECTED]
  [EMAIL PROTECTED] 
   
  this all domain on my sip server and place all according his domain not 
interdomain 
   
  Regards
   
   
   
  Hi
   
  for this kind of things
   
  OpenSER is the best, even Freeswitch can do the Job, but OpenSER there since 
long
  and testing Million users
   
  ram


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Re: [asterisk-users] Snom 360 lights not working on subscription

[Dr. Michael J. Chudobiak]

>> In order to get subscriptions working and the Snom 360 lights turns  
>> on, I have set everything just like all the pages in the net explain.
>>
>> So, I get subsciption working. I can list subscription on the  
>> asterisk and if I use the SIP trace function built in at the SNOM nad  
>> see NOTIFY messages and 200 OK responses. But I realized that content  
>> length = 0 in all messsages and there isn't any XML content in those  
>> Notify headers..

 > What we found is that even if you get the lights working, they go off
 > after a few days.

The BLF lights on the Snom 360s work for me (Asterisk 1.4, Snom 6.5.12 
firmware), but I reboot them nightly.

I have noticed that the Snom BLFs can stop working if the network is 
busy for a long period of time (i.e., longer than the re-registration 
period), like during system-wide backups and yum-upgrades. To avoid this 
problem, I have a cron job reboot the Snoms nightly after scheduled 
backups/upgrades. I'm not sure if this is a network congestion issue or 
a server CPU overload issue, or something else. Anyway, this arrangement 
does seem to be pretty reliable.

To reboot a Snom: 
http://www.voip-info.org/wiki/view/Asterisk+phone+snom#RebootingaSNOM360320.

Hope this helps.


- Mike

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[asterisk-users] Short format of SIP INVITE - how to change

[Vitaly]

My Asterisk box send INVITEs  in the short form, i.e.,
"f:" instead of "from", "v:" instead of "via" and so
on.
Is there a way to force asterisk to use full format?

thanks
Vitaly

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Re: [asterisk-users] opensr Vs freeswitch SIP proxy server

[ram]

On 10/23/07, satish patel <[EMAIL PROTECTED]> wrote:
>
> Dear all
>
>   I have plan for 5000 user register on sip server and call to
> each other according his/her domain ( Relam ) so which one is best for this
> type of aaplication or stablity to handle thousand of sip reqest i have
> study of both product but i need input from community end suggest me best
> one which can easy and stable for my production
>
> my reqierment is
>
> [EMAIL PROTECTED]
> [EMAIL PROTECTED]
>
> [EMAIL PROTECTED]
> [EMAIL PROTECTED]
>
> this all domain on my sip server and place all according his domain not
> interdomain
>
> Regards
>
>


Hi

for this kind of things

OpenSER is the best, even Freeswitch can do the Job, but OpenSER there since
long
and testing Million users

ram
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Re: [asterisk-users] [France CID] Does Zaptel support ETSI FSK?

[Vincent]

On Mon, 22 Oct 2007 16:09:39 -0700, Ira <[EMAIL PROTECTED]> wrote:
>try adding a wait(1) right in the beginning, worked for me.

Thanks but I had this before, and it makes no difference. Jared
explained above why CID isn't displayed when using 1.4.


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[asterisk-users] opensr Vs freeswitch SIP proxy server

[satish patel]

Dear all
   
I have plan for 5000 user register on sip server and call to each 
other according his/her domain ( Relam ) so which one is best for this type of 
aaplication or stablity to handle thousand of sip reqest i have study of both 
product but i need input from community end suggest me best one which can easy 
and stable for my production 
   
  my reqierment is 
   
  [EMAIL PROTECTED] 
  [EMAIL PROTECTED]
   
  [EMAIL PROTECTED]
  [EMAIL PROTECTED] 
   
  this all domain on my sip server and place all according his domain not 
interdomain 
   
  Regards
   
  Satish Patel

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[asterisk-users] G.729 codec between avaya and asterisk

[satish patel]

Dear all
   
  i have asterisk connected with avaya through E1 back-2-back now when 
i configure my sip client with g.729 codec then i m not able to put call from 
asterisk to avaya and when i user g.711 it is working fine so i dont know why i 
need G.729 on E1 Trunk it is TDM technologies then why my call fail in g.729 
case 
   
   
  [sip_phone]--[asterisk]-E1[Avaya][analog_phone]
   
  Asterisk sip client configure with g.711 alaw/ulaw
  Avaya phone client configure g.711 alaw/ulaw
   
  suggest how do it implement g.729 on this case what change i have to done on 
both part

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[asterisk-users] Internal Data Stream Error

[Lees, James (UK)]

t;Beginning with Asterisk 1.4, we moved all of the CallerID functionality
>from channel variables and applications to a single CALLERID dialplan
>function.  This should have been noted in UPGRADE.txt.  I also tried to
>warn you about it in my last email in this thread, but I guess I should
>have been more specific.

No problem. I should have read it more closely, but due to the number
of people having problems with Zaptel and CID, I was focused on that
part. Should have started asking people what the correct way was to
read CID information in Asterisk 1.4... Thanks.




--

Message: 25
Date: Mon, 22 Oct 2007 16:05:06 -0700 (PDT)
From: bilal ghayyad <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] G729a codecs + Asterisk 1.4.11
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=iso-8859-1

Dear Marc;

I readed your email about the codec G729a and I am now
also need to install the codec on my Asterisk. 

I typed from Asterisk CLI:

core show version and I got the following:

Asterisk SVN-branch-1.4-r72556 built by root @
localhost.localdomain on a i686 running Linux on
2007-06-30 13:08:08 UTC

So I beleive that my processor is i686, correct? But I
am not able to know which one to download:

The x86-32 or x86-64 ? Can you please advise.

Also, the nocona or the opteron versions?

Regards
Bilal

---
Good Morning,
Any help would be grateful to help me understanding
what's wrong...

I have bought 2 g729a licenses to digium and I would
like to have them
 works...
My processor is an Intel(R) Xeon(R) CPU  
E5310  @ 1.60GHz (4
 processors)
so I have downloaded the

http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64
/codec_g729a_v32_nocona.tar.gz
 codec
I have registered my license, copied the
codec_g729a.so into the
 /usr/lib/asterisk/modules folder and restarted my
asterisk

But on the CLI when I type
asterisk*CLI> show modules like 72
Module Description
   
  Use Count
codec_g726.so  ITU G.726-32kbps G726
Transcoder
 0
format_g729.so Raw G729 data  
   
  0
format_g726.so Raw G.726
(16/24/32/40kbps) data
 0
format_g723.so G.723.1 Simple
Timestamp File Format
 0

The codec_g729a.so doesn't appear..


Any idea how to solve the problem.

Thanks

Best Regards,

Marc LEURENT


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--

Message: 26
Date: Mon, 22 Oct 2007 16:09:39 -0700
From: Ira <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] [France CID] Does Zaptel support ETSI
FSK?
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"; format=flowed

At 02:18 PM 10/22/2007, you wrote:
>;nothing displayed
>exten => s,n,Verbose(${CALLERID})
>exten => s,n,Verbose(${CALLERIDNAME})
>exten => s,n,Verbose(${CALLERIDNUM})
>exten => s,n,NoOp(${CALLERID})
>exten => s,n,Verbose(${CALLERID})
>
>;CID at last!
>exten => s,n,Verbose(${CALLERID(num)})
>
>
>I'm running Asterisk 1.4. Does someone know why only the last
>statement does display the CID number while the others print nothing?


try adding a wait(1) right in the beginning, worked for me.

Ira




--

Message: 27
Date: Mon, 22 Oct 2007 19:15:55 -0400
From: Jason Lixfeld <[EMAIL PROTECTED]>
Subject: [asterisk-users] Voicemail playback on iPhone
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=US-ASCII; format=flowed

Anyone managed to get this to work?  What's the recipe?



--

Message: 28
Date: Tue, 23 Oct 2007 01:35:13 +0200
From: "Yitzhak Bar Geva" <[EMAIL PROTECTED]>
Subject: [asterisk-users] NAT traversal packet loss measurement
To: asterisk-users@lists.digium.com
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"

How can one measure the effect of NAT traversal packet loss?
We currently have no solution for NAT traversal for our SIP clients.
There
is no doubt that packets are getting lost. What is not clear is how much
damage this does. On the face of it, everything seems fine. Could this
be
so? Perhaps we're suffering a degradation in quality or our call setup
times
could be improved. How can we measure this?
What's the simplest method of preventing packet loss due to NAT
traversal in
a SIP environment?
Thanks,
Yitzhak
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An HTML attachment was scrubbed...
URL:
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Re: [asterisk-users] Video Conference

[Dovid B]

> Thanks for the responce.  Have you had any luck at all even with what one
> might not consider straight forward?  I am trying to avoid paying the 
> $1000+
> per location needed to purchase something from say Polycom or Tandberg.  I
> would even be willing to do something along the lines of a web app for 
> video
> and some how tie that together with the voice through Asterisk.  Just 
> don't
> want to look like one of the old dubbed over Japanese movies from when I 
> was
> a kid (lips move and then a couple seconds later you hear voice).
> JohnM

John,
Try contacting [EMAIL PROTECTED] They have some solution there 
that works with Asterisk. 



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Re: [asterisk-users] Best USB Handset and Softphone Combination

[Dovid B]

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

Sent: Friday, October 19, 2007 5:53 PM
Subject: Re: [asterisk-users] Best USB Handset and Softphone Combination


> On 10/19/07, Mike Clark <[EMAIL PROTECTED]> wrote:
>>
>> Do they play well with Vista?
>
> Hah - I have no idea.  We installed Vista on one laptop here when Dell
> started shipping it.  That lasted about 3 days and 10 support tickets
> from the user.  Then we reverted back to XP.  Haven't touched Vista
> since.
>
> -erik
>
Not to start a flame war mut M$ seems to let the cutomer be the Beta tester. 
IMHO I would stick with XP for now. Seems to be too many issues with Vista 
(when it comes to VOIP at least and I am sure there are other issues). 



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Re: [asterisk-users] app_swift issues

[Yair Hakak]

Hi list,
 just wanted to answer my own question for general knowledge - turns out
app_swift requires that the voice be set in swift.conf. If the default voice
in swift is not David-8KhZ then it will say no voice is found, even if a
swift voice is installed. Thanks to Earle for helping me with this.

-yair


On 10/22/07, Yair Hakak <[EMAIL PROTECTED]> wrote:
>
> Hi all,
>  i'm trying to integrate cepstral and asterisk, and i have a problem i'd
> appreciate any help with (i know it's a bit tangential, but i figure this is
> the place with the most knowledge of app_swift and asterisk).
> I've installed swift from cepstral.com with alison's voice, and it works
> fine, from the command line i can do swift "hello there" -o test.wav and
> then i play the wav and it includes the text. All good.
> I've also installed app_swift according to the instructions here
> http://www.mezzo.net/asterisk/app_swift.html, and "show application swift"
> from the asterisk CLI brings up the application installed.
>
> Now, when i put Swift('This is a test') in the extensions.conf file, i get
> the following:
> ERROR[3495]: app_cepstral.c:197 cepstral_speak: Failed to set voice.
>
> I have not touched swift.conf (i'm using the defaults), and, i should add
> that i have not yet purchased the cepstral voices so that when i run from
> the command line it sticks "this voice is unlicensed..." or something like
> that at the beginning of the file, if that makes some kind of difference.
>
> I found the problem referenced here: 
> http://www.cepstral.com/forum/viewtopic.php?t=56&sid=baa6669e9958920393c62510caa47123&PHPSESSID=df1bcc629c4b8b37617d2d72c8b0232e
>  but no solution...
>
> any help will be most appreciated,
>
> thanks,
>  yair
>
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Re: [asterisk-users] G729a codecs + Asterisk 1.4.11

[Marc LEURENT]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

To know your architecture, use the cmd: cat /proc/cpuinfo

After try to start to use the version below (i686):
http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-32/codec_g729a_v32_i686.tar.gz

Good luck


bilal ghayyad a écrit :
> Dear Marc;
> 
> I readed your email about the codec G729a and I am now
> also need to install the codec on my Asterisk. 
> 
> I typed from Asterisk CLI:
> 
> core show version and I got the following:
> 
> Asterisk SVN-branch-1.4-r72556 built by root @
> localhost.localdomain on a i686 running Linux on
> 2007-06-30 13:08:08 UTC
> 
> So I beleive that my processor is i686, correct? But I
> am not able to know which one to download:
> 
> The x86-32 or x86-64 ? Can you please advise.
> 
> Also, the nocona or the opteron versions?
> 
> Regards
> Bilal
> 
> ---
> Good Morning,
> Any help would be grateful to help me understanding
> what's wrong...
> 
> I have bought 2 g729a licenses to digium and I would
> like to have them
>  works...
> My processor is an Intel(R) Xeon(R) CPU  
> E5310  @ 1.60GHz (4
>  processors)
> so I have downloaded the
> 
> http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64/codec_g729a_v32_nocona.tar.gz
>  codec
> I have registered my license, copied the
> codec_g729a.so into the
>  /usr/lib/asterisk/modules folder and restarted my
> asterisk
> 
> But on the CLI when I type
> asterisk*CLI> show modules like 72
> Module Description
>
>   Use Count
> codec_g726.so  ITU G.726-32kbps G726
> Transcoder
>  0
> format_g729.so Raw G729 data  
>
>   0
> format_g726.so Raw G.726
> (16/24/32/40kbps) data
>  0
> format_g723.so G.723.1 Simple
> Timestamp File Format
>  0
> 
> The codec_g729a.so doesn't appear..
> 
> 
> Any idea how to solve the problem.
> 
> Thanks
> 
> Best Regards,
> 
> Marc LEURENT
> 
> 
> __
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