Re: [asterisk-users] Telephony Card: GSM slots + Analoge

[Mitul Limbani]

Dear Moises,

Does Sangoma manufacture 4 port gsm card? or is that an Openvox card?

Mitul
On May 26, 2012 12:23 PM, "Moises Silva"  wrote:

>
> On Sun, Apr 1, 2012 at 9:04 AM, bilal ghayyad  wrote:
>
>> Dears;
>>
>> I am looking to get a telephony card that has GSM slots (ability to place
>> my GSM card into it) in addition to analoge FXS and FXO.
>>
>> There is a card that I found it but really I do not know how much it is
>> reliable: http://www.atcom.cn/AX2G4A.html
>>
>> Did anyone tried atcom?
>>
>> Is there a similar cards like it (but to be USA, Canada or Europe
>> manufacturing)?
>>
>>
> There is nothing hybrid like that (GSM + Analog) in the NorthAmerica or
> Europe to my knowledge. We at Sangoma (from Canada) have a 4-port GSM card
> though which uses chan_dahdi (patching needed at the moment).
>
> *Moises Silva
> **Manager, Software Engineering***
>
> msi...@sangoma.com
>
> Sangoma Technologies
>
> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>
>
> t.   +1 800 388 2475 (N. America)
>
> t.   +1 905 474 1990 x128
>
> f.   +1 905 474 9223
>
>
>
> **
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Re: [asterisk-users] Telephony Card: GSM slots + Analoge

[Moises Silva]

On Sun, Apr 1, 2012 at 9:04 AM, bilal ghayyad  wrote:

> Dears;
>
> I am looking to get a telephony card that has GSM slots (ability to place
> my GSM card into it) in addition to analoge FXS and FXO.
>
> There is a card that I found it but really I do not know how much it is
> reliable: http://www.atcom.cn/AX2G4A.html
>
> Did anyone tried atcom?
>
> Is there a similar cards like it (but to be USA, Canada or Europe
> manufacturing)?
>
>
There is nothing hybrid like that (GSM + Analog) in the NorthAmerica or
Europe to my knowledge. We at Sangoma (from Canada) have a 4-port GSM card
though which uses chan_dahdi (patching needed at the moment).

*Moises Silva
**Manager, Software Engineering***

msi...@sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



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[asterisk-users] Help ! Audio not stored .

[Durgesh Mishra]

Hi; 



In Voicemail.conf  



If am am using 

format=h263|gsm ,and i want to store only audio , then it is not storing.In log 
it shows that video is deposite less then 5 second. If i want to store video 
and audio both then it will store properly. 



If am using   



format=gsm|h263 ,then my Xlite  softphone will go to haung. 



I just want to store audio and video both or some time only audio . 

1)Plz guide me which combination of codec will be usefull. 

2)Is there is any serial number signifance in format,ie one time if i use as 
format=h263|gsm and second time i am using format=gsm|h263,why  is diffrence  
come? 







Thanks 

Durgesh Mishra 

Rancore Technologies.--
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Re: [asterisk-users] Loss of RTP stream during DTMF collection

[Dave George]

Hi Kevin,

I have two asterisk boxes with the same issues.
Box 1: asterisk ver 1.4.21.2
Box 2: Asterisk 1.8.7.1

setup:
CDMA Phone <> CDMA Media Gateway WCM  Asterisk voice mail


The calls are SIP Based.  DTMF collection is when the user is entering a
password for voice mail access or voucher to recharge their account. 

voice mail:
user is prompted for a password. After password is entered I can see
asterisk playing the voice mail but no audio is heard on the phone.


Other scenario user dials into a voucher menu (Asterisk2billing) and is
prompted for a voucher. No audio after the voucher is entered.

The CDMA guys did a trace on their end and this is what they explained
is happening:

The voicemail problem is due to the time stamp jump on the RTP steam
sending WCM to BSC.   There are about 5 seconds gap between two
consecutive RTP packets.   It was caused by Asterisk not sending any RTP
packet to WCM.

How can I enable the option to allow asterisk to maintain the RTP stream
during DTMF collection?

Thanks,
Dave


>  Original Message 
> Subject: Re: [asterisk-users] Loss of RTP stream during DTMF collection
> From: "Kevin P. Fleming" 
> Date: Fri, May 25, 2012 5:38 pm
> To: asterisk-users@lists.digium.com
> 
> 
> On 05/25/2012 04:30 PM, Dave George wrote:
> > I am using asterisk for voice mail.  During DTMF collection Asterisk
> > stop sending any RTP Packets. The gap between two consecutive packets
> > are 4 seconds, which is huge enough to screw up the jitter buffer.  When
> > ever asterisk stops to receive DTMF, the RTP stream is cut and we loose
> > audio.
> >
> > I don't have this issue when calling from a SIP phone.  I only have this
> > issue when calling from one media gateway to the asterisk box.
> >
> > Any suggestions welcome.  Can I play some file in the back while
> > collecting DTMF?
> 
> You are missing quite a lot of crucial information required for anyone 
> to help you. First, what version of Asterisk are you using? Second, what 
> type of channel is being used to connect to Asterisk? You mention it 
> works from a SIP phone, but not from a media gateway.. is that gateway 
> also using SIP, or something else? What does 'during DTMF collection' 
> mean? Do you mean after a prompt has been played and the voicemail 
> application is waiting for input, or is this during prompt playback, or 
> something else?
> 
> Quite some time ago Asterisk was changed to ensure that silence would be 
> sent while an application was running and waiting for input from the 
> caller; if your version is older than this, then that could explain what 
> you are seeing. That's just a mildly-educated guess though.
> 
> -- 
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
> 
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Re: [asterisk-users] Loss of RTP stream during DTMF collection

[Kevin P. Fleming]

On 05/25/2012 04:30 PM, Dave George wrote:

I am using asterisk for voice mail.  During DTMF collection Asterisk
stop sending any RTP Packets. The gap between two consecutive packets
are 4 seconds, which is huge enough to screw up the jitter buffer.  When
ever asterisk stops to receive DTMF, the RTP stream is cut and we loose
audio.

I don't have this issue when calling from a SIP phone.  I only have this
issue when calling from one media gateway to the asterisk box.

Any suggestions welcome.  Can I play some file in the back while
collecting DTMF?


You are missing quite a lot of crucial information required for anyone 
to help you. First, what version of Asterisk are you using? Second, what 
type of channel is being used to connect to Asterisk? You mention it 
works from a SIP phone, but not from a media gateway.. is that gateway 
also using SIP, or something else? What does 'during DTMF collection' 
mean? Do you mean after a prompt has been played and the voicemail 
application is waiting for input, or is this during prompt playback, or 
something else?


Quite some time ago Asterisk was changed to ensure that silence would be 
sent while an application was running and waiting for input from the 
caller; if your version is older than this, then that could explain what 
you are seeing. That's just a mildly-educated guess though.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] Loss of RTP stream during DTMF collection

[Dave George]

I am using asterisk for voice mail.  During DTMF collection Asterisk
stop sending any RTP Packets. The gap between two consecutive packets
are 4 seconds, which is huge enough to screw up the jitter buffer.  When
ever asterisk stops to receive DTMF, the RTP stream is cut and we loose
audio. 

I don't have this issue when calling from a SIP phone.  I only have this
issue when calling from one media gateway to the asterisk box.

Any suggestions welcome.  Can I play some file in the back while
collecting DTMF?


Dave

 






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Re: [asterisk-users] Vitelity Setup

[Stephen J Alexander]

If your server says it is registered, that could be part of the problem.
Vitelity doesn't use trunk registration, only IP authentication. You should
not be using a registration string in your trunk definition. I don't know
if it will hurt but it won't help.

It sounds like you might have only 1 trunk defined, but you need 2; one for
inbound and one for outbound. Their servers for incoming calls and for
outgoing calls are separate. If fixing that doesn't do the job, make sure
that incoming traffic from Vitelity is correctly routed to your PBX (and
that they have the correct IP to send SIP traffic to).

Regards,

Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729


On Fri, May 25, 2012 at 4:12 PM, Ralph Green  wrote:

> Howdy,
>  Since the subject is Viteiy Setup, I don't think this is off topic.
> My big problem with Vitelity is getting my server to register for
> incoming calls.  I can make outgoing calls just fine.  My server says
> it is registered with Vitelity, but no calls come in.  Every attempt
> to call the number generates an email saying there was a failed call.
> I am using IAX, not SIP, and that is probably part of the problem.
> IAX should work better in several ways, but few enough people use it.
> Vitelity support has been unhelpful so far.  My suspicion is that
> there is a setting they need to make in their server so that calls go
> to the registered IAX server, instead of looking for a SIP
> registration, which is not there.  Has anyone here worked past such a
> problem?  Was there some special thing I need to ask Vitelity?
> Thanks,
> Ralph
>
>
> On 5/24/12, Stephen J Alexander  wrote:
> > If I were troubleshooting this, the next thing I would do is verify
> > connectivity on the relevant ports – more plainly, make sure that there's
> > not a firewall rule with unintended consequences somewhere between your
> > asterisk and your ISP. Otherwise, as Alejandro suggests – check with
> > Vitelity support.
> >
> > Regards,
> >
> > Stephen J Alexander
> > MPBX, LLC
> > http://mpbx.com
> > 832-713-6729
> >
> >
> > On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass  wrote:
> >
> >> On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N
> >>  wrote:
> >> > yes I did that, even then i am not able to make outbound and inbound
> as
> >> > well.
> >> >
> >> >
> >>
> >>
> >> That's weird. Guess you're gonna have to place a detailed ticket to
> >> them. It sounds like a network problem to me but without any detailed
> >> info it's hard to say. Maybe you can try sip set debug in the console
> >> for the IP and see if you can get an idea of what is happening at the
> >> packet level.
> >>
> >> We use Vitel, Skype SIP (we recently eliminated this one), and now
> >> Gafachi and they all seem to work per there set-up instructions right
> >> away.
> >>
> >> --
> >> Alejandro
> >>
> >> --
> >> _
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >>   http://www.asterisk.org/hello
> >>
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> >> To UNSUBSCRIBE or update options visit:
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> >>
> >
>
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Re: [asterisk-users] Vitelity Setup

[Danny Nicholas]

Is your IAX2 peer registered?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ralph Green
Sent: Friday, May 25, 2012 4:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Vitelity Setup

Howdy,
  Since the subject is Viteiy Setup, I don't think this is off topic.
My big problem with Vitelity is getting my server to register for incoming
calls.  I can make outgoing calls just fine.  My server says it is
registered with Vitelity, but no calls come in.  Every attempt to call the
number generates an email saying there was a failed call.
I am using IAX, not SIP, and that is probably part of the problem.
IAX should work better in several ways, but few enough people use it.
Vitelity support has been unhelpful so far.  My suspicion is that there is a
setting they need to make in their server so that calls go to the registered
IAX server, instead of looking for a SIP registration, which is not there.
Has anyone here worked past such a problem?  Was there some special thing I
need to ask Vitelity?
Thanks,
Ralph


On 5/24/12, Stephen J Alexander  wrote:
> If I were troubleshooting this, the next thing I would do is verify 
> connectivity on the relevant ports - more plainly, make sure that 
> there's not a firewall rule with unintended consequences somewhere 
> between your asterisk and your ISP. Otherwise, as Alejandro suggests - 
> check with Vitelity support.
>
> Regards,
>
> Stephen J Alexander
> MPBX, LLC
> http://mpbx.com
> 832-713-6729
>
>
> On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass  wrote:
>
>> On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N 
>>  wrote:
>> > yes I did that, even then i am not able to make outbound and 
>> > inbound as well.
>> >
>> >
>>
>>
>> That's weird. Guess you're gonna have to place a detailed ticket to 
>> them. It sounds like a network problem to me but without any detailed 
>> info it's hard to say. Maybe you can try sip set debug in the console 
>> for the IP and see if you can get an idea of what is happening at the 
>> packet level.
>>
>> We use Vitel, Skype SIP (we recently eliminated this one), and now 
>> Gafachi and they all seem to work per there set-up instructions right 
>> away.
>>
>> --
>> Alejandro
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>

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Re: [asterisk-users] Vitelity Setup

[Ralph Green]

Howdy,
  Since the subject is Viteiy Setup, I don't think this is off topic.
My big problem with Vitelity is getting my server to register for
incoming calls.  I can make outgoing calls just fine.  My server says
it is registered with Vitelity, but no calls come in.  Every attempt
to call the number generates an email saying there was a failed call.
I am using IAX, not SIP, and that is probably part of the problem.
IAX should work better in several ways, but few enough people use it.
Vitelity support has been unhelpful so far.  My suspicion is that
there is a setting they need to make in their server so that calls go
to the registered IAX server, instead of looking for a SIP
registration, which is not there.  Has anyone here worked past such a
problem?  Was there some special thing I need to ask Vitelity?
Thanks,
Ralph


On 5/24/12, Stephen J Alexander  wrote:
> If I were troubleshooting this, the next thing I would do is verify
> connectivity on the relevant ports – more plainly, make sure that there's
> not a firewall rule with unintended consequences somewhere between your
> asterisk and your ISP. Otherwise, as Alejandro suggests – check with
> Vitelity support.
>
> Regards,
>
> Stephen J Alexander
> MPBX, LLC
> http://mpbx.com
> 832-713-6729
>
>
> On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass  wrote:
>
>> On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N
>>  wrote:
>> > yes I did that, even then i am not able to make outbound and inbound as
>> > well.
>> >
>> >
>>
>>
>> That's weird. Guess you're gonna have to place a detailed ticket to
>> them. It sounds like a network problem to me but without any detailed
>> info it's hard to say. Maybe you can try sip set debug in the console
>> for the IP and see if you can get an idea of what is happening at the
>> packet level.
>>
>> We use Vitel, Skype SIP (we recently eliminated this one), and now
>> Gafachi and they all seem to work per there set-up instructions right
>> away.
>>
>> --
>> Alejandro
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>

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Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk and how many servers

[Danny Nicholas]

Realtime is probably better for administration, but do you want to throw a 
layer of complication into such a large undertaking?  I wouldn’t want 20,000 
people screaming at me because MYSQL crapped out.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ing CIP. 
Alejandro Celi Mariátegui
Sent: Friday, May 25, 2012 2:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk 
and how many servers

 


Asterisk Realtime is better for administration. 

Performance,  IMHO is the same issue. I'm not lucky to made large 
implementations to test these.

Regards,


-- 
Ing CIP. Alejandro Celi Mariátegui 

http://cipher.pe/web/nuestra-experiencia.html 




El vie, 25-05-2012 a las 07:06 -0700, bilal ghayyad escribió: 

 
Hi John;
 
For 20,000 users: Is it better to use Asterisk realtime configuration or to use 
conf files?
 
I readed the below link but did not understand which GUI that works with 
asterisk realtime?
 
http://www.freepbx.org/trac/wiki/AsteriskRealtime
 
 
Regards
Bilal
 

> 
> > My question is:
> >
> > Is it really possible to have the asterisk
> configuration in the database server instead of having it in
> conf files? HOW? I am asking this because what I noticed in
> AsteriskNow and in A2Billing and Vicidial or Goautodial that
> whatever I do configuration in the GUI, then the
> configuration will be generated in the conf files, so
> Asterisk will read from the conf files and not from the
> database directly. Is it right or I am confused and there is
> something else?
> >
> > If there is a method to let the configuration to be
> taken from the database (and not from the configuration),
> then HOW? Because even in AsteriskNow, the configuration
> will be generated in a conf files.
> 
> Hi Bilal,
> 
> You want to look the Asterisk realtime configuration
> features if you 
> want to run your configuration from a database rather than
> configuration 
> files.
> 
> This should point you in the right direction and get you
> started:  
> http://www.voip-info.org/wiki/view/Asterisk+RealTime
> 
> It should be noted that if you're wanting to use AsteriskNow
> (which 
> relies on FreePBX for its gui configuration features), then
> Asterisk 
> realtime configuration will not work as it is not compatible
> at this 
> time.  Other web gui's might work, but I am not
> familiar with them.  
> FreePBX's sentiment on the subject is shared here:  
> http://www.freepbx.org/trac/wiki/AsteriskRealtime
> 
> -John
> 
> On 05/24/2012 05:46 PM, bilal ghayyad wrote:
> > Thanks for all for the help and kindly reply.
> >
> > One last point that will help me alot:
> >
> > I am thinking to have 4 Servers running Asterisk and 2
> Servers to be for database. The load to be distributed on
> the 4 Asterisk Servers with ability to be redundant (using
> any redundancy technique). The 4 Asterisk Servers to take
> the configuration from the Database Server and actually
> because there is 2 Database servers, then it will be
> redundant to each other (in case one database failed, the
> other will take over).
> >
> > My question is:
> >
> > Is it really possible to have the asterisk
> configuration in the database server instead of having it in
> conf files? HOW? I am asking this because what I noticed in
> AsteriskNow and in A2Billing and Vicidial or Goautodial that
> whatever I do configuration in the GUI, then the
> configuration will be generated in the conf files, so
> Asterisk will read from the conf files and not from the
> database directly. Is it right or I am confused and there is
> something else?
> >
> > If there is a method to let the configuration to be
> taken from the database (and not from the configuration),
> then HOW? Because even in AsteriskNow, the configuration
> will be generated in a conf files.
> >
> > Special thanks for the advise.
> >
> > Regards
> > Bilal
> > -
> >
> >>> Hi All;
> >>>
> >>> I need to use Asterisk for 20 000 users, so
> which
> >> asterisk version to be used? Is there asterisk
> version that
> >> supports 20,000 users on one hardware machine?
> >>> Can I use one strong hardware server i7 with 64
> GB RAM
> >> and fast hard desk to handle 20 000 users, and
> concurrent
> >> calls 2000? Or I need multiple servers, how much?
> >>> If I am going to use multiple servers (until
> now I do
> >> not know how much, and I do not know if the barrier
> will be
> >> the asterisk software or the hardware), then do I
> have to
> >> use special SIP proxy or I have to use load
> balancer)? In
> >> this case, I have to use asterisk Database (so all
> the
> >> servers will read/write from the database)?
> >>> What about AsteriskNow, can it support?
> >> AsteriskNOW is a GUI on top of Asterisk; it does
> not change
> >> the ability
> >> of the system to handle call load.
> >>
> >> Modern versions of Asterisk can easily handle
> 2,000
> >> simu

Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk and how many servers

[Ing CIP. Alejandro Celi]

Asterisk Realtime is better for administration. 

Performance,  IMHO is the same issue. I'm not lucky to made large
implementations to test these.

Regards,

-- 
Ing CIP. Alejandro Celi Mariátegui 

http://cipher.pe/web/nuestra-experiencia.html 



El vie, 25-05-2012 a las 07:06 -0700, bilal ghayyad escribió:

> Hi John;
> 
> For 20,000 users: Is it better to use Asterisk realtime configuration or to 
> use conf files?
> 
> I readed the below link but did not understand which GUI that works with 
> asterisk realtime?
> 
> http://www.freepbx.org/trac/wiki/AsteriskRealtime
> 
> 
> Regards
> Bilal
> 
> 
> > 
> > > My question is:
> > >
> > > Is it really possible to have the asterisk
> > configuration in the database server instead of having it in
> > conf files? HOW? I am asking this because what I noticed in
> > AsteriskNow and in A2Billing and Vicidial or Goautodial that
> > whatever I do configuration in the GUI, then the
> > configuration will be generated in the conf files, so
> > Asterisk will read from the conf files and not from the
> > database directly. Is it right or I am confused and there is
> > something else?
> > >
> > > If there is a method to let the configuration to be
> > taken from the database (and not from the configuration),
> > then HOW? Because even in AsteriskNow, the configuration
> > will be generated in a conf files.
> > 
> > Hi Bilal,
> > 
> > You want to look the Asterisk realtime configuration
> > features if you 
> > want to run your configuration from a database rather than
> > configuration 
> > files.
> > 
> > This should point you in the right direction and get you
> > started:  
> > http://www.voip-info.org/wiki/view/Asterisk+RealTime
> > 
> > It should be noted that if you're wanting to use AsteriskNow
> > (which 
> > relies on FreePBX for its gui configuration features), then
> > Asterisk 
> > realtime configuration will not work as it is not compatible
> > at this 
> > time.  Other web gui's might work, but I am not
> > familiar with them.  
> > FreePBX's sentiment on the subject is shared here:  
> > http://www.freepbx.org/trac/wiki/AsteriskRealtime
> > 
> > -John
> > 
> > On 05/24/2012 05:46 PM, bilal ghayyad wrote:
> > > Thanks for all for the help and kindly reply.
> > >
> > > One last point that will help me alot:
> > >
> > > I am thinking to have 4 Servers running Asterisk and 2
> > Servers to be for database. The load to be distributed on
> > the 4 Asterisk Servers with ability to be redundant (using
> > any redundancy technique). The 4 Asterisk Servers to take
> > the configuration from the Database Server and actually
> > because there is 2 Database servers, then it will be
> > redundant to each other (in case one database failed, the
> > other will take over).
> > >
> > > My question is:
> > >
> > > Is it really possible to have the asterisk
> > configuration in the database server instead of having it in
> > conf files? HOW? I am asking this because what I noticed in
> > AsteriskNow and in A2Billing and Vicidial or Goautodial that
> > whatever I do configuration in the GUI, then the
> > configuration will be generated in the conf files, so
> > Asterisk will read from the conf files and not from the
> > database directly. Is it right or I am confused and there is
> > something else?
> > >
> > > If there is a method to let the configuration to be
> > taken from the database (and not from the configuration),
> > then HOW? Because even in AsteriskNow, the configuration
> > will be generated in a conf files.
> > >
> > > Special thanks for the advise.
> > >
> > > Regards
> > > Bilal
> > > -
> > >
> > >>> Hi All;
> > >>>
> > >>> I need to use Asterisk for 20 000 users, so
> > which
> > >> asterisk version to be used? Is there asterisk
> > version that
> > >> supports 20,000 users on one hardware machine?
> > >>> Can I use one strong hardware server i7 with 64
> > GB RAM
> > >> and fast hard desk to handle 20 000 users, and
> > concurrent
> > >> calls 2000? Or I need multiple servers, how much?
> > >>> If I am going to use multiple servers (until
> > now I do
> > >> not know how much, and I do not know if the barrier
> > will be
> > >> the asterisk software or the hardware), then do I
> > have to
> > >> use special SIP proxy or I have to use load
> > balancer)? In
> > >> this case, I have to use asterisk Database (so all
> > the
> > >> servers will read/write from the database)?
> > >>> What about AsteriskNow, can it support?
> > >> AsteriskNOW is a GUI on top of Asterisk; it does
> > not change
> > >> the ability
> > >> of the system to handle call load.
> > >>
> > >> Modern versions of Asterisk can easily handle
> > 2,000
> > >> simultaneous calls,
> > >> even with media (non-transcoded) passing through
> > the server.
> > >> We have a
> > >> community member who has improved chan_sip in
> > Asterisk 10
> > >> (and later) to
> > >> be able to handle 10,000 simultaneous calls.
> > >>
> > >> Handling 20,000 registrations is probably more of

Re: [asterisk-users] Function not Registered??

[Mark Wiater]

On 5/25/2012 3:18 AM,  Lee, John (Sydney) said:
>
>  -- Executing [*1223*1**1900@incoming:78] Set("SIP/1900-08ee1da8", 
> "DEVSTATE(Custom:cfalw1900)=INUSE") in new stack
>
I use
   
'Set(DEVICE_STATE(Custom:var)=BUSY)'

in my 1.4 dialplans to set device state.

mark
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[asterisk-users] URA

[Luis H. Forchesatto]

Hi

Recently our asterisk system stopped beign recognized by URA in others
telephones exchanges. What's the troubleshoot steps for this issue?


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Re: [asterisk-users] Dual- or Quad ISDN cards for PCI-X Slots

[Kevin P. Fleming]

On 05/25/2012 11:10 AM, Michelle Konzack wrote:


I am hit by some frustrations because my Server has only two PCI-X slots
and my Eicon Diva 4BRI-8M, which should work fine with Asterisk, is only
PCI 2.0 standard and does not fit into the PCI-X slot.


This does not make sense; PCI 2.0 cards should fit just fine into PCI-X 
slots. Do you mean PCI-Express instead? That's very different.


--
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Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] hangup not detected?

[Justin Killen]

Swift() is an asterisk wrapper around the text-to-speech engine cepstral. Looks 
like this is a dev issue - I'll start a new thread on the dev mailing list.


Justin Killen
Senior Programmer / Analyst
All American Asphalt
951-736-7600 x 2060
jkil...@allamericanasphalt.com

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada
Sent: Thursday, May 24, 2012 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup not detected?

Looks like Swift() (whatever that is) is not returning ?
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[asterisk-users] Huawei K3765-HV with Asterisk?

[Michelle Konzack]

Hi again,

does someone use the USB-Stick "Huawei K3765-HV" with Asterisk?

I have an 4-Port TT-Hub (Cypress Chip) and 4 USB-Sticks with 4 different
GSM Provider (Germany, France, Turkey, and Iran)

Currently I use  the  "Vodafone Easybox 803 A"  together  with  an  ISDN
connection to my Asterisk Server, an analog telephone (PA710) and an USR
Sportster Vi 14.400 Fax-Modem (HylaFax).

Grmpf, I was not abele to send faxes from my workstations direktly using
Asterisk

And then the last question:

Does someone know, how the OpenSMS API is working?

The EasyBox 803A does support it and I need to get the SMS from the USB-
Stick.  It is enough, if thex could be fetched and send as mail to me.

Thanks, Greetings and nice Day/Evening
Michelle Konzack

-- 
# Debian GNU/Linux Consultant ##
   Development of Intranet and Embedded Systems with Debian GNU/Linux
   Internet Service Provider, Cloud Computing


itsystems@tdnet Jabber  linux4miche...@jabber.ccc.de
Owner Michelle Konzack

Gewerbe Strasse 3   Tel office: +49-176-86004575
77694 Kehl  Tel mobil:  +49-177-9351947
Germany Tel mobil:  +33-6-61925193  (France)

USt-ID:  DE 278 049 239

Linux-User #280138 with the Linux Counter, http://counter.li.org/


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[asterisk-users] Dual- or Quad ISDN cards for PCI-X Slots

[Michelle Konzack]

Hello ISDN Users.

I am hit by some frustrations because my Server has only two PCI-X slots
and my Eicon Diva 4BRI-8M, which should work fine with Asterisk, is only
PCI 2.0 standard and does not fit into the PCI-X slot.

Currently I use two AVM Fritz! cards, but while my Xeon 604 2000MHz  had
a load-average of less then 0.5 it is now increasing to more then 4,  if
I have 20 VoIP and two ISDN calls.

So, the cheap AVM cards have to replaced.

Since I do not have currenly the money to buy a Eicon  Diva  Server  3.0
quad-port, can you recommend me inexpensive  Dual-  or  Quad-Port  cards
which I could get used on eBay?

Note:   I have to connect my Alice Box (ADSL2+) and two Vodafone
EasyBox 803A (using the Huawei K3765-HV USB-Stick) to it.

Thanks, Greetings and nice Day/Evening
Michelle Konzack

-- 
# Debian GNU/Linux Consultant ##
   Development of Intranet and Embedded Systems with Debian GNU/Linux
   Internet Service Provider, Cloud Computing


itsystems@tdnet Jabber  linux4miche...@jabber.ccc.de
Owner Michelle Konzack

Gewerbe Strasse 3   Tel office: +49-176-86004575
77694 Kehl  Tel mobil:  +49-177-9351947
Germany Tel mobil:  +33-6-61925193  (France)

USt-ID:  DE 278 049 239

Linux-User #280138 with the Linux Counter, http://counter.li.org/


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[asterisk-users] Digium's new Community Support Manager - Rusty Newton

[Kevin P. Fleming]

We'd like you all to help us welcome Rusty Newton to Digium's Asterisk
development and community support team! Rusty has been with Digium for
over five years, starting in the Technical Support department and then
moving to a sales position where he assisted customers with Asterisk and
Switchvox solutions to their business needs. Prior to joining Digium he
spent more than five years in the telecom industry, installing,
configuring and maintaining PBXs. A couple of weeks ago he moved into a
new role (for him and for Digium), Community Support Manager.

In this role he'll be the primary person responsible for ensuring that
Digium's community services are providing what the community members
need, that the systems are operating properly, and that issues and
questions are getting the attention they deserve. He'll be working
closely with our Community Director as well, especially for events like
AstriCon and others. He works directly with the software development
team at Digium, which will allow him to focus almost exclusively on
technical issues and discussions.

We're quite excited that he has taken on this role and we expect that
you will soon see the benefits of his activities across the community!

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file

[Michael L. Young]

- Original Message - 

> From: "Jayesh Labade" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Friday, May 25, 2012 2:09:58 AM
> Subject: Re: [asterisk-users] Asterisk MixMonitor starts recording 44
> bytes file

> Hello Michael,

> Thanks a lot for your immediate help. After applying patch MixMonitor
> started works normally,

> I can understand that this can be Happen in asterisk 10.4 but as a
> stable and Long support version 1.8.12.0 this should not happen. I
> got same error in both version.

> Anyways this patch solved my problem.

Jayesh,

Glad to hear that the information helped figure out what was going on and also 
provided a fix.

In the 1.8 line, this has been fixed as well and will be in future releases.

In an ideal world, there would be no bugs in software.  LTS doesn't mean "bug 
free".  It means that it will be supported over a longer period of time which 
should result in more real world use and more bug fixing resulting in a more 
stable product with time.

Regards,
Michael

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Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk and how many servers

[bilal ghayyad]

Hi John;

For 20,000 users: Is it better to use Asterisk realtime configuration or to use 
conf files?

I readed the below link but did not understand which GUI that works with 
asterisk realtime?

http://www.freepbx.org/trac/wiki/AsteriskRealtime


Regards
Bilal


> 
> > My question is:
> >
> > Is it really possible to have the asterisk
> configuration in the database server instead of having it in
> conf files? HOW? I am asking this because what I noticed in
> AsteriskNow and in A2Billing and Vicidial or Goautodial that
> whatever I do configuration in the GUI, then the
> configuration will be generated in the conf files, so
> Asterisk will read from the conf files and not from the
> database directly. Is it right or I am confused and there is
> something else?
> >
> > If there is a method to let the configuration to be
> taken from the database (and not from the configuration),
> then HOW? Because even in AsteriskNow, the configuration
> will be generated in a conf files.
> 
> Hi Bilal,
> 
> You want to look the Asterisk realtime configuration
> features if you 
> want to run your configuration from a database rather than
> configuration 
> files.
> 
> This should point you in the right direction and get you
> started:  
> http://www.voip-info.org/wiki/view/Asterisk+RealTime
> 
> It should be noted that if you're wanting to use AsteriskNow
> (which 
> relies on FreePBX for its gui configuration features), then
> Asterisk 
> realtime configuration will not work as it is not compatible
> at this 
> time.  Other web gui's might work, but I am not
> familiar with them.  
> FreePBX's sentiment on the subject is shared here:  
> http://www.freepbx.org/trac/wiki/AsteriskRealtime
> 
> -John
> 
> On 05/24/2012 05:46 PM, bilal ghayyad wrote:
> > Thanks for all for the help and kindly reply.
> >
> > One last point that will help me alot:
> >
> > I am thinking to have 4 Servers running Asterisk and 2
> Servers to be for database. The load to be distributed on
> the 4 Asterisk Servers with ability to be redundant (using
> any redundancy technique). The 4 Asterisk Servers to take
> the configuration from the Database Server and actually
> because there is 2 Database servers, then it will be
> redundant to each other (in case one database failed, the
> other will take over).
> >
> > My question is:
> >
> > Is it really possible to have the asterisk
> configuration in the database server instead of having it in
> conf files? HOW? I am asking this because what I noticed in
> AsteriskNow and in A2Billing and Vicidial or Goautodial that
> whatever I do configuration in the GUI, then the
> configuration will be generated in the conf files, so
> Asterisk will read from the conf files and not from the
> database directly. Is it right or I am confused and there is
> something else?
> >
> > If there is a method to let the configuration to be
> taken from the database (and not from the configuration),
> then HOW? Because even in AsteriskNow, the configuration
> will be generated in a conf files.
> >
> > Special thanks for the advise.
> >
> > Regards
> > Bilal
> > -
> >
> >>> Hi All;
> >>>
> >>> I need to use Asterisk for 20 000 users, so
> which
> >> asterisk version to be used? Is there asterisk
> version that
> >> supports 20,000 users on one hardware machine?
> >>> Can I use one strong hardware server i7 with 64
> GB RAM
> >> and fast hard desk to handle 20 000 users, and
> concurrent
> >> calls 2000? Or I need multiple servers, how much?
> >>> If I am going to use multiple servers (until
> now I do
> >> not know how much, and I do not know if the barrier
> will be
> >> the asterisk software or the hardware), then do I
> have to
> >> use special SIP proxy or I have to use load
> balancer)? In
> >> this case, I have to use asterisk Database (so all
> the
> >> servers will read/write from the database)?
> >>> What about AsteriskNow, can it support?
> >> AsteriskNOW is a GUI on top of Asterisk; it does
> not change
> >> the ability
> >> of the system to handle call load.
> >>
> >> Modern versions of Asterisk can easily handle
> 2,000
> >> simultaneous calls,
> >> even with media (non-transcoded) passing through
> the server.
> >> We have a
> >> community member who has improved chan_sip in
> Asterisk 10
> >> (and later) to
> >> be able to handle 10,000 simultaneous calls.
> >>
> >> Handling 20,000 registrations is probably more of a
> concern
> >> for Asterisk
> >> at this point; I've never heard of anyone
> attempting to
> >> handle that many
> >> on one system.
> >>
> >> In spite of all this, though, the other advice
> you've
> >> received in this
> >> thread is sound: even if a single system can handle
> the
> >> load, doing so
> >> is asking for a major problem if that system
> experiences a
> >> failure.
> >> You'd be much better off to at least split the load
> across
> >> two machines,
> >> both of which should be large enough to handle the
> entire
> >> load when
> >> necessary.
> >>
> >> -- 
> >

Re: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file

[A J Stiles]

On Thursday 24 May 2012, Jayesh Labade wrote:
> Hello All,
> 
> I have installaed asterisk 10.4 in my machine. Now suddenly MixMonitor
> application starts generating 44 Bytes of Recording file.
> Is this new tye of Bug? Help me..

44 bytes is very interesting, as this is the canonical length of a .wav 
header.  It sounds as though something is crashing immediately after the 
recording starts, leaving no data.

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Disable All Asterisk Features (blind xfer, disconnect, etc)

[[Digital^Dude] ®]

You can unload the features module maybe

On Wed, May 23, 2012 at 9:18 PM, Eduardo Pimenta
wrote:

> Hi Guys,
>
> is there any way to disable all Asterisk Features? We are having false
> dtmf detections and randon calls being put on-hold and suspect that dtmf
> features is the cause.
>
> Changing features.conf aparently keeps the default options. Since we dont
> use it, is there any way to disable it?
>
>
> Thanks,
>
> Eduardo
>
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[asterisk-users] Asterisk Atxfer

[[Digital^Dude] ®]

Hello,

I need to return to the original call leg that I wanted to transfer the
call to. in case the destination IVR has put me in a rather long queue.
Please suggest a way I can hang up the atxfer leg and return to the first
call leg.
The hangup parameter in dial app using '*' key works only till the
destination is ringing...

Please share possible ways I could go about this problem.

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[asterisk-users] Function not Registered??

[Lee, John (Sydney)]

Hi all,



I am running the same Asterisk 1.4.21.2 with the same configuration on all the 
servers in the region.

I got this function called func_devstate which I use to control the lights of 
the Polycom phones.

This module works well for all the Asterisk servers except this one.



To get it to work, I basically compile this module together with the others and 
there is no need to explicitly load it in modules.conf.

The problem is when my script uses function DEVSTATE, the Asterisk console 
shows that it is not registered.



However, when I did a module show, it was there.



I did restart Asterisk or include it in module.conf but it did not resolve the 
problem.



Do you have any clues why this is happening?

Thanks in advance.



-- Executing [*1223*1**1900@incoming:78] Set("SIP/1900-08ee1da8", 
"DEVSTATE(Custom:cfalw1900)=INUSE") in new stack
[May 25 11:59:46] ERROR[8913]: pbx.c:1564 ast_func_write: Function DEVSTATE not 
registered



/usr/lib/asterisk/modules/func_devstate.so
/usr/src/asterisk-1.4.21.2/funcs/.func_devstate.makeopts
/usr/src/asterisk-1.4.21.2/funcs/.func_devstate.moduleinfo
/usr/src/asterisk-1.4.21.2/funcs/.func_devstate.o.d
/usr/src/asterisk-1.4.21.2/funcs/func_devstate.c
/usr/src/asterisk-1.4.21.2/funcs/func_devstate.o
/usr/src/asterisk-1.4.21.2/funcs/func_devstate.so



*CLI> module show like func_devstate.so

Module Description Use Count
func_devstate.so Gets or sets a device state in the dialp 0
1 modules loaded

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