[Asterisk-Users] Re: ZapRAS

[Daniel Nyström]

>Daniel,
>  we have the same problem when our PRI line drops and Zapras has to
>reconnect.  You will also notice that the pppd process does not die
>when Zapras does and the ppp connection cannot re-establish itself.
>What we normally do is restart asterisk and then kill the pppd process
>with the command:  killall -9 pppd.
>How much of your resources does the pppd process take up when Zapras 
>executes?

>Maybe try making pppd a lower priority than asterisk.
>Double check your configuration and make sure that you've done all
>that you needed to in order to properly setup zapras.  go to digiums
>website and look at documentation.

Thanks for your reply. Now I retried ZapRAS and, when not in panic, it 
only required an restart of Asterisk to recover the sound.

But the pppd process was gone and nothing was taking more than 0.2% CPU.
Seems like ZapRAS destroy something within Asterisk then?
Regarding ZapRAS, I don't even get it to work at all. More on that later.
How are your ZapRAS configured? Do you connect to an ISP through it?
I've tried the new app_pppd as well, and that one connects. But the 
connetion dies within a few second (just stop responding/working). See 
my other earlier posts about that.
I've gone through Digiums website more than once to get this to work, 
but there seems to be only a README-file in the ftp area for information.

Which page did you mean?
I've been trying for month to get this working. And alot of questions 
have been posted to the mailing list from me. Even the IRC channels has 
been visited.
But no one seems to have this problem, or are even using the 
ZapRAS/app_pppd.

I request any tips regarding ISDN dialup Internet access.
Thanks!
--
Daniel
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[Asterisk-Users] ZapRAS

[Daniel Nyström]

I'm trying to use ZapRAS to enable ppp connection through my E1.
After the ZapRAS command is executed, all sound is crappy on all lines!
The only solution is to reboot the machine (or halt it, and then power 
it on since Digium's hardware doesn't like reboots).

Anyone know how this can happen?!
I'm using * 1.0.6 on Dell PowerEdge 1850 which are told (too late 
though) not to work very well, but should that really be the problem 
with ZapRAS?!

--
Daniel
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[Asterisk-Users] PPPD problem please help

[Daniel Nyström]

After a big help from Peter Svensson, I got ISDN Data-calls up and 
running.
But now when everything seems connected, pppd has been authorized by 
other peer and even got an IP address, the whole connection seems to 
stop working.
Very unregulary, the PPPD's EchoReq's stop being answered, and of course 
all TCP/IP-traffic as well.

It takes between 0-2 minutes till the connection breaks.
I've setup 'lcp-echo-interval 30' and 'lcp-echo-failure 4' (something 
like that). And it could reply to 1-2 EchoReq's sometimes, sometimes none.

BUT!
The really strange thing is; when my pppd has determined that the serial 
link is closed (the ISDN connection), it will send an Terminate Request!
And that Terminate Request are confirmed by other peer! How strange is 
that?!
I've spoken to my ISP and even got their log from my call. And it all 
seems correct from their side. The EchoReq's doesn't seems to reach them 
though. But the Terminal Request does afterwards.


I think this is a Nobel Prize issue. :)
--
Daniel

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[Asterisk-Users] ISDN RAS and data calls

[Daniel Nyström]

It seems like when I use PPPD-command, or ZapRAS, Asterisk doesn't make 
it a "data" call, but a regular voice-call.
My ISP change their behaivour depending on the incoming call-type (data 
or voice).
If it's voice, they try to open up a V.90 connection. Else (data call) 
it will reply with PPP directly.

The both methods uses the same modem pool number.

How can I tell Asterisk to inititate a data-call instead of voice-call?
I'm using .call-files to connect to the ISP.
--
Daniel
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[Asterisk-Users] Serious ZapRAS problem!

[Daniel Nyström]

Hi!

I've been trying to get ZapRAS or PPPD to work. Neither does!
All i get is LCP: timeout sending Config-Requests

But after trying, all voicelines get crazy! It sounds like robots when 
somebody calls!
And since the zaptel drivers can't unload (the server hangs totaly if I 
try!), I have to reboot the whole server!

The robot-voice is only on our side, it sounds fine at the other end.

My server is an Dell PowerEdge 1850 with an Digium TE110P card to PRI.
I'm using "stable" Asterisk 1.0.6 and I'm located in Sweden with TDC 
Song as telco.

--
Daniel
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[Asterisk-Users] DIAL with FastAGI and Answer Supervision

[Daniel Nyström]

Hi!
I'm using FastAGI (agi://) to make some calls. To do the dialing i use "EXEC 
DIAL Zap/g1/...".
But how can I make "answer supervision" with FastAGI? DIAL command won't return 
until call is finished.
Thanks in advance!
--
Daniel
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[Asterisk-Users] Unchanged sound through Asterisk

[Daniel Nyström]

Hi!

To me, it seems like Asterisk are involved in alternating the sound/voice 
running through it.
One thing is that it mutes DTMF digits.
I also got an Adit 600 channel bank connected via MGCP, which _might_ have 
something to do with it,
but I can't find any settings in it, regarding DTMF mutes.

How can I make sure Asterisk is _not_ "changing"/"transforming" the sound (as 
muting DTMF etc.) in any way?

Thanks!
--
Daniel
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[Asterisk-Users] DTMF detection with Adit 600

[Daniel Nyström]

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It seems like Asterisk are having problems detecting DTMF digits when
using an Adit 600 channel bank via MGCP.
I've tried to turn on RFC 2833 on both Adit and Asterisk, but no
digits at all are working then.
Anyone experienced simular with Adit or other channel banks?
I'm also unable to use V.90 modem through my setup (Adit600 via MGCP
- -> Asterisk -> E1).
Fax worked once though.. Does the Echo Cancelling make the problems
with V.90?
- --
Daniel 
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[Asterisk-Users] ChanIsAvail for MGCP

[Daniel Nyström]

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ChanIsAvail does not work with MGCP channels, as said in the wiki.
But other applications works simular, like Queue and Dial.
What's really the problem with ChanIsAvail?
Is it possible to use Queue and Dial to make a working ChanIsAvail?
I will take a better look in the source when back at work on monday,
but some tips and facts will help for sure.
- --
Daniel
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Re: [Asterisk-Users] Strange area codes when dialing outgoing calls on EuroISDN E1 FIXED!!

[Daniel Nyström]

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administrator tootai wrote:
| Daniel Nyström a écrit :
|
|> My server is located in Sweden. And as many European countries,
|> we use a 0 to indicate area codes, and 00 to indicate
|> international calls. And, when not having any leading 0, the call
|> is a local call. But when dialing out through Asterisk, I can't
|> use leading zeros! I havn't tried international calls, but
|> non-local calls must begin directly with the area code without
|> the leading zero to work. This is really problematic, cause there
|> are no chance to call local calls, or service numbers, or even
|> 112 (Swedish 911). According to the Telco, the number sent should
|> include leading zeros, so the problem seems to be within
|> Asterisk. Any tips is appriciated!
|>
|>
| Hi Daniel,
|
| asterisk is not your problem, it's your dialplan which is not
| adapted. We are using 0 for long distance or area calls without
| problem. Someting like dial(ZAP/1/0{EXEN}) should call your number
| prefixed with 0 If you remove the leading 0 number will be dialed
| as is. Or perhaps you're using a gateway PBX who automaticaly
| add/remove prefixes.
|
Your were right! The PRIDIALPLAN was not set (didn't though I needed
it). Now it all works GREAT!
- --
Daniel
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[Asterisk-Users] Strange area codes when dialing outgoing calls on EuroISDN E1

[Daniel Nyström]

My server is located in Sweden. And as many European countries, we use a 0 to 
indicate area codes, and 00 to indicate international calls.
And, when not having any leading 0, the call is a local call.
But when dialing out through Asterisk, I can't use leading zeros! I havn't 
tried international calls, but non-local calls must begin directly with the 
area code without the leading zero to work.
This is really problematic, cause there are no chance to call local calls, or 
service numbers, or even 112 (Swedish 911).
According to the Telco, the number sent should include leading zeros, so the 
problem seems to be within Asterisk.
Any tips is appriciated!
--
Daniel
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[Asterisk-Users] Echo cancelling with Adit 600

[Daniel Nyström]

Do anyone have experience with echo cancelling on Adit 600?
My Adit 600 consist of 5*8 FXS cards and 1 CMG Router using MGCP to Asterisk.
I've turned on Echo Cancelling with 64ms as longest delay (that's maximum).
But there still are great echo with delay when dialing through the telco 
(through an E1 and EuroISDN).

Any advice will be appriciated!

Thanks!
--
Daniel
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[Asterisk-Users] Asterisk + Adit 600 questions

[Daniel Nyström]

Is it possible to make Asterisk to execute a task when a called party answeres?
Does the MGCP protocol include support for notificate when a call is answered?
I have one Adit 600 w/ 40 FXS lines. When a call is initiated from such line to 
the 
PSTN through our E1 EuroISDN, I would like the Adit to, somehow, indicate on 
the FXS-line that the other user has answered (lifted his/her handset).
By changing battery polarity or maybe an signal? 
Automatic equipment does use almost every FXS-line of ours, and they need to
know when the call is answered.

Any ideas please?
--
BR
Daniel
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[Asterisk-Users] Indicating when other party has answered

[Daniel Nyström]

Here in Sweden when I make a call through the regular POTS, I get an polarity 
reversal when the callee has lift his phone and answered.
Now I've got an Adit 600 with 40 FXS channels and want to "emulate" an regular 
POTS. But the Adit doesn't seem to support polarity reversal.
Is there other standards how to indicate to the caller that the callee has 
answered the call? How does it work in other countries?

Thanks!
--
Daniel Nyström
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[Asterisk-Users] More 2-way radio controlling in *

[Daniel Nyström]

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It seems like the Radio discussions is closing in on something I was
interested in.
How about controlling 30 2-way radios via E1 and 30-channel "Mux"
(channel bank?) with E&M signalling?
I think the Mux uses CAS and each channel has Audio out, PTT, Audio
IN, Busy. 6-wire connection i guess?
That should be a really nice setup with Asterisk!
Anyone tried something like this?
- --
Daniel
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[Asterisk-Users] Amphenol cables?

[Daniel Nyström]

A little off-topic maybe, but it's still for the Adit used with Asterisk. ;)

I wonder where I can buy 50 pin Amphenol cables, with connector on one side, 
and open cables on the other for mounting in our own patch panels.
In Europe, or Sweden preferably.
It's said to be very common on telcos, but I can't find it for sale anywhere!

Thanks!
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[Asterisk-Users] Adit 600 MGCP configuration

[Daniel Nyström]

I've finally got my Adit 600 and are configuring it right now.
But I have to say, there aren't much documentation for it.
I've setup MGCP and Asterisk seems to find it.
But all channels (40 FXS channels) are "Down"!
But the MGCP itself is "Up" according to the statistics.
I can't find any documents how to set each channel to "Up" in the CLI.

Any suggestions?

Thanks!
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[Asterisk-Users] E&M and other Radio-based signalling

[Daniel Nyström]

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Have anyone tried using this?
I've looked at app_rpt, and that's a nice project, but have anyone
tried using Asterisk for radio services using a Mux e.g.? I was
thinking of using an E&M Mux (or channel bank i think) with
TX/RX/BUSY/PTT functionality.
Or even tried to decode any signalling commonly used with radio
communication?
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Re: [Asterisk-Users] E1 and/or Euro-ISDN specifications?

[Daniel Nyström]

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What's exactly Euro-ISDN? Is it G.931? I don't really get this.
Is there a Q/G/E document for Euro-ISDN?
I've downloaded two out of three fron ITU, so I would like to know for
sure! :)
Thanks!
Peter Svensson wrote:
| On Tue, 15 Feb 2005, Daniel Nyström wrote:
|
|> Where can I get E1 and/or Euro-ISDN specifications/data sheets?
|> Are there specs for other E./G./Q./etc. protocols as well?
|
|
| The specifications are built one on top of another. Each just lists
| the changes and clarifications relative to the underlaying
| specification. E.g. for the Swedish incumbent operator Telia you
| have:
|
| * Telia ISDN Notes for Suppliers * ETSI specifications (e.g. ETSI
| ETS 300 403 01 etc) * ITU specifications (e.g. Q.931, Q.921 and
| lots and lots of others)
|
| The Telia implementation specification is free, as are the ETSI
| specification (at least for some uses). The ITU specifications are
| a bit expensive, but you can download three for free.
|
| Most of the time the ITU specifications are at the "bottom level",
| but not always.
|
| Peter
|
|
| ___ Asterisk-Users
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[Asterisk-Users] E1 and/or Euro-ISDN specifications?

[Daniel Nyström]

Where can I get E1 and/or Euro-ISDN specifications/data sheets?
Are there specs for other E./G./Q./etc. protocols as well?

Thanks!
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[Asterisk-Users] DTMF CLIP in Sweden and others

[Daniel Nyström]

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Are this currently working with CVS-HEAD?
I've got an X100P-clone, and I've patched the zaptel drivers.
But the Asterisk patches seems to be there.
But I can't make it receive Caller-ID!
Btw, by doing a cvs checkout asterisk, the HEAD-version will be
downloaded right?
Thanks!
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[Asterisk-Users] Individual contexts pending on Caller-ID?

[Daniel Nyström]

Hi! Is it possible to handle incoming calls with different contexts pending on 
the callerid ?
E.g. like you are able to define different contexts on each Zap-channel.

Thanks!
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Re: [Asterisk-Users] Adit 600

[Daniel Nyström]

I've just ordered an Adit 600 w/ 5xFXS cards and one CMG cards. 
As of my discussion with CarrierAccess, it seems to work great.
I've also begun an configuration (mgcp.conf) until it arrives, and also there 
it seems to have great capabilities.
There are alot of data sheets and information of all Adit 600 pieces on the 
CarrierAccess website.
Take a look at those and see if it fits your needs.

- Original Message - 
From: "Isaac McDonald" <[EMAIL PROTECTED]>
To: 
Sent: Thursday, January 27, 2005 4:25 PM
Subject: [Asterisk-Users] Adit 600


> Has anyone had any success using the Adit 600 with the CMG card talking 
> MGCP to asterisk? I want to have a central asterisk server with 10 Adit 
> 600's at various locations providing 24 FXS ports
> 
> Thanks,
> 
> Isaac
> 
>
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Re: [Asterisk-Users] Some more hardware and E1 questions

[Daniel Nyström]

As long as the bootloader exists on both disks, and boot order are including 
both disks, there aren't any problems even booting with a failured disk.
But since SATA is (often) Hot Plug, you could change the failed disk while 
running.

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

Sent: Tuesday, January 25, 2005 7:50 PM
Subject: Re: [Asterisk-Users] Some more hardware and E1 questions


IMHO hardware RAID trumps software RAID. In order to use the latter 
your system must still be operational to some extent.

-mark

On Jan 24, 2005, at 11:10 PM, Gary wrote:

> better solution rather than have a machine with raid is to investigate
> ISCSI :-)
>
> On Mon, 24 Jan 2005 09:40:10 +0100, Daniel Nystr"m wrote:
>
>> I will be using Debian, and as long as the Linux Kernel supports the 
>> SATA controller, the rest shouldn't be any problems.
>> If it's SATA RAID, I probably will use ordinary Linux software RAID, 
>> since it's more powerful than the simple one in the controller.
>>
>> - Original Message -
>> From: "Leo Ann Boon" <[EMAIL PROTECTED]>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>> 
>> Sent: Saturday, January 22, 2005 6:13 AM
>> Subject: Re: [Asterisk-Users] Some more hardware and E1 questions
>>
>>
>>>
>>>
>>> Daniel Nystr"m wrote:
>>>
 Hi again folks! ;)

 As before, I will transform one E1 30 Channel PRI into 30 FXS 
 channels using Adit 600.

 Now I'm into choosing server platform. And the two opponents are:
 * Dell PowerEdge 750 w/ SCSI RAID (or even SATA RAID1)
 * FujistuSiemens PRIMERGY RX100 S2 (SATA RAID1)


>>> If you're planning to use SATA RAID on PE750, make sure your Linux
>>> distro supports. Your best bet - use Redhat Enterprise Linux or one 
>>> of
>>> it derivatives. I'm using Centos 3, it autodetects the RAID whilst
>>> Mandrake 10 failed.
>>>
 As I've seen people having problem with HP server, I havn't looked 
 at it at all.

 What experience do you have with the alternatives above? Which 
 would you recommend?

 And another question at the same time; what's really E1?
 How is E1 devices connected? Seems like regular Cat5 cables, but it 
 problably ian't?
 If anyone's using Adit 600, did they send all cables required for 
 connecting to the FXS channels? Seems like a very unique "plug" on 
 the side of Adit.

 Thanks!

 BR
 Daniel Nystr"m
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>>>
>>>
>> ___
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>
> .
>
>
> ___
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--
Mark Eissler, [EMAIL PROTECTED]
Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com



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[Asterisk-Users] BUSY-tone on incoming calls?

[Daniel Nyström]

Is it possible to make the telco send an busy signal when an incoming call are 
supposed to dial a group which has all lines busy?
Since I will get many public phonenumbers into my E1 (from telco), it will be 
sliced up into a few groups. There might be channels availible in the E1, but 
not on the other side of Asterisk (the office side).

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[Asterisk-Users] Turn off DTMF recognition pending on CallerID

[Daniel Nyström]

Is it possible to turn off DTMF recognition (and all transfer services etc.) 
pending on CallerID (or FXS channel)?
Some of the FXS channels I will setup soon, is going to work exactly like POTS.
It will be used by people not knowing their within Asterisk.
They will be pretty confused when "Transfer" is playbacked in the handset. :)

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[Asterisk-Users] XEON or not

[Daniel Nyström]

Are there much performance differences when using XEON or not?
In my case, I will go with muLaw both in and out of Asterisk. Are there really 
any processing at all if it's using same codec all the way?

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Re: [Asterisk-Users] Some more hardware and E1 questions

[Daniel Nyström]

I will be using Debian, and as long as the Linux Kernel supports the SATA 
controller, the rest shouldn't be any problems.
If it's SATA RAID, I probably will use ordinary Linux software RAID, since it's 
more powerful than the simple one in the controller.

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

Sent: Saturday, January 22, 2005 6:13 AM
Subject: Re: [Asterisk-Users] Some more hardware and E1 questions


> 
> 
> Daniel Nyström wrote:
> 
> >Hi again folks! ;)
> >
> >As before, I will transform one E1 30 Channel PRI into 30 FXS channels using 
> >Adit 600.
> >
> >Now I'm into choosing server platform. And the two opponents are:
> > * Dell PowerEdge 750 w/ SCSI RAID (or even SATA RAID1)
> > * FujistuSiemens PRIMERGY RX100 S2 (SATA RAID1)
> >  
> >
> If you're planning to use SATA RAID on PE750, make sure your Linux 
> distro supports. Your best bet - use Redhat Enterprise Linux or one of 
> it derivatives. I'm using Centos 3, it autodetects the RAID whilst 
> Mandrake 10 failed.
> 
> >As I've seen people having problem with HP server, I havn't looked at it at 
> >all.
> >
> >What experience do you have with the alternatives above? Which would you 
> >recommend?
> >
> >And another question at the same time; what's really E1?
> >How is E1 devices connected? Seems like regular Cat5 cables, but it 
> >problably ian't?
> >If anyone's using Adit 600, did they send all cables required for connecting 
> >to the FXS channels? Seems like a very unique "plug" on the side of Adit.
> >
> >Thanks!
> >
> >BR
> >Daniel Nyström
> >___
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> >To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >  
> >
> 
>
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Re: [Asterisk-Users] Adit 600 as VoIP router (MGCP) and Asterisk

[Daniel Nyström]

As CarrierAccess states, there can be potential mismatch regarding the TDM 
signaling required to terminate the voice channels onto the FXS cards.
I'm not sure I understand this fully.
He also says "Although E1 still remains as another option, given a compatible 
signaling pattern". Since I'm a real newbie in this area, I don't know what the 
signaling pattern would be?

Of course I could ask them all this questions, but it can take a long time 
until answer arrive. One thing is the time difference. And now, the seller has 
gone on a trip for the weekend, and won't be back until Monday. Which will 
become tuesday in Sweden. And my project is really urgent!
Thanks for helping me out!

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

Sent: Friday, January 21, 2005 2:42 PM
Subject: Re: [Asterisk-Users] Adit 600 as VoIP router (MGCP) and Asterisk


> On Fri, 21 Jan 2005, Daniel Nyström wrote:
> 
> > Do you think it's hearable? All communication will be on a dedicated
> > Fast Ethernet link (just a cross-over cable). And it will still use aLaw
> > codec (same as Euro ISDN afaik).
> 
> > Since E1 = 2048Kbps and FastEth = 100Mbps, I concidered it fast enough
> > to not make any latency. It seems like MGCP also can use UDP. In that
> > case, there won't be any retransmission or anything.
> 
> What you will notcie is the packetization delay of 20ms (or 30 depening on
> the codecs) which yeilds 40-60ms round trip time possibly combined with
> jitter buffers. The transmission time, once packetized is usually not very
> significant, it is the packetization and possible user level queues that 
> mess up the latency.
> 
> The delay on almost any voip technology can transform a nice sidetone to a 
> nasty echo. This is one of many reasons telecom people strive to minimize 
> latency.
> 
> > Do you have any experiences using MGCP of any kind?
> 
> Unfortunatly not. 
> 
> Peter
> 
> 
> 
>
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Re: [Asterisk-Users] Adit 600 as VoIP router (MGCP) and Asterisk

[Daniel Nyström]

Do you think it's hearable? All communication will be on a dedicated Fast 
Ethernet link (just a cross-over cable). And it will still use aLaw codec (same 
as Euro ISDN afaik).
Since E1 = 2048Kbps and FastEth = 100Mbps, I concidered it fast enough to not 
make any latency. It seems like MGCP also can use UDP. In that case, there 
won't be any retransmission or anything.
Do you have any experiences using MGCP of any kind?

Thanks for the tip!


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

Sent: Friday, January 21, 2005 11:50 AM
Subject: Re: [Asterisk-Users] Adit 600 as VoIP router (MGCP) and Asterisk


> On Fri, 21 Jan 2005, Daniel Nyström wrote:
> 
> > Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router 
> > using MGCP IP protocol, instead of controlling it through an E1.
> > Have anyone tried this configuration? How does MGCP works? I've tried to 
> > search for it on Google, but I only find the protocol specification for it.
> > Is Asterisk fully capable of this? I can't find any documentatin covering 
> > the use of MGCP in Asterisk.
> > My mission is still to transform one E1 (30ch) from the telco, into 30 FXS 
> > channels, all via Asterisk. :)
> > 
> > Do anyone have a successful story about using MGCP with Asterisk?
> > 
> > Do I still have the full power of Asterisk even when using MGCP? I would 
> > like it to be, like if I were using FXS PCI-cards localy in the server. :)
> 
> One thing you will loose by going to mgcp is the very low latency you have 
> on a pure tdm solution. Low latency makes echo supression a lot easier.
> 
> Peter
> 
> 
>
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Re: [Asterisk-Users] Adit 600 as VoIP router (MGCP) and Asterisk

[Daniel Nyström]

The thing about PCI-cards were just a way to illustrate how I'm thinking, not a 
sollution. I'm sorry for the confusion i brought.
It's already clear that an Adit 600 will work as an Channel Bank.
The problem was just that, either we use on E1 to the Adit, but there might be 
some issues with that, since we're also using one PRI E1 to the telco.
Then we´re going to use an Ethernet connection to the Adit, and configure it as 
an MG with MGCP.

That's why I was wondering if there are any people in the list which have used 
Asterisk with Adit (or other channel banks) controlled by MGCP.
There are also almost no documentation regarding Asterisk and MGCP. If I get 
this to work, I might contribute to that part on the Wiki.

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

Sent: Friday, January 21, 2005 10:36 AM
Subject: Re: [Asterisk-Users] Adit 600 as VoIP router (MGCP) and Asterisk


> On Fri, 2005-01-21 at 09:06 +0100, Daniel Nyström wrote:
> > Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP
> > router using MGCP IP protocol, instead of controlling it through an
> > E1.
> > Have anyone tried this configuration? How does MGCP works? I've tried
> > to search for it on Google, but I only find the protocol specification
> > for it.
> > Is Asterisk fully capable of this? I can't find any documentatin
> > covering the use of MGCP in Asterisk.
> > My mission is still to transform one E1 (30ch) from the telco, into 30
> > FXS channels, all via Asterisk. :)
> > 
> > Do anyone have a successful story about using MGCP with Asterisk?
> > 
> > Do I still have the full power of Asterisk even when using MGCP? I
> > would like it to be, like if I were using FXS PCI-cards localy in the
> > server. :)
> 
> 30 channels of FXS on PCI cards currently would require 8 cards. Not
> financially a good idea not to mention that if you stick to the 2 cards
> per machine that you would be up to 4 machines just for those 30 ports. 
> 
> I don't know what the MGCP card costs for the ADIT. Compare it to the
> cost of 1 or 2 T1 ports via Digium cards.  
> -- 
> Steven Critchfield <[EMAIL PROTECTED]>
> 
> 
>
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[Asterisk-Users] Adit 600 as VoIP router (MGCP) and Asterisk

[Daniel Nyström]

Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router 
using MGCP IP protocol, instead of controlling it through an E1.
Have anyone tried this configuration? How does MGCP works? I've tried to search 
for it on Google, but I only find the protocol specification for it.
Is Asterisk fully capable of this? I can't find any documentatin covering the 
use of MGCP in Asterisk.
My mission is still to transform one E1 (30ch) from the telco, into 30 FXS 
channels, all via Asterisk. :)

Do anyone have a successful story about using MGCP with Asterisk?

Do I still have the full power of Asterisk even when using MGCP? I would like 
it to be, like if I were using FXS PCI-cards localy in the server. :)
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[Asterisk-Users] Some more hardware and E1 questions

[Daniel Nyström]

Hi again folks! ;)

As before, I will transform one E1 30 Channel PRI into 30 FXS channels using 
Adit 600.

Now I'm into choosing server platform. And the two opponents are:
 * Dell PowerEdge 750 w/ SCSI RAID (or even SATA RAID1)
 * FujistuSiemens PRIMERGY RX100 S2 (SATA RAID1)

As I've seen people having problem with HP server, I havn't looked at it at all.

What experience do you have with the alternatives above? Which would you 
recommend?

And another question at the same time; what's really E1?
How is E1 devices connected? Seems like regular Cat5 cables, but it problably 
ian't?
If anyone's using Adit 600, did they send all cables required for connecting to 
the FXS channels? Seems like a very unique "plug" on the side of Adit.

Thanks!

BR
Daniel Nyström
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Re: [Asterisk-Users] Resellers in Europe

[Daniel Nyström]

Their server seems to be down though...

- Original Message - 
From: "Wilson Pickett" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial 
Discussion" 
Sent: Wednesday, January 19, 2005 3:26 PM
Subject: Re: [Asterisk-Users] Resellers in Europe


> > I've had ordered some items (VoIP starter kit, TDM400P, a couple of ATAs
> > and an IAXy) from Digitnetworks to Reunion Island (which *technically*
> > is in the EU despite its remote location) and with UPS shipping it came
> > fast (about 1 week - but expensive though). No major troubles except
> 
> We've ordered from Eikonex in France with good results.
> 
>
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[Asterisk-Users] Resellers in Europe

[Daniel Nyström]

Do anyone knows abount European resellers of these products:
 * Digium Wildcard TE410P
 * CarrierAccess Adit 600

Preferably in Sweden, but Europe is also better.

Have anyone within EU ordered products from these companies directly from the 
US?
In that case, how is the service and delivery?

Thanks, folks!
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Re: [Asterisk-Users] Prefered server hardware

[Daniel Nyström]

Are you using dual PSU's or something? What redundance do you have on your 
system?
Do you use any channel banks? Is it only telco connections on the TE405P?
I will use one E1 from the telco, and one from Asterisk to the Adit. Is it 
enough with one TE405P, or should I use two different cards?

Thanks!

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

Sent: Tuesday, January 18, 2005 3:10 PM
Subject: Re: [Asterisk-Users] Prefered server hardware


> On Tue, 18 Jan 2005, Daniel Nyström wrote:
> 
> > What server hardware would you recommend for an Asterisk system which
> > are really critical? The additional hardware will probably be two digium
> > TE110P cards, and an Adit 600 platform.
> > 
> > If it's possible to run on -48VDC, It would be great!
> > 
> > Are there any experiences with any HP or FujitsuSiemens systems? Or
> > other "complete server" systems?
> 
> We use Fujitsu servers. No observed problems with a single TE405P. A lot 
> of people have problems with at least one model of HP server. Perhaps you 
> should try searching the mail archives for the brands you are considering.
> 
> Most brand name servers can be configured with a -48V power supply. 
> 
> Peter
> 
> 
> 
>
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[Asterisk-Users] Prefered server hardware

[Daniel Nyström]

What server hardware would you recommend for an Asterisk system which are 
really critical?
The additional hardware will probably be two digium TE110P cards, and an Adit 
600 platform.

If it's possible to run on -48VDC, It would be great!

Are there any experiences with any HP or FujitsuSiemens systems? Or other 
"complete server" systems?

Thanks!

BR
Daniel Nyström
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Re: [Asterisk-Users] CAS voice signalling?

[Daniel Nyström]

Oh, thanks!
How do I know which codec is used on Adit 600?
Does the server need to reencode it at all, or is the codec the same on Euro 
ISDN?
If it has to reencode everything, it really seems to be CPU critical when using 
30 FSX lines into 30 Euro ISDN lines.

Btw, when using Adit for connecting 30 handsets. Is it FXS or FXO modules I 
need? As I've seen, there is alot of misunderstanding in that particulary case.

BR
Daniel Nyström

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

Sent: Monday, January 17, 2005 2:57 PM
Subject: Re: [Asterisk-Users] CAS voice signalling?


> On Mon, 2005-01-17 at 14:33 +0100, Daniel Nyström wrote:
> > According to CarrierAccess, the Adit 600 uses CAS for voice signalling. 
> > What is this?
> > This should not be a problem for Asterisk?
> > Does the Asterisk server need to reencode CAS into aLaw when going to Euro 
> > ISDN?
> 
> CAS is Channel Associated Signaling. It isn't dependent on alaw or ulaw
> it has to do with where the signaling bits are. 
> 
> Adit 600's are well known to work with asterisk. 
> -- 
> Steven Critchfield <[EMAIL PROTECTED]>
> 
> 
>
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[Asterisk-Users] CAS voice signalling?

[Daniel Nyström]

According to CarrierAccess, the Adit 600 uses CAS for voice signalling. What is 
this?
This should not be a problem for Asterisk?
Does the Asterisk server need to reencode CAS into aLaw when going to Euro ISDN?

BR
Daniel Nyström
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[Asterisk-Users] Euro ISDN and Caller ID (Sweden)

[Daniel Nyström]

Do anyone have experiences with Euro ISDN in Sweden?
Does CallerID work properly? Both in and out.
Do anyone know of a reseller for Digium cards and/or CarrierAccess Adit 600 in 
Sweden or Europe (EU)?

Thanks!

BR
Daniel Nyström
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Re: [Asterisk-Users] Hardware issues

[Daniel Nyström]

Is it still possible to acheive all features in Asterisk, like having Digium 
cards for all channels?
I'm looking at an Adit 600 with four 8-ch FXS service cards. Is that to prefer?

BR
Daniel
- Original Message - 
From: "Jim Van Meggelen" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 

Sent: Friday, January 14, 2005 12:46 PM
Subject: RE: [Asterisk-Users] Hardware issues


> [EMAIL PROTECTED] wrote:
> > Since i need to run at least 30 FXS-channels, what hardware
> > should I use? Both Motherboard etc and FXS-hardware? There
> > seems to be only 4-channel cards from digium. Then I would
> > need at least 8 cards. 8 FXS-cards + 1 E1-card. What
> > motherboard handles this? And to they all need unique IRQ's?
> 
> You'll probably want to look into channel banks.
> 
> http://www.voip-info.org/wiki-Asterisk+Channel+Bank
> 
> 
> -- 
> No virus found in this outgoing message.
> Checked by AVG Anti-Virus.
> Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 12/01/2005
>  
> 
> 
>
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[Asterisk-Users] Hardware issues

[Daniel Nyström]

Since i need to run at least 30 FXS-channels, what hardware should I use?
Both Motherboard etc and FXS-hardware?
There seems to be only 4-channel cards from digium. Then I would need at least 
8 cards.
8 FXS-cards + 1 E1-card. What motherboard handles this?
And to they all need unique IRQ's?

BR
Daniel Nyström
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[Asterisk-Users] Grouping lines pending on Called ID

[Daniel Nyström]

I will be using one E1 to the telco, and there will be 4 static phone numbers, 
and one number serie with 1000 numbers ranging from e.g. 555-1000 to 555-1999.
There will be 30 FSX lines on the other side of Asterisk.
Is it possible to "group" those 30 FSX lines pending on which number was dialed?
Let's say line 1-4 are for the static numbers, and 5-30 for the other 1000 
numbers.

Are there any documentation with this issue?

Best Regards
Daniel Nyström

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[Asterisk-Users] Asterisk with Euro ISDN, etc

[Daniel Nyström]

Hi folks!
 
Our company are going to buy an E1 line 
with Euro ISDN and 30 lines (channels).
 
This is how it will be configured:
3 Lines, of the total of 30, is going to be for the 
company phones, and share one phonenumber (eg. 555-12340).
1 Line will be dedicated to a specific unique 
phonenumber (Fax) (eg. 555-54321).
The rest of the lines/channels (26) will be used by 
(by, not for) our customers, and will be redirected via our system to mobile 
devices (for more info: http://www.westel.se, 
and choose English in up-left corner).
The large "group" will use a range of 1000 
phonenumbers, which in turn will specify which mobile device it will redirect to 
(eg. 555-4 to 555-40999).
 
All lines/channels need to be connected to analog 
phones! And with the "large group", it has to deliver all Caller-ID and which of 
the thousend number was called. Preferably with DTMF. And on out-going calls, it 
also has to receive destination phonenumber, and preferably even it's own number 
(one of the thousend numbers).
 
Since we are going to provide unique numbers to each mobile device (just 
like a cellular), it requires alot of unusual features.
And about that out-going calls receiving the callers number, it would be 
nice to present the number from which mobile device the call is made. Again like 
regular cellulars.
 
We already have analoge interfaces for our current exchanger, and each 
of them are connected to regular PSTN lines (each with individual accounts and 
numbers). That's why we need analog interface from Asterisk to our 
exchanger.
 
Is this possible with Asterisk?
 
Hope this wasn't too confusing. Just let me know if there are anything 
unclear, and I will try to explain it in a better way.
 
Happy new year!
 
Best Regards
Daniel 
Nyström
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