Re: [asterisk-users] Converting GSM calls to SIP

[Vahan Yerkanian]

On 4/15/10 1:26 AM, Tonty T wrote:
That's is all the overhead I am trying to avoid.  What I need is a DID 
with unlimited channel, but they do not offer DIDs in that country.  I 
wanted to know for example when I get a DID from lets say Vitelity, 
with unlimited channel, what are they using to forward the calls via 
SIP or IAX to my server?  If I knew the details of the process, I 
could probably tell them to used this method and route the short code 
to me via SIP.  And if it requires hardware I could invest in it 
myself and have them host it.


If their switch doesn't support SIP or doesn't have SIP module 
installed, there isn't much you can do to get traffic in pure SIP form. 
Ask them if they can and willing to serve you the traffic via multiple 
E3 or even better, STM fiber links. STM over fiber is the cheapest way 
to transport that much channels by means of cabling - you just need 2 
strands for TX/RX or even 1 strand if you go with WDM. However the 
carrier crade hardware for it is *very expensive*. On your side you 
demux STM link(s) into E3/E1s using expensive carrier grade equipment 
like Cisco's $25k+ (used) STM cards for Cisco 7500 and up models or if 
you're smart enough to know where to dig, dirt cheap (~$2K for STM-1 to 
24E1) Taiwanese/Chinese media converters.


Oh and yes, this isn't a task for a single Asterisk server. The most 
I've seen a single box capable of is 16 E1s (2 x 8E1 cards) in a single 
chassis doing only G711a to SIP conversion.


HTH,
Vahan


On Wed, Apr 14, 2010 at 2:46 PM, Jeff Brower > wrote:


> On Wed, Apr 14, 2010 at 10:33 AM, Tonty T mailto:ton...@gmail.com>> wrote:
>
>> This is a solution they proposed, using GSM gateways, but it
wont let me
>> handle 1000 simultaneous calls, the other option was using an
E1 but the
>> cost would be too much to deploy 35 E1s to support that many
calls.  There
>> might be a better way of doing it.
>>
>>
> If you are planning on having 1000 simultaneous calls, you're
going to be
> looking at a hefty price tag one way or the other.  Things to
consider - if
> you're going to have 1000 concurrent calls going out over VoIP
trunks (SIP /
> IAX / whatever), you need to have enough bandwidth to
comfortably handle
> that many calls (each g729 is 8Kb/s bandwidth (but you need to
pay a license
> fee for each channel of g729), each g711alaw is 64Kb/s, etc).
That amount
> of bandwidth won't be cheap, plus the cost of the ITSP giving
your 1000
> concurrent channels to call on.  On the other hand, if you have
a bank of
> E1's, which support (I think) at max 30 concurrent voice
channels, you'd
> need 34 available E1 spans.  I'm not sure if you can get 34
spans working in
> a single asterisk server (there was some discussion about this
recently on
> this list), and you'd have the cost of 34 E1 spans as well.

All good points.  It might be worth mentioning that including
IP/UDP/RTP packet overhead, actual bandwidth is 40 kbps
for G729 and 96 kbps for G711.

-Jeff



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Re: [asterisk-users] [Conference] Audio/Video

[Ioan Indreias]

We have used with success BBB (BigBlueButton - open source -
http://bigbluebutton.org) and I recommend to try their demo in order
to see if this solution gives all you need.

Voice conf is based on Asterisk.

HTH,
Ioan Indreias
www.modulo.ro

On Thu, Apr 15, 2010 at 2:04 AM, Stéphane Bauland  wrote:
> Le 04/15/2010 12:11 AM, Hans Witvliet a écrit :
>> On Wed, 2010-04-14 at 22:55 +0200, Stéphane Bauland wrote:
>>> Hi guys,
>>>
>>> I'm planning of creating a speech/video conference application. This
>>> application will provide a system to see/listen to each personn present
>>> in the conference.
>>>
>>> So each ppl will have a audio and video stream.
>>>
>>> I'm wondering if you know a way to do this with asterisk or if it's
>>> supported ?
>>>
>>> If it is, i'm asking you about some documentation or related article (if
>>> you know ones) where i could find more informations.
>>>
>>> Else, do you know any other way to do this ?
>>>
>>> Best regards,
>>>
>>
>> Would love to see a _working_ video conf.
>> afaicr it's currently vapor-ware
>> Are you thinking of letting asterisk doing video multiplexing?
>> Or are you aiming just for a conference with a small number of
>> participants?
>>
>> hw
>>
>
> We (cause we are a team) are planning of doing a multi user conference
> software at a end school project.
>
> The way we go is, we are looking throught jungle (xmpp ext for jabber)
> to create conference between many people. We don't want to set a
> limitation about "how many participant of a conference).
>
> But right now, i'm discovering asterisk, and i need some informations
> from people like you that know the soft and his capatibilities...
>
> So i think yes, we want to do video multiplexing.
>
> Do you think a software like that could use asterisk as a backend ?
>
> And, do you know any other software that is doing the same thing using
> asterisk ?
>
> --
> Stéphane Bauland
>

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Re: [asterisk-users] [Conference] Audio/Video

[Jamie A. Stapleton]

http://www.projectdiastar.org/ looks promising...

On Apr 14, 2010, at 7:04 PM, Stéphane Bauland wrote:

Le 04/15/2010 12:11 AM, Hans Witvliet a écrit :
On Wed, 2010-04-14 at 22:55 +0200, Stéphane Bauland wrote:
Hi guys,

I'm planning of creating a speech/video conference application. This
application will provide a system to see/listen to each personn present
in the conference.

So each ppl will have a audio and video stream.

I'm wondering if you know a way to do this with asterisk or if it's
supported ?

If it is, i'm asking you about some documentation or related article (if
you know ones) where i could find more informations.

Else, do you know any other way to do this ?

Best regards,


Would love to see a _working_ video conf.
afaicr it's currently vapor-ware
Are you thinking of letting asterisk doing video multiplexing?
Or are you aiming just for a conference with a small number of
participants?

hw


We (cause we are a team) are planning of doing a multi user conference
software at a end school project.

The way we go is, we are looking throught jungle (xmpp ext for jabber)
to create conference between many people. We don't want to set a
limitation about "how many participant of a conference).

But right now, i'm discovering asterisk, and i need some informations
from people like you that know the soft and his capatibilities...

So i think yes, we want to do video multiplexing.

Do you think a software like that could use asterisk as a backend ?

And, do you know any other software that is doing the same thing using
asterisk ?

--
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CBSi - Connecting your problems with solutions.
Telephone:  (804) 412-1601
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Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)

[Goke M Aruna]

Hi Huu,

Asterisk support ss7.
Check chan_ss7 and libss7, both project are active and working like charm.

Thanks

On 4/15/10, huu giang  wrote:
> Dear Goke,
>
> I don't use ISDN to connect to MSC, it connect to ISDN network.
> There are other people deploy IVR using this protocol.
>
> About ISUP/SS7, supporting technical from the vendor I bought Sangoma Card,
> they said that If I want to connect to MSC, I have to use ISUP/SS7 protocl,
> but now Asterisk doesn't support SS7 protocol and I have to buy a SS7
> package to install on Asterisk Server so Astersik can work with SS7.
>
> Is it right ?, It is the first time I deploy Asterisk, so please consult me.
>
> Thanks
>
> Hiện tại,
> nếu anh dùng luồng
> ISDN thì không cần báo hiệu SS7 nhưng anh muốn kết nối trực tiếp vào MSC
>  của Telco(SS7)
> là anh cần phải có giao thức SS7 để chúng bắt tay làm việc, nhưng hiện
> tại trên
> tổng đài soft không hổ trợ, anh cần mua gói phần mền SS7 để cài đặt vào
> tổng
> đài để chúng làm việc với giao thức SS7.
>
> --- On Wed, 4/14/10, Goke M Aruna  wrote:
>
> From: Goke M Aruna 
> Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core
> network (MSC)
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Date: Wednesday, April 14, 2010, 4:24 AM
>
> hello Huu,
>
> Can you share their explanation with me at least, I can gain from it too.
>
> Thanks
>
> On Wed, Apr 14, 2010 at 10:01 AM, huu giang  wrote:
>
>
> Hi Goke,
>
> Some experienced people said me to use ISDN to connect to MSC.
>
> Thanks very much.
>
>
> --- On Wed, 4/14/10, Goke M Aruna  wrote:
>
>
> From: Goke M Aruna 
> Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core
> network (MSC)
>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Date: Wednesday, April 14, 2010, 1:50 AM
>
>
> Hello Huu,
>
> use E1/SS7 signaling or if you MSC speak SIP, then use SIP.
>
> Thanks
>
> On Tue, Apr 13, 2010 at 11:46 AM, huu giang  wrote:
>
>
>
>
> Hi all,
>
> My Asterisk connect to GSM core network (connect directly to MSC) through E1
> lines. What the kind of protocol is used ?. It is ISUP/SS7 protocol ?
>
> Thanks in advance
>
>
>
>
>
>
>
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[asterisk-users] How can I record the conversations in a conference call?

[Renato bianchini]

Hello,

I wanna record the conversations in a conference call, anyone know how can I do 
it? I've already configurated a room on meetme.conf but I don't know as I can 
record the conversations.

I'm using SUSE 11 and Asterisk 1.6.2.

Thank you so much for help me.

Bye





  

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Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)

[huu giang]

Dear Goke,

I don't use ISDN to connect to MSC, it connect to ISDN network.
There are other people deploy IVR using this protocol.

About ISUP/SS7, supporting technical from the vendor I bought Sangoma Card, 
they said that If I want to connect to MSC, I have to use ISUP/SS7 protocl, but 
now Asterisk doesn't support SS7 protocol and I have to buy a SS7 package to 
install on Asterisk Server so Astersik can work with SS7.

Is it right ?, It is the first time I deploy Asterisk, so please consult me.

Thanks

Hiện tại, 
nếu anh dùng luồng
ISDN thì không cần báo hiệu SS7 nhưng anh muốn kết nối trực tiếp vào MSC
 của Telco(SS7)
là anh cần phải có giao thức SS7 để chúng bắt tay làm việc, nhưng hiện 
tại trên
tổng đài soft không hổ trợ, anh cần mua gói phần mền SS7 để cài đặt vào 
tổng
đài để chúng làm việc với giao thức SS7. 

--- On Wed, 4/14/10, Goke M Aruna  wrote:

From: Goke M Aruna 
Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core 
network (MSC)
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Wednesday, April 14, 2010, 4:24 AM

hello Huu,

Can you share their explanation with me at least, I can gain from it too.

Thanks

On Wed, Apr 14, 2010 at 10:01 AM, huu giang  wrote:


Hi Goke,

Some experienced people said me to use ISDN to connect to MSC. 

Thanks very much.


--- On Wed, 4/14/10, Goke M Aruna  wrote:


From: Goke M Aruna 
Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core 
network (MSC)

To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Wednesday, April 14, 2010, 1:50 AM


Hello Huu,

use E1/SS7 signaling or if you MSC speak SIP, then use SIP.

Thanks

On Tue, Apr 13, 2010 at 11:46 AM, huu giang  wrote:




Hi all,

My Asterisk connect to GSM core network (connect directly to MSC) through E1 
lines. What the kind of protocol is used ?. It is ISUP/SS7 protocol ?

Thanks in advance






  
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Re: [asterisk-users] FastAGiin Windows Server

[David Backeberg]

On Wed, Apr 14, 2010 at 8:55 PM, Edwin Quijada
 wrote:
>
> My problem is that I need to execute windows app using IVR in Asterisk so we

What is the windows app that you cannot replace on Linux?

How about wrapping THAT program with simple inputs and outputs, and
build a network interface on top of it, then bounce interface calls
back and forth from linux?

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Re: [asterisk-users] FastAGiin Windows Server

[Edwin Quijada]

My problem is that I need to execute windows app using IVR in Asterisk so we 
need FastAGI using perl. I saw Asterisk::fastagi but everything for this is in 
Linux and i dont know if it works in windows.

 

I need to know if somebody has used fastagi in windows with perl becuase I have 
a lot of agi in perl 

 TIA
> Date: Wed, 14 Apr 2010 10:04:27 -0700
> From: asterisk@sedwards.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] FastAGiin Windows Server
> 
> On Wed, 14 Apr 2010, Edwin Quijada wrote:
> 
> > I wanna know if can run my AGI scripts as fastAGI scripts in Windoes 
> > server.
> 
> Seems like a move in the wrong direction to me, but no 
> -- you can't run an AGI script via fastagi() without changes.
> 
> > I need a lot of script done in perl and I wanna move to windows server. 
> > I checked Asterisk::fastagi but I see that everything is for Linux.
> 
> Fastagi is a protocol. You could implement it in most languages on most 
> OSs.
> 
> (Everything is for Linux because that's where "server stuff" belongs.)
> 
> > Somebody has idea to do this in perl. I dont want to change the 
> > language.
> 
> Somebody should re-think their ideas :)
> 
> Saying you "need a lot of script done" implies you haven't done it yet. 
> I'd suggest changing your language to C. You can execute XXX AGIs written 
> in C in the time it takes to load Perl and parse your script. Maybe you 
> wouldn't even need to use a separate server.
> 
> -- 
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
> 
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[asterisk-users] Regarding remote registration of SIP user on zoiper

[Vinod Parameswaran]

Hello list,

I am new to this list and have been using Asterisk as part of my research 
project for about 2 weeks now.

I would like to get your thoughts on a scenario that I am attempting at the 
moment. I haven't had luck until now.

In this scenario, I am trying to register a SIP user configured on the zoiper 
client installed on a laptop, which is on the same Local Area Network, with the 
Asterisk server running on a different Linux box.
On the same Linux box, there is another zoiper client running with 2 users who 
are registered with the Asterisk server on the local domain.

The IP address of the Linux box is static. The SIP port configured on this box 
for incoming requests is 5060.

The configuration for the SIP user on zoiper client reads as follows:
(I have only mentioned below those settings that have been enabled)
Domain : 
Username: 
Password: 
Caller ID Name: 
Registration Expiry: 3600
SIP Port: 5060
Use default STUN
Use DTMF RFC-2833

On the Linux box, I have the following entry for the user in sip.conf:

[]
type=friend
host=
fromuser=
context=outgoing

On the Linux box, I have the following entry for the user in extensions.conf:

[outgoing]
exten =>102,1,Dial(SIP/)
exten =>102,n,Congestion(7)
exten =>102,n,Hangup()

[phones]
include => internal
include => outgoing

With the above configuration, I intend to achieve two objectives:

1. To register the user  configured on the zoiper client that is 
installed on the Laptop, with the Asterisk server on the Linux Box. Both the 
Laptop and the Linux Box are on the same LAN.
2. To successfully place a call from one of the users configured on the zoiper 
client installed on the Linux box and registered with the Asterisk server on 
the local domain, to the user  configured on the zoiper client that 
is installed on the Laptop.

However, my observation has been that the zoiper client running on the laptop 
tries to register the user  with the Asterisk server on the Linux 
box without success.
I tried placing a call to the user  configured on the zoiper client 
that is installed on the Laptop, from the zoiper client running on the Linux 
box, but that failed with the error "no route to destination". Based on this 
observation, I presume that there is some issue related to the zoiper client on 
the laptop being able to talk to the Asterisk server on the Linux box, and vice 
versa. I have been able to successfully ping the Linux box from the laptop and 
the laptop from the Linux box.

I would appreciate if you could help me with your suggestions in debugging this 
scenario.

Thanks and regards
Vinod

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[asterisk-users] Queue call to specific queuemember

[Asterisk Maniac]

Hi all,
  What would be the best way to send a call to a queue as usual, but telling
that it should be awsered by some specific member?
Thanks already
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Re: [asterisk-users] [Conference] Audio/Video

[David Backeberg]

On Wed, Apr 14, 2010 at 4:55 PM, Stéphane Bauland  wrote:
> I'm planning of creating a speech/video conference application. This
> application will provide a system to see/listen to each personn present
> in the conference.
> Else, do you know any other way to do this ?

http://en.wikipedia.org/wiki/CU-SeeMe
it was kindof a solved problem,
but that's not really around anymore.

these days, ichat and google chat and Ekiga do one-on-one chat well.

The problem is n-to-n chat.

Take a look at openmcu, and good luck.

Unfortunately, the products that work well AND are turnkey generally
require money, ranging from a little to literally millions for a
full-featured Cisco telepresence solution.

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Re: [asterisk-users] [Conference] Audio/Video

[Stéphane Bauland]

Le 04/15/2010 12:11 AM, Hans Witvliet a écrit :
> On Wed, 2010-04-14 at 22:55 +0200, Stéphane Bauland wrote:
>> Hi guys,
>>
>> I'm planning of creating a speech/video conference application. This
>> application will provide a system to see/listen to each personn present
>> in the conference.
>>
>> So each ppl will have a audio and video stream.
>>
>> I'm wondering if you know a way to do this with asterisk or if it's
>> supported ?
>>
>> If it is, i'm asking you about some documentation or related article (if
>> you know ones) where i could find more informations.
>>
>> Else, do you know any other way to do this ?
>>
>> Best regards,
>>
>
> Would love to see a _working_ video conf.
> afaicr it's currently vapor-ware
> Are you thinking of letting asterisk doing video multiplexing?
> Or are you aiming just for a conference with a small number of
> participants?
>
> hw
>

We (cause we are a team) are planning of doing a multi user conference
software at a end school project.

The way we go is, we are looking throught jungle (xmpp ext for jabber)
to create conference between many people. We don't want to set a 
limitation about "how many participant of a conference).

But right now, i'm discovering asterisk, and i need some informations
from people like you that know the soft and his capatibilities...

So i think yes, we want to do video multiplexing.

Do you think a software like that could use asterisk as a backend ?

And, do you know any other software that is doing the same thing using
asterisk ?

-- 
Stéphane Bauland

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Re: [asterisk-users] Conference Meetme

[Philipp von Klitzing]

Hi!

> I need the server to handle about 300 - 400 simultaneous meetme 
> conferences, 5-10 participants in each, 
> 
> Actually I need to know, if I will get an IBM X3650 M2,QuadCore, 4-6
> GB RAM, 8MB cache, how many simultaneous meetme conferences I can
> operate on a this server. 

There is no simple answer for you - look here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+dimensioning

Philipp


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[asterisk-users] [Conference] Audio/Video

[Stéphane Bauland]

Hi guys,

I'm planning of creating a speech/video conference application. This
application will provide a system to see/listen to each personn present
in the conference.

So each ppl will have a audio and video stream.

I'm wondering if you know a way to do this with asterisk or if it's
supported ?

If it is, i'm asking you about some documentation or related article (if
you know ones) where i could find more informations.

Else, do you know any other way to do this ?

Best regards,

-- 
Stéphane Bauland

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[asterisk-users] Interpbx connection

[khalid touati]

Hi Guys,
i've connecting two pbx server successfully for several times using the
following config:

register => USPBX:myp...@122.11.176.35 

[PBX1]
type=friend
host=122.11.176.35
trunk=yes
sercret=mypass
context=external
deny=0.0.0.0/0.0.0.0
permit=122.11.176.35/255.255.255.240
insecure=very
allow=all
nat=yes
qualify=yes
canreinvite=no

in the other and it's the analog.

but now i can only dial from one end, and the other en d is giving me this
error.

Apr 14 16:44:21 ERROR[26502]: chan_sip.c:6659 register_verify: Peer 'PBX1'
is trying to register, but not configured as host=dynamic

when dialing a fast busy signal and it sauys in the CLI: CONGESTION. any
help please!!!

-- 
Abdullah
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Re: [asterisk-users] Converting GSM calls to SIP

[Tonty T]

That's is all the overhead I am trying to avoid.  What I need is a DID with
unlimited channel, but they do not offer DIDs in that country.  I wanted to
know for example when I get a DID from lets say Vitelity, with unlimited
channel, what are they using to forward the calls via SIP or IAX to my
server?  If I knew the details of the process, I could probably tell them to
used this method and route the short code to me via SIP.  And if it requires
hardware I could invest in it myself and have them host it.


On Wed, Apr 14, 2010 at 2:46 PM, Jeff Brower  wrote:

> > On Wed, Apr 14, 2010 at 10:33 AM, Tonty T  wrote:
> >
> >> This is a solution they proposed, using GSM gateways, but it wont let me
> >> handle 1000 simultaneous calls, the other option was using an E1 but the
> >> cost would be too much to deploy 35 E1s to support that many calls.
>  There
> >> might be a better way of doing it.
> >>
> >>
> > If you are planning on having 1000 simultaneous calls, you're going to be
> > looking at a hefty price tag one way or the other.  Things to consider -
> if
> > you're going to have 1000 concurrent calls going out over VoIP trunks
> (SIP /
> > IAX / whatever), you need to have enough bandwidth to comfortably handle
> > that many calls (each g729 is 8Kb/s bandwidth (but you need to pay a
> license
> > fee for each channel of g729), each g711alaw is 64Kb/s, etc). That amount
> > of bandwidth won't be cheap, plus the cost of the ITSP giving your 1000
> > concurrent channels to call on.  On the other hand, if you have a bank of
> > E1's, which support (I think) at max 30 concurrent voice channels, you'd
> > need 34 available E1 spans.  I'm not sure if you can get 34 spans working
> in
> > a single asterisk server (there was some discussion about this recently
> on
> > this list), and you'd have the cost of 34 E1 spans as well.
>
> All good points.  It might be worth mentioning that including IP/UDP/RTP
> packet overhead, actual bandwidth is 40 kbps
> for G729 and 96 kbps for G711.
>
> -Jeff
>
>
>
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[asterisk-users] Vestec vs Lumenvox

[Danny Nicholas]

Hi listers,

  I'm a 1.4 holdout who uses mostly Suse platforms.  I bit the
bullet and installed a Centos box to get Lumenvox up and running, but now
see this "new" offering of Vestec that supports OpenSuse and Windows in
addition to the Platforms supported by Lumenvox.  Anybody out there working
with Vestec that can give me a heads up on how it works or doesn't?

 

Thanks

Danny Nicholas

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Re: [asterisk-users] Conference Meetme

[torintino1]

I need the server to handle about 300 - 400 simultaneous meetme conferences, 
5-10 participants in each,

Actually I need to know, if I will get an IBM X3650 M2, QuadCore, 4-6 GB RAM, 
8MB cache,
how many simultaneous meetme conferences I can operate on a this server.

Thanks



From: Zeeshan Zakaria 
Sent: Wednesday, April 14, 2010 8:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Subject: Re: [asterisk-users] Conference Meetme


Last year I did a lab test for a customer who wanted conferencing solution for 
his organization, on a 2 x dual core xeon with 4GB type server, which had 120 
zap channels and I put all the channels in mutiple conferences, from 4 to 20 
users per conference and let it running for two weeks. Munin graph showed that 
CPU load was only 6 to 7 percent during this period, no conference dropped and 
asterisk didn't crash, and I occasionally used it to make calls and run other 
processes, and no call quality issues. Now that server is in production and 
customer is happy with it. I don't know about SIP which will use more 
processing, but I am sure a decent server of today can handle a good number of 
conferences with a good number of users each. What numbers you are looking for?

Zeeshan A Zakaria

--
Sent from my Android phone with K-9 Mail.


  On 2010-04-14 12:56 PM, "Steve Edwards"  wrote:


  On Wed, 14 Apr 2010, torinti...@hotmail.com wrote:

  > How many simultaneous conference meetme setups...

  How long is a piece of string?

  0) A better subject yields better answers

  1) A more detailed question yields a more detailed answer.

  A reasonably configured Asterisk server can handle XXX callers with Y
  callers in each of Z conferences.

  --
  Thanks in advance,
  -
  Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
  Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Converting GSM calls to SIP

[Jeff Brower]

> On Wed, Apr 14, 2010 at 10:33 AM, Tonty T  wrote:
>
>> This is a solution they proposed, using GSM gateways, but it wont let me
>> handle 1000 simultaneous calls, the other option was using an E1 but the
>> cost would be too much to deploy 35 E1s to support that many calls.  There
>> might be a better way of doing it.
>>
>>
> If you are planning on having 1000 simultaneous calls, you're going to be
> looking at a hefty price tag one way or the other.  Things to consider - if
> you're going to have 1000 concurrent calls going out over VoIP trunks (SIP /
> IAX / whatever), you need to have enough bandwidth to comfortably handle
> that many calls (each g729 is 8Kb/s bandwidth (but you need to pay a license
> fee for each channel of g729), each g711alaw is 64Kb/s, etc). That amount
> of bandwidth won't be cheap, plus the cost of the ITSP giving your 1000
> concurrent channels to call on.  On the other hand, if you have a bank of
> E1's, which support (I think) at max 30 concurrent voice channels, you'd
> need 34 available E1 spans.  I'm not sure if you can get 34 spans working in
> a single asterisk server (there was some discussion about this recently on
> this list), and you'd have the cost of 34 E1 spans as well.

All good points.  It might be worth mentioning that including IP/UDP/RTP packet 
overhead, actual bandwidth is 40 kbps
for G729 and 96 kbps for G711.

-Jeff



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Re: [asterisk-users] Conference Meetme

[Zeeshan Zakaria]

Last year I did a lab test for a customer who wanted conferencing solution
for his organization, on a 2 x dual core xeon with 4GB type server, which
had 120 zap channels and I put all the channels in mutiple conferences, from
4 to 20 users per conference and let it running for two weeks. Munin graph
showed that CPU load was only 6 to 7 percent during this period, no
conference dropped and asterisk didn't crash, and I occasionally used it to
make calls and run other processes, and no call quality issues. Now that
server is in production and customer is happy with it. I don't know about
SIP which will use more processing, but I am sure a decent server of today
can handle a good number of conferences with a good number of users each.
What numbers you are looking for?

Zeeshan A Zakaria

--
Sent from my Android phone with K-9 Mail.

On 2010-04-14 12:56 PM, "Steve Edwards"  wrote:

On Wed, 14 Apr 2010, torinti...@hotmail.com wrote:

> How many simultaneous conference meetme setups...
How long is a piece of string?

0) A better subject yields better answers

1) A more detailed question yields a more detailed answer.

A reasonably configured Asterisk server can handle XXX callers with Y
callers in each of Z conferences.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Converting GSM calls to SIP

[Warren Selby]

On Wed, Apr 14, 2010 at 10:33 AM, Tonty T  wrote:

> This is a solution they proposed, using GSM gateways, but it wont let me
> handle 1000 simultaneous calls, the other option was using an E1 but the
> cost would be too much to deploy 35 E1s to support that many calls.  There
> might be a better way of doing it.
>
>
If you are planning on having 1000 simultaneous calls, you're going to be
looking at a hefty price tag one way or the other.  Things to consider - if
you're going to have 1000 concurrent calls going out over VoIP trunks (SIP /
IAX / whatever), you need to have enough bandwidth to comfortably handle
that many calls (each g729 is 8Kb/s bandwidth (but you need to pay a license
fee for each channel of g729), each g711alaw is 64Kb/s, etc).  That amount
of bandwidth won't be cheap, plus the cost of the ITSP giving your 1000
concurrent channels to call on.  On the other hand, if you have a bank of
E1's, which support (I think) at max 30 concurrent voice channels, you'd
need 34 available E1 spans.  I'm not sure if you can get 34 spans working in
a single asterisk server (there was some discussion about this recently on
this list), and you'd have the cost of 34 E1 spans as well.

Just some additional things for you to consider when building out your
solution.

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] 1.6.0 verses 1.6.2

[John Rose]

Hello,

I can reproduce the results always.

Hardware: 
E-5200 2.5 Ghz
4 gig ram
Centos 5.4

I've tested most versions from 1.6.0.11 to 1.6.0.26.
Compared to most version of the 1.6.1.x and 1.6.2.x.

I originate calls from one box using FastAGI to another Asterisk box
that is the call sink.

1.6.1.x and 1.6.2.x always start using hi CPU at around 160+ calls and
coredumps soon afterward at 180-210 calls depending on what exactly is
happening on the call. Then I switch over to the 1.6.0.x version on the
same box with same configs (just do a make install) and can get 400+
simultaneous calls easily and sometimes over 500 before the cpu is too
high.

The call sink is just playing a few files then waiting then sending a
DTMF. The call generator is doing the same playing a couple files, not
much else.

1.6.1 and 1.6.2 suck up more cpu apparently for some reason than version
1.6.0...

John


> -Original Message-
> 
> On Tue, Apr 13, 2010 at 04:25:49PM -0600, John Rose wrote:
> > Why do versions 1.6.2 and 1.6.1 use much more CPU resources that
> 1.6.0?
> > I can get 400+  SIP/G.711
> >
> > calls running on this dual core box with 1.6.0 but the cpu maxes out
> and
> > core dumps at approx. 180 calls when version 1.6.1/2 is running.
> 
> Could you please be more specific? Have you tested those versions on
> the
> same system? Can you reproduce those results?
> 
> What exact versions? On what system?
> 
> --
>Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
> 
> --

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[asterisk-users] Sending SMS problems.

[Agazzini Maurizio]

Dear Sir,

I'm trying to configure the SMS capabilities of my provider (Telecom
Italia) with my asterisk. I tryed with a normal phone (SMSC number
42100) and all is working fine, incoming and outgoung SMS.

Now I'm trying to configure my asterisk for support SMS, the incoming
messages are working fine (incoming number 042111) dialplan match and
SMS recived. I'm getting troubles in sending messages...

I have done a lot of tests, mainly using a call file. Here the structure:



Channel: SIP/alice-voip/42100
Application: SMS
Data: 0,st
MaxRetries: 10
RetryTime: 30
WaitTime: 30



My test has been done with 2 lines, the first one is a VoIP line, the
second-one is an ISDN (with DAHDI). In all cases when sending the SMS
asterisk seem to don't detect the call pickup of the peer. Using the
SMSC number I get a log similar to these one (SIP):

-- Attempting call on SIP/alice-voip/42100 for application SMS(0,p(300))
(Retry 4)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

(ISDN)

-- Attempting call on DAHDI/g0/42100 for application SMS(0at) (Retry 4)
-- Making new call for cr 32786
-- Requested transfer capability: 0x00 - SPEECH
-- Channel 0/1, span 2 got hangup request, cause 16

As you can see in both cases the SMS App isn't launched. I tryed all
app_sms loptions (p,t,s,a) but seem not an app_sms problem.

If I change the SMSC number and I put my phone number I get that:

-- Attempting call on SIP/alice-voip/XX for application SMS(0,p(300))
(Retry 4)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
sms argc 2 queue <0> opts  addr <> body <>
 initial delay 300ms

The SMS app is launched... The pickup is detected only if I call another
number and not the SMSC one!

The SMSC number is correct, it's working with a normal phone and I can
call it by hand.

Any ideas?

Thank you in advance.

Maurizio

PS: I think can be a stupid error of configuration, but I can't figure
out by myself...
PSS: anyone have experience with SMS over SIP (MIME type vnd.3gpp.sms-tl)?

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Re: [asterisk-users] FastAGiin Windows Server

[Steve Edwards]

On Wed, 14 Apr 2010, Edwin Quijada wrote:

> I wanna know if can run my AGI scripts as fastAGI scripts in Windoes 
> server.

Seems like a move in the wrong direction to me, but no 
-- you can't run an AGI script via fastagi() without changes.

> I need a lot of script done in perl and I wanna move to windows server. 
> I checked Asterisk::fastagi but I see that everything is for Linux.

Fastagi is a protocol. You could implement it in most languages on most 
OSs.

(Everything is for Linux because that's where "server stuff" belongs.)

> Somebody has idea to do this in perl. I dont want to change the 
> language.

Somebody should re-think their ideas :)

Saying you "need a lot of script done" implies you haven't done it yet. 
I'd suggest changing your language to C. You can execute XXX AGIs written 
in C in the time it takes to load Perl and parse your script. Maybe you 
wouldn't even need to use a separate server.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] two FreePBX servers with load balancing

[Hector Muñoz]

Hi there,

Any of you know how can i configure two PBX servers with FreePBX with load
balancing? Do I need a SIP server like OpenSER? Any of you have some
configuration reference?

I have installed AsteriskNow in both FreePBX servers.

Thank you!

Regards
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Re: [asterisk-users] Conference Meetme

[Steve Edwards]

On Wed, 14 Apr 2010, torinti...@hotmail.com wrote:

> How many simultaneous conference meetme setups can be supported in the 
> same time on Asterisk, and what are the corresponding server's specs for 
> this.

How long is a piece of string?

0) A better subject yields better answers

1) A more detailed question yields a more detailed answer.

A reasonably configured Asterisk server can handle XXX callers with Y 
callers in each of Z conferences.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Converting GSM calls to SIP

[Jeff Brower]

Tonty-

> This is more or less the idea.  I was not thinking about the E3 then break
> it down, because I am not sure they provide E3s, they suggest me invest into
> multiple E1 cards to support as many call as I can

Ok but how do you get the data?  35 E1s is a lot of cabling for an external 
connection.

-Jeff

> On Wed, Apr 14, 2010 at 11:56 AM, Jeff Brower wrote:
>
>> Tonty-
>>
>> > This is a solution they proposed, using GSM gateways, but it wont let me
>> > handle 1000 simultaneous calls, the other option was using an E1 but the
>> > cost would be too much to deploy 35 E1s to support that many calls.
>>  There
>> > might be a better way of doing it.
>>
>> Can you explain the "multiple E1" approach?  Are you saying you would
>> connect to your GSM provider using an E3 line
>> and then break that out into multiple E1s that can be used with
>> Asterisk-compatible PCI/PCIe cards?
>>
>> If that's not accurate, please clarify.
>>
>> -Jeff
>>
>> > On Wed, Apr 14, 2010 at 11:08 AM, William Stillwell (Lists) <
>> > william.stillwell-li...@ablebody.net> wrote:
>> >
>> >>
>> http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> >> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
>> >> *Sent:* Wednesday, April 14, 2010 10:52 AM
>> >> *To:* asterisk-users@lists.digium.com
>> >> *Subject:* [asterisk-users] Converting GSM calls to SIP
>> >>
>> >>
>> >>
>> >> I have asked a GSM operator in my country if he can route a number or a
>> >> short code to my asterisk server via SIP (since they dont give DIDs in
>> my
>> >> country) the operator said they do not support SIP, they have no way of
>> >> converting GSM calls to SIP to then send them to me.  I would like to
>> know
>> >> what is needed from the operator side to do this, what kind of material
>> is
>> >> needed, or what can be done from their side to send SIP calls to  my
>> server.
>> >>
>> >> Thank you
>>
>>
>


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[asterisk-users] FastAGiin Windows Server

[Edwin Quijada]

Hi!
I wanna know if can run my AGI scripts as fastAGI scripts in Windoes server. I 
need a lot of script done in perl and I wanna move to windows server. I checked 
Asterisk::fastagi but I see that everything is for Linux.

Somebody has idea to do this in perl. I dont want to change the language.

TIA

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[asterisk-users] Conference Meetme

[torintino1]

How many simultaneous conference meetme setups can be supported in the same 
time on Asterisk, and what are the corresponding server's specs for this.

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Re: [asterisk-users] Converting GSM calls to SIP

[Jeff Brower]

Tonty-

> This is a solution they proposed, using GSM gateways, but it wont let me
> handle 1000 simultaneous calls, the other option was using an E1 but the
> cost would be too much to deploy 35 E1s to support that many calls.  There
> might be a better way of doing it.

Can you explain the "multiple E1" approach?  Are you saying you would connect 
to your GSM provider using an E3 line
and then break that out into multiple E1s that can be used with 
Asterisk-compatible PCI/PCIe cards?

If that's not accurate, please clarify.

-Jeff

> On Wed, Apr 14, 2010 at 11:08 AM, William Stillwell (Lists) <
> william.stillwell-li...@ablebody.net> wrote:
>
>>  http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk
>>
>>
>>
>>
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
>> *Sent:* Wednesday, April 14, 2010 10:52 AM
>> *To:* asterisk-users@lists.digium.com
>> *Subject:* [asterisk-users] Converting GSM calls to SIP
>>
>>
>>
>> I have asked a GSM operator in my country if he can route a number or a
>> short code to my asterisk server via SIP (since they dont give DIDs in my
>> country) the operator said they do not support SIP, they have no way of
>> converting GSM calls to SIP to then send them to me.  I would like to know
>> what is needed from the operator side to do this, what kind of material is
>> needed, or what can be done from their side to send SIP calls to  my server.
>>
>> Thank you


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Re: [asterisk-users] Converting GSM calls to SIP

[Tonty T]

This is a solution they proposed, using GSM gateways, but it wont let me
handle 1000 simultaneous calls, the other option was using an E1 but the
cost would be too much to deploy 35 E1s to support that many calls.  There
might be a better way of doing it.



On Wed, Apr 14, 2010 at 11:08 AM, William Stillwell (Lists) <
william.stillwell-li...@ablebody.net> wrote:

>  http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk
>
>
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
> *Sent:* Wednesday, April 14, 2010 10:52 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Converting GSM calls to SIP
>
>
>
> I have asked a GSM operator in my country if he can route a number or a
> short code to my asterisk server via SIP (since they dont give DIDs in my
> country) the operator said they do not support SIP, they have no way of
> converting GSM calls to SIP to then send them to me.  I would like to know
> what is needed from the operator side to do this, what kind of material is
> needed, or what can be done from their side to send SIP calls to  my server.
>
> Thank you
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] Ring Two Extensions Simultaneously with different caller ID values? [SOLVED]

[Lincoln King-Cliby]

Luki, 

Thanks for the quick response -- that did exactly what I was looking for and 
was even easier than I thought. 

Lincoln

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-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luki
Sent: Tuesday, April 13, 2010 10:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Ring Two Extensions Simultaneously with different 
caller ID values?

Lincoln,

> Is there any way to feed different caller ID information to both sets while
> keeping them ringing simultaneously? My idea is to prefix the called
> extension to the name field (so as not to break redial/callback features on
> the phones)

you can do this with a local channel, like:

Set(__TARGET=${EXTEN})
Dial(SIP/phone1&Local/pho...@common_area)

[common_area]
exten => _phone.,1,Set(CALLERID(name)=${TARGET}: ${CALLERID(name)})
exten => _phone.,n,Dial(SIP/${EXTEN})

Something like that. I hope I got all the () and {} right, I don't do
that much dial-plan coding anymore...

Luki

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Re: [asterisk-users] Converting GSM calls to SIP

[William Stillwell (Lists)]

http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
Sent: Wednesday, April 14, 2010 10:52 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Converting GSM calls to SIP

 

I have asked a GSM operator in my country if he can route a number or a
short code to my asterisk server via SIP (since they dont give DIDs in my
country) the operator said they do not support SIP, they have no way of
converting GSM calls to SIP to then send them to me.  I would like to know
what is needed from the operator side to do this, what kind of material is
needed, or what can be done from their side to send SIP calls to  my server.

Thank you

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Re: [asterisk-users] 1.6.0 verses 1.6.2

[Andrea Cristofanini]

This sound strange, i have running on asterisk 1.4
1090 calls with no problem 24/24 h.
(24 giga ram 4 dual core xeon )

Maybe is the configuration or  configuration tuning missing in somewhere.
Andrea

Il 14/04/2010 10:45, Tzafrir Cohen ha scritto:
> On Tue, Apr 13, 2010 at 04:25:49PM -0600, John Rose wrote:
>> Why do versions 1.6.2 and 1.6.1 use much more CPU resources that 1.6.0?
>> I can get 400+  SIP/G.711
>>
>> calls running on this dual core box with 1.6.0 but the cpu maxes out and
>> core dumps at approx. 180 calls when version 1.6.1/2 is running.
> 
> Could you please be more specific? Have you tested those versions on the
> same system? Can you reproduce those results?
> 
> What exact versions? On what system?
> 


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[asterisk-users] Converting GSM calls to SIP

[Pascal Bruno]

I have asked a GSM operator in my country if he can route a number or a
short code to my asterisk server via SIP (since they dont give DIDs in my
country) the operator said they do not support SIP, they have no way of
converting GSM calls to SIP to then send them to me.  I would like to know
what is needed from the operator side to do this, what kind of material is
needed, or what can be done from their side to send SIP calls to  my server.

Thank you
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Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)

[Ngo-Vi Hoai-Anh]

You can take a look here http://www.gl.com/images/GL_Network_GSM.gif .

As far as I know Asterisk (FreeSwitch, Yate or what so ever for PBX 
system) is normally connected to the GSM core network thru the GMSC 
(Gateway MSC). They talk with each other SS7 ISUP.

Theoretically seen one might have a GMSC talks DSS1 and VoIP (SIP, H323, 
IAX etc...) as well but I don't think it is standard.

Goke M Aruna schrieb:
> hello Huu,
>
> Can you share their explanation with me at least, I can gain from it too.
>
> Thanks
>
> On Wed, Apr 14, 2010 at 10:01 AM, huu giang  > wrote:
>
> Hi Goke,
>
> Some experienced people said me to use ISDN to connect to MSC.
>
> Thanks very much.
>
>
> --- On *Wed, 4/14/10, Goke M Aruna / >/* wrote:
>
>
> From: Goke M Aruna mailto:gok...@gmail.com>>
> Subject: Re: [asterisk-users] protocol used to connect
> Asterisk and GSM core network (MSC)
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>  >
> Date: Wednesday, April 14, 2010, 1:50 AM
>
>
> Hello Huu,
>
> use E1/SS7 signaling or if you MSC speak SIP, then use SIP.
>
> Thanks
>
> On Tue, Apr 13, 2010 at 11:46 AM, huu giang
>  > wrote:
>
> Hi all,
>
> My Asterisk connect to GSM core network (connect directly
> to MSC) through E1 lines. What the kind of protocol is
> used ?. It is ISUP/SS7 protocol ?
>
> Thanks in advance
>
>
>
> --
> 
> _
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>
>
> -Inline Attachment Follows-
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>


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[asterisk-users] 1.6.2.6: can't upgrade from 1.6.1.18

[sean darcy]

I'm running 1.6.1.18 on an older ubuntu machine. I upgraded to 
dahi-linux-2.3.0. That went fine, and it works. But I decided to use the 
opportunity to upgrade to 1.6.2.6. That didn't work.

configure, make menuselect, make, make install all went fine, or at 
least seemed to. But it hangs starting up here:

[Apr 13 20:15:28] VERBOSE[1612] codec_speex.c: -- CODEC SPEEX: 
Setting preprocessor Dereverb
Decay to 0.40
[Apr 13 20:15:28] VERBOSE[1612] codec_speex.c: -- CODEC SPEEX: 
Setting preprocessor Dereverb
Level to 0.30
[Apr 13 20:15:28] VERBOSE[1612] translate.c:   == Registered translator 
'speextolin' from format
speex to slin, cost 1


And whatever does startup doesn't work. I can get into the CLI, but core 
stop now or core restart now, or even help is not valid.

I've rebuilt it twice. Any suggestions?

sean


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Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)

[Goke M Aruna]

hello Huu,

Can you share their explanation with me at least, I can gain from it too.

Thanks

On Wed, Apr 14, 2010 at 10:01 AM, huu giang  wrote:

> Hi Goke,
>
> Some experienced people said me to use ISDN to connect to MSC.
>
> Thanks very much.
>
>
> --- On *Wed, 4/14/10, Goke M Aruna * wrote:
>
>
> From: Goke M Aruna 
> Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM
> core network (MSC)
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> Date: Wednesday, April 14, 2010, 1:50 AM
>
>
> Hello Huu,
>
> use E1/SS7 signaling or if you MSC speak SIP, then use SIP.
>
> Thanks
>
> On Tue, Apr 13, 2010 at 11:46 AM, huu giang 
> http://mc/compose?to=huugiang...@yahoo.com>
> > wrote:
>
>> Hi all,
>>
>> My Asterisk connect to GSM core network (connect directly to MSC) through
>> E1 lines. What the kind of protocol is used ?. It is ISUP/SS7 protocol ?
>>
>> Thanks in advance
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>
>
>
> -Inline Attachment Follows-
>
> --
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>
>
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Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)

[huu giang]

Hi Goke,

Some experienced people said me to use ISDN to connect to MSC. 

Thanks very much.


--- On Wed, 4/14/10, Goke M Aruna  wrote:

From: Goke M Aruna 
Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core 
network (MSC)
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Wednesday, April 14, 2010, 1:50 AM

Hello Huu,

use E1/SS7 signaling or if you MSC speak SIP, then use SIP.

Thanks

On Tue, Apr 13, 2010 at 11:46 AM, huu giang  wrote:


Hi all,

My Asterisk connect to GSM core network (connect directly to MSC) through E1 
lines. What the kind of protocol is used ?. It is ISUP/SS7 protocol ?

Thanks in advance





  
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Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)

[Goke M Aruna]

Hello Huu,

use E1/SS7 signaling or if you MSC speak SIP, then use SIP.

Thanks

On Tue, Apr 13, 2010 at 11:46 AM, huu giang  wrote:

> Hi all,
>
> My Asterisk connect to GSM core network (connect directly to MSC) through
> E1 lines. What the kind of protocol is used ?. It is ISUP/SS7 protocol ?
>
> Thanks in advance
>
>
>
> --
> _
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Re: [asterisk-users] 1.6.0 verses 1.6.2

[Tzafrir Cohen]

On Tue, Apr 13, 2010 at 04:25:49PM -0600, John Rose wrote:
> Why do versions 1.6.2 and 1.6.1 use much more CPU resources that 1.6.0?
> I can get 400+  SIP/G.711
> 
> calls running on this dual core box with 1.6.0 but the cpu maxes out and
> core dumps at approx. 180 calls when version 1.6.1/2 is running.

Could you please be more specific? Have you tested those versions on the
same system? Can you reproduce those results?

What exact versions? On what system?

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Re: [asterisk-users] Merge master.csv files

[Tzafrir Cohen]

On Tue, Apr 13, 2010 at 05:46:03PM +0100, Ricardo Coelho wrote:
> Hi there,
> 
> Does asterisk keeps the master.csv open between writes? Right now I have 2 
> asterisk nodes sharing every configuration file (by using a distributed 
> filesystem) except the master.csv files. If asterisk does not keep master.csv 
> file open between writes, then I can share the master.csv file between both 
> nodes right?If not, then any suggestions to merge both master.csv files?

Sort the lines by some time-stamp?

("merge-sort": sort -m . Though hopefully not sort -R)

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Re: [asterisk-users] DAHDI-Linux and DAHDI-Tools 2.3.0 Released

[Tzafrir Cohen]

On Tue, Apr 13, 2010 at 10:43:41PM +0100, Gordon Henderson wrote:
> On Tue, 13 Apr 2010, Asterisk Development Team wrote:
> 
> > * Static /dev/dahdi files are not generated at install time since udev is 
> > used
> >  on all the supported distributions.  build_tools/make_static_devs is
> >  available for those users who still need the static device files.
> 
> Please do not ever remove the static_devs script - I do not use udev and 
> never will in my embedded systems. There's simply no need for it when your 
> hardware never changes.

Busybox has mdev. I also believe that with latest kernels there's an
option to generate device files on /dev (though never really looked into
it).

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Re: [asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime

[Per Jessen]

Per Jessen wrote:

>> Just start it with safe_asterisk.
>> 
>> http://linux.die.net/man/8/safe_asterisk
>> 
>> Unless my info is out of date, it will kill two birds with one stone.
>> Asterisk will restart itself, and you will get a core dump.
>> 
>> Thanks,
>> Steve Totaro
> 
> Hi Steve
> 
> I've got three such core dumps now - do I just open a bugreport?
> 

See https://issues.asterisk.org/view.php?id=17178


/Per Jessen, Zürich

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