Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Tzafrir Cohen]

On Tue, May 26, 2009 at 07:52:53PM +0300, Tzafrir Cohen wrote:
> On Tue, May 26, 2009 at 12:44:26PM -0400, Jon Pounder wrote:
> > Wilton Helm wrote:
> > 
> > one thing I missed mentioning about fxs devices - the linksys/sipura 
> > ones actually allow you to set line characteristics on the slic inside 
> > it. you can vary from the 600ohm default, and tweak gains a bit. Some 
> > mix of a capacitive line or different resistance may help. never tried 
> > myself but there are a ton of things you can play with.
> 
> Any of those are actually important?

For the sake of completeness: try:

  /sbin/modinfo wctdm

or:

  /sbin/modinfo wctdm24xxp

You'll see quite a few parameters, many of which are essentially SLIC
(or DAA, for the FXO port) tweaks.

I suppose that if it were useful, there were already some demand to make
it tweakable (safely. Merely writing an arbitrary value to some register
at some point may or may not be wise). Hence my question.

-- 
   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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Re: [asterisk-users] SIP over VPN

[Aurimas Skirgaila]

run tcpdump, while trying to connect to asterisk to see what ports are
requiered.
default SIP port is UDP 5060, but as mentioned before all your traffic
should go over VPN so port openening shouldn't be a problem



On Wed, May 27, 2009 at 8:40 AM, Marco Sambo  wrote:

> Ok,
> but if I want to open only SIP port on firewall, which ones? I have the
> following situation:
>
> computer A (softphone)  firewall  computer B (asterisk)
>
> and I dont' want to open any ports, only SIP and voice.
>
>
>
>
> 2009/5/26 David Gibbons 
>
>>   Assuming you mean the firewall in front of the client, you don’t need
>> to open any ports as long as the VPN client is tunneling all traffic to and
>> from the Asterisk server.
>>
>>
>>
>> I  set NAT=yes in the config file for the extensions behind a VPN.
>>
>>
>>
>> -Dave
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marco Sambo
>> *Sent:* Tuesday, May 26, 2009 11:21 AM
>> *To:* asterisk-users@lists.digium.com
>> *Subject:* [asterisk-users] SIP over VPN
>>
>>
>>
>> Hi all,
>> I have a question. I have a VPN and I want to use a SIP softphone on my
>> notebook using with asterisk. But I have some problem with firewall and
>> port.
>> Someone knows which ports I should open on my firewall??? I can't connect
>> ...
>>
>> Thanks all.
>>
>> Marco
>>
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>
>
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-- 
Mvh,
Aurimas Skirgaila
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Re: [asterisk-users] Bug or feature in 1.6.1 (Was: How to register with TCP transport) ?

[Olivier]

I filed https://issues.asterisk.org/view.php?id=15202
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[asterisk-users] Asterisk memory problems

[Ikin Wirawan]

Hi guys, we have the following problem

After putting our Asterisk/PHP application on production, there is one big
problem, ie memory leak after a period of usage (about 20MB after 2 minutes)

There are also two more info that may help:
1) Asterisk consumes 495mb of memory on this production server, while only
42kB on local/development machine
2) agi_ccmain (and other agi scripts) consumes 146mb of memory on
production, while only 32kB on local

Also on local, we do not experience the memory leak (or maybe since the
memory usage itself is small it's hardly noticeable)

The only difference that I am aware of is that on production, DAHDI is
installed

My questions are:
- How come Asterisk, and my PHP/AGI scripts consume so much memory on
production compared to local machines?
- Any idea why there is a memory leak? IAll of the PHP scripts are done
executing, so memory should all be released. Is it a bug on Asterisk?

We are using Asterisk 1.6.0.6 & CentOS 5.2 on production

Any help is kindly appreciated

Sincerely,



-- 
Ikin Wirawan
Chief Executive Officer
PT Walden Global Services
Integrity, Learning, Sharing, Excellence
http://www.wgs.co.id
http://www.kiranatama.com - Web 2.0 development
http://www.qorser.com - VoIP solutions
http://www.hellomedia.co.id - Digital design & promotions
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[asterisk-users] PHP AGI Problems

[Atlanticnynex]

(Accidentally posted this to asterisk-dev, should be here)

fgets is only returning one character... either when run as an AGI or
run as a test on PHP on CLI...
Example, enter , then fgets returns '3'.

Also, GET DATA seems to be returning early and the loop keeps
prompting 'invalid'...
Any suggestions on how to improve my AGI class so it actually works?

Thanks.

[code]


#!/usr/bin/php
selectdb("switchboard");

while ($error_count < 5 && !$msg_result) {
   $result = $AGI->send_cmd("GET DATA /bswitch/menu/enter-msg-id 3000 4");
   $msg_id = $result[result];

   if ($msg_id < 1000){
   echo "\ndebug: msgid < 1000, invalid ($msg_id)!\n";
   //var_dump($msg_id);
   $error_count += 1;
   $AGI->send_cmd("EXEC PLAYBACK /bswitch/menu/invalid-msg-id");
   }
   else{
   echo "\ndebug: ($msg_id) okay valid msgid, lets check sql\n";
   $msg_result=check_msg($msg_id);
   if(!$msg_result){
   $error_count +=1;
   }
   }
}

$row = mysql_fetch_array($messageresult);
$AGI->send_cmd("EXEC CONTROL STREAM FILE $row[path]");
$AGI->send_cmd("EXEC PLAYBACK beep");


function check_msg($ID){
   $msg_result=$database->querydb("**sql query filtered**");
   return $msg_result;
}

?>






_get_agivars();
   }

}
?>



[/code]

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Re: [asterisk-users] SIP over VPN

[Marco Sambo]

Ok,
but if I want to open only SIP port on firewall, which ones? I have the
following situation:

computer A (softphone)  firewall  computer B (asterisk)

and I dont' want to open any ports, only SIP and voice.




2009/5/26 David Gibbons 

>  Assuming you mean the firewall in front of the client, you don’t need to
> open any ports as long as the VPN client is tunneling all traffic to and
> from the Asterisk server.
>
>
>
> I  set NAT=yes in the config file for the extensions behind a VPN.
>
>
>
> -Dave
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marco Sambo
> *Sent:* Tuesday, May 26, 2009 11:21 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] SIP over VPN
>
>
>
> Hi all,
> I have a question. I have a VPN and I want to use a SIP softphone on my
> notebook using with asterisk. But I have some problem with firewall and
> port.
> Someone knows which ports I should open on my firewall??? I can't connect
> ...
>
> Thanks all.
>
> Marco
>
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Wilton Helm]

[snip]

>You may be able to boost the battery voltage with a simple dc adapter in 
>series to get the line build out capability you need. Just make sure its 
>floating with respect to ground and wire it in. Don't be afraid of 
>hurting the phone, you won't.

But it is possible to hurt the ATA if it causes more current than expected
to flow.  It is even possible to force reverse voltage into the ATA that
way.  A line fault would make it even more likely.

Wilton



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Re: [asterisk-users] asterisk-addon 1.6.1 problem

[Rilawich Ango]

I follow it to set [readhost.asterisk] and [writehost.asterisk] and
extconfig.conf sippeers =>
mysql,readhost.asterisk/writehost.asterisk,sipfriends. However the
error message still existed.  Can you give me an example of
res_mysql.conf and extconfig.conf?

On Tue, May 26, 2009 at 10:33 PM, Tilghman Lesher
 wrote:
> On Tuesday 26 May 2009 02:52:18 Rilawich Ango wrote:
>> Hi all,
>>   I download asterisk-addon 1.6.1 but the VoIP phone failed to
>> register to the system with the message below.
>>
>> [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317
>> realtime_mysql: MySQL RealTime: Invalid database specified: asterisk
>> [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317
>> realtime_mysql: MySQL RealTime: Invalid database specified: asterisk
>> sip
>>
>> I use the same configuration file (res_mysql.conf & extconfig.conf) in
>> 1.6.0 but failed.  Any big change in 1.6.1?
>
> Please read UPGRADE.txt in the asterisk-addons directory.
>
> --
> Tilghman
>
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[John Novack]

Jon Pounder wrote:
> John Novack wrote:
>> If this is an emergency phone situation then I would question the 
>> wisdom of even considering using Asterisk.
>> Conventional telephony solutions exist that will easily cover the 
>> loop length and provide the reliability that should be required  by 
>> risk management in such a situation.
>>   
> why are you going on the assumption asterisk is somehow inherently 
> less reliable than a "conventional" solution ?
>
Because it is.
A simple solution is best.
fewer items to fail.
The OP has given no reason to develop a complex solution for what is 
presented as a fairly simple problem to provide communications in an 
emergency situation over a short loop in conventional telephony
3Km isn't a long loop in the telephone world, though some postings would 
incorrectly say otherwise.
> I am not trying to start any sort of war here, but is that based on 
> any sort of facts ? hardware wise its basically all the same 
> electronics whether they were meant as a general purpose computer or a 
> telephony specific computer - they all fail eventually and the MTBF is 
> usually related to the relative price in the specific market.
Why do you assume a "telephony specific computer" is even needed?

KISS!

John Novack

-- 
Dog is my co-pilot


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[asterisk-users] Call in progress tones

[Mikel Lindsaar]

Hello all,

I've played with background and play sounds apps and googled around
and asked the list before to no avail.

Does anyone know of a way to have tones played during the call
progress stage of the call?

We (especially on some international circuits) get up to 5 seconds of
silence before the phone starts ringing or is busy.

I don't want to force "R" on the Dial app as then you can get "ring
ring, ring ring, rin, beep beep beep" when the phone is busy.

What I want is something to play during the call setup or "making
progress" stage of the call.  just a series of beeps about 800ms apart
until the phone call is actually set up... so then you would get
something like "bip, bip, bip, bip, bip, ring ring, ring ring..." for
ringing or "bip, bip, bip, bip, bip, beep, beep, beep..." for busy.

Any ideas?

Mikel

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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Lyle Giese]

Even with 'conventional' PBXs, there is such a thing as power fail
devices where the extension is cut to a telco pots line for dial tone if
the PBX goes down.

Jon Pounder wrote:
> John Novack wrote:
>   
>> If this is an emergency phone situation then I would question the wisdom 
>> of even considering using Asterisk.
>> Conventional telephony solutions exist that will easily cover the loop 
>> length and provide the reliability that should be required  by risk 
>> management in such a situation.
>>   
>> 
> why are you going on the assumption asterisk is somehow inherently less 
> reliable than a "conventional" solution ?
>
> I am not trying to start any sort of war here, but is that based on any 
> sort of facts ? hardware wise its basically all the same electronics 
> whether they were meant as a general purpose computer or a telephony 
> specific computer - they all fail eventually and the MTBF is usually 
> related to the relative price in the specific market. I have not really 
> had any software reliability problems in years of running asterisk 
> (although some do and I am sure there are firmware revs for pbx's that 
> have issues too)
>
> so why make that general statement ?
>
> as far as risk management - any one system can fail, end of story. Risk 
> management would entail a backup system if failure of the primary is not 
> acceptable. In a tunnel application physical damage to the wiring is 
> probably a lot more likely than a hardware failure, be it from accident, 
> fire, collapse etc., meaning when you need the phone most, it is least 
> likely to work. Those factors would affect any hardwired telephony 
> solution equally.
>
>   
>> John Novack
>>
>> asterisk-us...@rogg.is wrote:
>>   
>> 
>>> Appreciate all your input folks. Much of it very helpful in the greater
>>> context of the initial question.
>>>
>>> Thank you for the suggestion of using various wireless devices, but I'm
>>> stuck with fixed wiring since this is a security/emergency phone(s)
>>> installation underground in large tunnels.
>>>
>>> Also, switching to VOIP is not really the answer here because then I'm
>>> forced to solve a lot of power, repeaters/switches problems that arise. So
>>> I'm actually worse of than using the analog connections I think.
>>>
>>> I do have some control over the wiring/cable chosen for this project but
>>> still forced to find a solution where I can feed the analog "phone line" the
>>> total 3km line distance.
>>>
>>> I would love to find a way to do this in the Asterisk context with some sort
>>> of FXS feed, either from Digium (or compatible) hardware or any of the
>>> available ATA boxes. The Sapura box suggestion may be something and I'll
>>> look closer into that as well as continuing to look for other ways to do
>>> this.
>>>
>>> tnx!
>>>
>>> Baldvin
>>>
>>>   
>>> 
>>>   
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Hans Witvliet
 Sent: 26. maí 2009 19:42
 To: novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [asterisk-users] Maximum cable length for analog phone
 from FXS port

 I would suggest making a wifi connection with directional hi-gain
 antenna's.
 Ans a small box at the other end. Have a look at:
 http://www.fit-pc.net/fitpc-2-p-2.html or http://www.fit-
 pc.info/downloads/handleidingen/fit_pc_2_eng.pdf

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>>>   
>>> 
>>>   
>>   
>> 
>
>
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Hans Witvliet]

On Tue, 2009-05-26 at 21:29 +, asterisk-us...@rogg.is wrote:
> Appreciate all your input folks. Much of it very helpful in the greater
> context of the initial question.
> 
> Thank you for the suggestion of using various wireless devices, but I'm
> stuck with fixed wiring since this is a security/emergency phone(s)
> installation underground in large tunnels.
> 
> Also, switching to VOIP is not really the answer here because then I'm
> forced to solve a lot of power, repeaters/switches problems that arise. So
> I'm actually worse of than using the analog connections I think.
> 
> I do have some control over the wiring/cable chosen for this project but
> still forced to find a solution where I can feed the analog "phone line" the
> total 3km line distance.
> 
> I would love to find a way to do this in the Asterisk context with some sort
> of FXS feed, either from Digium (or compatible) hardware or any of the
> available ATA boxes. The Sapura box suggestion may be something and I'll
> look closer into that as well as continuing to look for other ways to do
> this.
> 

Ah! that explains.
As ordinary ethernet and wifi are impossible, and lines are too long for
ordinary analoge phones, options are limited.

Another idea is to use baseband modems, just like the pair used by
telco's for DSL. You can bridge 5KM or more...

If it's for a small number of phones, T1/E1 modems might be
impractible/expensive. And i presume you don't have a DSLAM lying around
there ;-)

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Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router

[Tarek Sawah]

through a test .. i was able to send calls from Asterisk 1.4 to a PSTN number 
through a cisco router with a channel bank.. Audio worked well..  i setup a 
dial plan in asterisk to Dial(${ext...@ciscoip)  and authorise the cisco 
router's ip on the asterisk server and treat the calls comming from it like any 
other SIP calls inside the server.. 


--
AHD Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP

Syria: +963 944 618286

USA: +1 347 562 2308






> Date: Sat, 16 May 2009 14:46:27 +0300
> From: timotsm...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
> 
> Hi,
> 
> In our office, we're slowly migrating from a cisco call manager set up
> to asterisk. Problem is management doesn't want to buy any other
> hardware  as they had already invested a lot in cisco. The main cause
> of this is asterisk's added features like unique FAX number for
> everyone in the company (which will be the same as phone DID), Voice
> mail, Auto Answer etc yet we need thousands of dollars to add those to
> our cisco call manager 4.1 set up.
> 
> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
> and also a dialpeer to forward on the router to forward calls to my
> asterisk. It works properly but the problem is there is NO AUDIO! I
> have tried to change codec but no sucess!
> 
> Has anyone had the above set up working successfully? Attached are some confs.
> 
> Thanks a lot for your assistance.
> 
> Kind Regards,
> Wilson

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Re: [asterisk-users] Bandwidth management and ADSL Router

[Alex Balashov]

bilal ghayyad wrote:
> Dear Eric;
> 
> Sangoma has ADSL router? And does that router support bandwidth division 
> capability?

Internal ADSL cards;  not external router appliances.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] Bandwidth management and ADSL Router

[bilal ghayyad]

Dear Eric;

Sangoma has ADSL router? And does that router support bandwidth division 
capability?

Dear jas;

About what u mentioned: it is related to linux, do u know a dsl router that 
does bandwidth divion?

Any help?
Regards
Bilal


 

> I've had good luck using a sangoma S518 ADSL card in a
> linux box.  the
> logging capabilities are supurb (cought my provider not
> providing what they
> said they were and great for troubleshooting as it logs
> line speed and
> dropouts to the second).  support is also top
> notch.  once installed it
> looks to the system like any other interface.  Since
> it looks to the system
> like any other interface you have the full power of
> routing, bridging,
> firewalling, iptables, neumerous queing schemes, etc. 
> everything linux has
> to offer.  It has served me well and is extremely
> flexable.
> 
> Eric Fort
> FortConsulting
> 
> On Tue, May 26, 2009 at 4:32 AM, bilal ghayyad 
> wrote:
> 
> >
> > Hi All;
> >
> > I discover that most of the voice cutting complain are
> coming from the
> > Internet bandwidth when we are connecting two remote
> offices togethor via
> > Asterisk or any other IP PBX.
> >
> > Anyone has an idea on a ADSL router that work as ADSL
> + Bandwidth division?
> > So we can resolve the problem of providing a
> guaranteed bandwidth for the
> > voice packets instead of suffering the voice cutting?
> >
> > Regards
> > Bilal



  

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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Jon Pounder]

John Novack wrote:
> If this is an emergency phone situation then I would question the wisdom 
> of even considering using Asterisk.
> Conventional telephony solutions exist that will easily cover the loop 
> length and provide the reliability that should be required  by risk 
> management in such a situation.
>   
why are you going on the assumption asterisk is somehow inherently less 
reliable than a "conventional" solution ?

I am not trying to start any sort of war here, but is that based on any 
sort of facts ? hardware wise its basically all the same electronics 
whether they were meant as a general purpose computer or a telephony 
specific computer - they all fail eventually and the MTBF is usually 
related to the relative price in the specific market. I have not really 
had any software reliability problems in years of running asterisk 
(although some do and I am sure there are firmware revs for pbx's that 
have issues too)

so why make that general statement ?

as far as risk management - any one system can fail, end of story. Risk 
management would entail a backup system if failure of the primary is not 
acceptable. In a tunnel application physical damage to the wiring is 
probably a lot more likely than a hardware failure, be it from accident, 
fire, collapse etc., meaning when you need the phone most, it is least 
likely to work. Those factors would affect any hardwired telephony 
solution equally.

> John Novack
>
> asterisk-us...@rogg.is wrote:
>   
>> Appreciate all your input folks. Much of it very helpful in the greater
>> context of the initial question.
>>
>> Thank you for the suggestion of using various wireless devices, but I'm
>> stuck with fixed wiring since this is a security/emergency phone(s)
>> installation underground in large tunnels.
>>
>> Also, switching to VOIP is not really the answer here because then I'm
>> forced to solve a lot of power, repeaters/switches problems that arise. So
>> I'm actually worse of than using the analog connections I think.
>>
>> I do have some control over the wiring/cable chosen for this project but
>> still forced to find a solution where I can feed the analog "phone line" the
>> total 3km line distance.
>>
>> I would love to find a way to do this in the Asterisk context with some sort
>> of FXS feed, either from Digium (or compatible) hardware or any of the
>> available ATA boxes. The Sapura box suggestion may be something and I'll
>> look closer into that as well as continuing to look for other ways to do
>> this.
>>
>> tnx!
>>
>> Baldvin
>>
>>   
>> 
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>>> boun...@lists.digium.com] On Behalf Of Hans Witvliet
>>> Sent: 26. maí 2009 19:42
>>> To: novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial
>>> Discussion
>>> Subject: Re: [asterisk-users] Maximum cable length for analog phone
>>> from FXS port
>>>
>>> I would suggest making a wifi connection with directional hi-gain
>>> antenna's.
>>> Ans a small box at the other end. Have a look at:
>>> http://www.fit-pc.net/fitpc-2-p-2.html or http://www.fit-
>>> pc.info/downloads/handleidingen/fit_pc_2_eng.pdf
>>>
>>> ___
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>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>> 
>>>   
>>
>>
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>>   
>> 
>
>   


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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Jon Pounder]

asterisk-us...@rogg.is wrote:

couple last words on this - if that is the application, then ringing the 
remote terminal may not even be necessary, you really only care about 
the hookswitch and audio which is a different thing entirely from ringing.

You may be able to boost the battery voltage with a simple dc adapter in 
series to get the line build out capability you need. Just make sure its 
floating with respect to ground and wire it in. Don't be afraid of 
hurting the phone, you won't.

Check out dialplans for the ATA's for warmline and hotline for emergency 
phones you will probably want this. This is one I use for example

warm dial - wait 3 sec and if not 3 digit number dial 100
( P3 <:100>|xxx )

ie: if they know who they want locally on the pbx they have 3 seconds to 
dial it, otherwise the ata just dials extension 100 which is an 
entrypoint to one of the IVR trees on my system.

> Appreciate all your input folks. Much of it very helpful in the greater
> context of the initial question.
>
> Thank you for the suggestion of using various wireless devices, but I'm
> stuck with fixed wiring since this is a security/emergency phone(s)
> installation underground in large tunnels.
>
> Also, switching to VOIP is not really the answer here because then I'm
> forced to solve a lot of power, repeaters/switches problems that arise. So
> I'm actually worse of than using the analog connections I think.
>
> I do have some control over the wiring/cable chosen for this project but
> still forced to find a solution where I can feed the analog "phone line" the
> total 3km line distance.
>
> I would love to find a way to do this in the Asterisk context with some sort
> of FXS feed, either from Digium (or compatible) hardware or any of the
> available ATA boxes. The Sapura box suggestion may be something and I'll
> look closer into that as well as continuing to look for other ways to do
> this.
>
> tnx!
>
> Baldvin
>
>   
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>> boun...@lists.digium.com] On Behalf Of Hans Witvliet
>> Sent: 26. maí 2009 19:42
>> To: novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial
>> Discussion
>> Subject: Re: [asterisk-users] Maximum cable length for analog phone
>> from FXS port
>>
>> I would suggest making a wifi connection with directional hi-gain
>> antenna's.
>> Ans a small box at the other end. Have a look at:
>> http://www.fit-pc.net/fitpc-2-p-2.html or http://www.fit-
>> pc.info/downloads/handleidingen/fit_pc_2_eng.pdf
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>
>
>
>
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>   


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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[John Novack]

If this is an emergency phone situation then I would question the wisdom 
of even considering using Asterisk.
Conventional telephony solutions exist that will easily cover the loop 
length and provide the reliability that should be required  by risk 
management in such a situation.

John Novack

asterisk-us...@rogg.is wrote:
> Appreciate all your input folks. Much of it very helpful in the greater
> context of the initial question.
>
> Thank you for the suggestion of using various wireless devices, but I'm
> stuck with fixed wiring since this is a security/emergency phone(s)
> installation underground in large tunnels.
>
> Also, switching to VOIP is not really the answer here because then I'm
> forced to solve a lot of power, repeaters/switches problems that arise. So
> I'm actually worse of than using the analog connections I think.
>
> I do have some control over the wiring/cable chosen for this project but
> still forced to find a solution where I can feed the analog "phone line" the
> total 3km line distance.
>
> I would love to find a way to do this in the Asterisk context with some sort
> of FXS feed, either from Digium (or compatible) hardware or any of the
> available ATA boxes. The Sapura box suggestion may be something and I'll
> look closer into that as well as continuing to look for other ways to do
> this.
>
> tnx!
>
> Baldvin
>
>   
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>> boun...@lists.digium.com] On Behalf Of Hans Witvliet
>> Sent: 26. maí 2009 19:42
>> To: novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial
>> Discussion
>> Subject: Re: [asterisk-users] Maximum cable length for analog phone
>> from FXS port
>>
>> I would suggest making a wifi connection with directional hi-gain
>> antenna's.
>> Ans a small box at the other end. Have a look at:
>> http://www.fit-pc.net/fitpc-2-p-2.html or http://www.fit-
>> pc.info/downloads/handleidingen/fit_pc_2_eng.pdf
>>
>> ___
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>> 
>
>
>
>
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>   

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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[asterisk-users]

Appreciate all your input folks. Much of it very helpful in the greater
context of the initial question.

Thank you for the suggestion of using various wireless devices, but I'm
stuck with fixed wiring since this is a security/emergency phone(s)
installation underground in large tunnels.

Also, switching to VOIP is not really the answer here because then I'm
forced to solve a lot of power, repeaters/switches problems that arise. So
I'm actually worse of than using the analog connections I think.

I do have some control over the wiring/cable chosen for this project but
still forced to find a solution where I can feed the analog "phone line" the
total 3km line distance.

I would love to find a way to do this in the Asterisk context with some sort
of FXS feed, either from Digium (or compatible) hardware or any of the
available ATA boxes. The Sapura box suggestion may be something and I'll
look closer into that as well as continuing to look for other ways to do
this.

tnx!

Baldvin

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Hans Witvliet
> Sent: 26. maí 2009 19:42
> To: novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] Maximum cable length for analog phone
> from FXS port
> 
> I would suggest making a wifi connection with directional hi-gain
> antenna's.
> Ans a small box at the other end. Have a look at:
> http://www.fit-pc.net/fitpc-2-p-2.html or http://www.fit-
> pc.info/downloads/handleidingen/fit_pc_2_eng.pdf
> 
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Re: [asterisk-users] Indications.conf and tone generation volume

[John Novack]

Well, really they are a LOT too loud, as they are meant to simulate 
tones used in a bygone era where there was inband signaling that was 
muted to the caller.

John Novack


Lee Spenadel wrote:
>
> Sorry if this is a repost – I never saw a copy of this go out last week.
>
> Can anyone tell me if there is a way to vary the output levels (dB) of 
> the tones generated in indications.conf? I generate a few custom tones 
> and sometimes people tell me they are a little too loud.
>
> Thanks
>
> Lee
>
> 
>
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Re: [asterisk-users] Fax Machines across carrier SIP trunk? General recommendation?

[Alex Balashov]

Modem and/or analog passthrough over SIP trunk, not on a LAN?  I 
wouldn't bother - it doesn't even work very well on an uncontended LAN 
due to excessive jitter, let alone over the Internet or semi-private 
Layer 2 cloud product.

T.38 or bust.  The other's fax mileage is measured in gallons per mile, 
not miles per gallon.

Jason Aarons (US) wrote:

> Customer has a Verizon Business SIP trunk, I’m still used to PRI T1 
> myself for local service.  The fax machines are having some issues (I 
> can use analog phone to call out fine)  and I’m checking on modem 
> passthrough with Verizon, but wonder if any else is using Verizon 
> Business for SIP trunk and what your faxing milage was? Did they support 
> G711 and modem-passthough, etc? Also checking QoS, etc.
> 
>  
> 
> 
> 
> * Disclaimer: This e-mail communication and any attachments may contain 
> confidential and privileged information and is for use by the designated 
> addressee(s) named above only. If you are not the intended addressee, 
> you are hereby notified that you have received this communication in 
> error and that any use or reproduction of this email or its contents is 
> strictly prohibited and may be unlawful. If you have received this 
> communication in error, please notify us immediately by replying to this 
> message and deleting it from your computer. Thank you. *
> 
> 
> 
> 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] h extension and channel variables

[Benny Amorsen]

Thomas Kenyon  writes:

> In that case if there is an intervening call that is shorter, then the 
> $calledID will be wrong.

That isn't how Asterisk variables work. They aren't global to all calls,
they are local to the call you happen to be in. So no, an intervening
call won't cause problems.


/Benny


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[asterisk-users] Fax Machines across carrier SIP trunk? General recommendation?

[Jason Aarons (US)]

Customer has a Verizon Business SIP trunk, I'm still used to PRI T1
myself for local service.  The fax machines are having some issues (I
can use analog phone to call out fine)  and I'm checking on modem
passthrough with Verizon, but wonder if any else is using Verizon
Business for SIP trunk and what your faxing milage was? Did they support
G711 and modem-passthough, etc? Also checking QoS, etc.

 




-
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
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[asterisk-users] multiple bind ports with TCP and UDP

[Olivier]

Hi,

In this thread
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/223399/focus=223401,
one conclusion was that an easy way to set 2 different trunks with
different binding ports was to use TCP and UDP transport.

serverA - serverB
|   |



Has anyone successfully set this up ?
I'm using 1.6.1 and I've trouble to do this.

Regards
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Re: [asterisk-users] Add Monitor application to call suppresses audio

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

David Backeberg wrote:

>> 5) exten => s,n,Dial(${OUTGOING_PRI}/${ANSWERINGSVC},15,r);
>> 6) exten => s,n,Goto(s-${DIALSTATUS},1);

> What is the 6 for?
> What is the goto supposed to do?

Hi David.

The '6' is in case I get a "CHANUNAVAIL" or other error back from the
Dial command.   If the call is connected then I never get to '6'.

I have determined that the only calls I seem to be having trouble
monitoring are the ones sent to my answering service.  If I terminate
the call to my cell phone, my home POTS line,  a POTS line here in the
office or even to the inbound PRI at the office, things work fine.  I
can even record calls to the answering service's published number.  It's
just when I go to the number assigned to us that there is trouble and
I'm currently chasing down the owner of that service to see exactly what
 I'm dropping into there.

Thanks!

Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)

iD8DBQFKHEbRCFu3bIiwtTARAvlIAJ0Se61+0k6W3ixwZOm8/Sz+ixZqXQCgqLnz
2kLwyY8bHLrs/aaGd9nrho8=
=Tbri
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Hans Witvliet]

On Tue, 2009-05-26 at 10:26 -0400, John Novack wrote:
> That is a pretty long run.
> The type of analog phone can be an issue. How LITTLE loop current will
> it operate on? Most need more than 20 Ma to signal properly, and the
> voltage output of the ATA needs to be known
> Type of signaling? DTMF? pulse?
> Interconnection cable wire size and capacitance will affect high
> frequency response, loop current, inductive pickup and pulse shaping
> to name just a few. The ATA requirements need to be known. A total
> loop resistance of 500 ohms should work, but go out to 1200 and most
> will fail
> Do you really have control over this or will you be renting a pair
> from the local telco?
> Protection should be applied on both ends for safety of the user(s)
> and devices.
> There MUST be a better way???
> 

I would suggest making a wifi connection with directional hi-gain
antenna's.
Ans a small box at the other end. Have a look at:
http://www.fit-pc.net/fitpc-2-p-2.html or
http://www.fit-pc.info/downloads/handleidingen/fit_pc_2_eng.pdf

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[asterisk-users] Indications.conf and tone generation volume

[Lee Spenadel]

Sorry if this is a repost - I never saw a copy of this go out last week.

 

 

Can anyone tell me if there is a way to vary the output levels (dB) of the
tones generated in indications.conf?  I generate a few custom tones and
sometimes people tell me they are a little too loud.

 

Thanks

Lee

 

 

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[asterisk-users] Bug or feature in 1.6.1 (Was: How to register with TCP transport) ?

[Olivier]

Hi,

Digging on this case :

2009/5/26 Olivier 

> Hi,
>
> In my sip.conf, I've got :
> [general](+)
> ;   
> register=>tcp://trunk4ipbx:passw...@192.168.100.129
> 
> register=>trunk4ipbx:passw...@192.168.100.129
>
> When I'm using the TCP line instead of the other, I've got :
> [May 26 17:58:42] NOTICE[2859]: chan_sip.c:20169 sip_parse_host: '/' is not
> a valid port number on line 25 of sip.conf. using default.
> [May 26 17:58:42] WARNING[2859]: chan_sip.c:6560 sip_register: Format for
> registration is
> [transport://]user[:secret[:authuse...@domain[:port][/extension][~expiry] at
> line 25
>
>
> Is this 
> "register=>tcp://trunk4ipbx:passw...@192.168.100.129"
> statement correct ?
>
> Regards
>


I read in chan_sip.c that block inside sip_register :

   /* split [/contact][~expiry] */
expire = strchr(buf, '~');
if (expire)
*expire++ = '\0';
callback = strrchr(buf, '/');// My comment: contact is
search at the end of input register line
if (callback)
*callback++ = '\0';
if (ast_strlen_zero(callback))
callback = "s";

sip_parse_host(buf, lineno, &username, &portnum, &transport);

Given an input line such as "register=>tcp://
trunk4ipbx:passw...@192.168.100.129 ",
register line is truncated as the last occurence of '/' is the "tcp://"
string.
When commenting out this "callback = strrchr(buf, '/');" , input line
"register=>tcp://trunk4ipbx:passw...@192.168.100.129"
seems to be processed appropriately.

My question is "is this legal to input register lines without any /contact
field ?
If positive, then there is a bug is 1.6.1.
If negative, would you agree to have a more appropriate logging than
"sip_parse_host: '/' is not a valid port number ..." ?

Regards
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Re: [asterisk-users] STUN setting in Asterisk 1.6.X

[Moises Silva]

> [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
> stun failed
>
>        No matter which STUN server I point to I get those messages.  Am I
> missing some other setting?

Hey Carlos,

That just means the stun request failed, there are several reasons for
that, I won't even try to guess. So, first try this on the Asterisk
CLI:

"stun set debug on"

That should give you (and us) more information to troubleshoot why the
stun request failed (also enable debug and verbosity as usual).

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
L3R 9T3 Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com

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Re: [asterisk-users] Silly (??) question about chan_dahdi

[Martin]

You define context= for the channels in dahdi.conf
and then in extensions.conf you define those numbers in that
particular context name

eg:

dahdi.conf

context=incoming
channel => 1-15,17-31

extensions.conf

[incoming]
exten => _X.,1,Answer
exten => _X.,2,Echo

and it will "react" to all numbers that come on that circuit and do
Echo app on incoming calls

Martin

On Tue, May 26, 2009 at 1:30 PM, Stefan-Michael Guenther
 wrote:
> Hi,
>
> these are my first steps with DAHDI and I finally managed to get
> asterisk to load chan_dahdi (after I found out, that I need libpri).
>
> But how do I tell chan_dahdi on which isdn numbers it should react? I
> haven't found a parameter like "incomingmsn" for chan_capi in the
> documentation.
>
> Thanks for your help,
>
> Stefan
>
>
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Re: [asterisk-users] Add Monitor application to call suppresses audio

[David Backeberg]

On Wed, May 13, 2009 at 11:53 AM, Barry L. Kline  wrote:
> If I insert a Monitor() prior to dialing the outbound call, I get no
> audio in the recording and the caller hears no audio.   Occasionally it
> works (perhaps 1 out of 5 times) but most of the time the caller can't
> hear the callee, and vice versa.
>
> The fully working code looks like this:
> 1) exten => s,n(place),Verbose(4,Dialing answering service);
> 2) exten => s,n,Playback(vrec_prompts/this-call-may-be-recorded);
> 3) exten => s,n,Set(GROUP()=ANSSVC);
> 4) exten =>
> s,n,Set(CALLFILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(num)});
> 5) exten => s,n,Dial(${OUTGOING_PRI}/${ANSWERINGSVC},15,r);
> 6) exten => s,n,Goto(s-${DIALSTATUS},1);

What is the 6 for?
What is the goto supposed to do?

This could certainly explain why the first call works and not the
subsequent calls.
Why don't you want to just hangup the call after 5 completes?

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[asterisk-users] Silly (??) question about chan_dahdi

[Stefan-Michael Guenther]

Hi,

these are my first steps with DAHDI and I finally managed to get 
asterisk to load chan_dahdi (after I found out, that I need libpri).

But how do I tell chan_dahdi on which isdn numbers it should react? I 
haven't found a parameter like "incomingmsn" for chan_capi in the 
documentation.

Thanks for your help,

Stefan


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[asterisk-users] Strange message in CLI

[Joseph L. Casale]

While I was in the console looking for something else, this appeared when I 
called in on my cell.

[May 26 12:17:26] NOTICE[3364]: chan_sip.c:17229 handle_request_invite: Sending 
fake auth rejection for user "xxx  xxx xx" 
;tag=as04e93fb9

What does this mean? Searching the net simply brought me to the source files.

Thanks!
jlc

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Re: [asterisk-users] Bandwidth management and ADSL router

[Eric Fort]

I've had good luck using a sangoma S518 ADSL card in a linux box.  the
logging capabilities are supurb (cought my provider not providing what they
said they were and great for troubleshooting as it logs line speed and
dropouts to the second).  support is also top notch.  once installed it
looks to the system like any other interface.  Since it looks to the system
like any other interface you have the full power of routing, bridging,
firewalling, iptables, neumerous queing schemes, etc.  everything linux has
to offer.  It has served me well and is extremely flexable.

Eric Fort
FortConsulting

On Tue, May 26, 2009 at 4:32 AM, bilal ghayyad  wrote:

>
> Hi All;
>
> I discover that most of the voice cutting complain are coming from the
> Internet bandwidth when we are connecting two remote offices togethor via
> Asterisk or any other IP PBX.
>
> Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division?
> So we can resolve the problem of providing a guaranteed bandwidth for the
> voice packets instead of suffering the voice cutting?
>
> Regards
> Bilal
>
>
>
>
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Re: [asterisk-users] Asterisk and Data Modem

[Robert Boardman]

Jon Morgan wrote:
> Hi All,
>
> We have an Asterisk 1.4.21.2 box which uses a 2 port Digium card to bridge
> calls, as follows:
>
> ISDN Provider <---> Span 1(pri_cpe) <---> Span 2(pri_net) <> Phone
> System 
>
> The company that looks after our internal phone system can no longer dial in
> using their data modem in order to maintain the internal phone system.  Is
> there any way we can configure our asterisk to allow them to dial in using
> their modem?
>
> Regards,
>
> Jon.
>
>
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>   
Hi jon

What system is it?

you need to set the transfer capability

eg
exten => _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?8:)
exten => _X.,2,Noop
exten => _X.,3,ringing
exten => _X.,4,set(CDR(accountcode)=${EXTEN})
exten => _X.,5,Noop
exten => _X.,6,dial(ZAP/g2/${EXTEN},,r)
exten => _X.,7,hangup
exten => _X.,8,Set(CHANNEL(transfercapability)=DIGITAL)
exten => _X.,9,dial(ZAP/g2/${EXTEN})
exten => _X.,n,hangup


Regards
Robb

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[asterisk-users] STUN setting in Asterisk 1.6.X

[Carlos Chavez]

I have been trying out several stun servers with Asterisk 1.6.0.9 and
1.6.1.0 and I keep getting the following message:

[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed

No matter which STUN server I point to I get those messages.  Am I
missing some other setting?  

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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[asterisk-users] No Voice - only "noisy audio"

[Diogo Saad]

Hi Folks,

I'm trying to use my mobile as a trunk via bluetooth - calls done in a
softphone go thru GSM network and calls destinated to my mobile are answered
at the softphone.

I have asterisk configured to do so but I'm facing an issue - Audio is
audible but it’s not intelligible. I feel like the audio is breaking.
Below is the asterisk log. I also get lots of “hci_scodata_packet: hci0 SCO
packet for unknown connection handle X” and "btusb_isoc_complete: hci0
corrupted SCO packet" entries in kernel logs.

Can anybody please help?
Tks
++
13:37:17 chan_sip.c: Allocating new SIP dialog for
42eb60ff0430607e7eb97cc86...@192.168.0.204 - OPTIONS (No RTP)
13:37:17 acl.c: Found IP address for this socket
13:37:17 chan_sip.c: Initializing initreq for method OPTIONS - callid
5bae8a561541036e45990a137366c...@192.168.0.204
13:37:17 chan_sip.c: Trying to put 'OPTIONS si' onto UDP socket destined for
192.168.0.84:27928
13:37:17 chan_sip.c: Stopping retransmission on '
5bae8a561541036e45990a137366c...@192.168.0.204' of Request 102: Match Found
13:37:17 chan_sip.c: Destroying SIP dialog
5bae8a561541036e45990a137366c...@192.168.0.204
13:37:40 acl.c: Found IP address for this socket
13:37:40 netsock.c:   == Using SIP RTP CoS mark 5
13:37:40 chan_sip.c: Setting NAT on RTP to Off
13:37:40 chan_sip.c: Allocating new SIP dialog for
N2ExYWUwNGJkODAzZGMyYjBmYzQwMTY1YzgwMGQ5MWM. - INVITE (With RTP)
13:37:40 chan_sip.c:  Received INVITE (5) - Command in SIP INVITE
13:37:40 chan_sip.c: Setting NAT on RTP to Off
13:37:40 chan_sip.c: Trying to put 'SIP/2.0 40' onto UDP socket destined for
192.168.0.84:27928
13:37:40 chan_sip.c:  Received ACK (6) - Command in SIP ACK
13:37:40 chan_sip.c: Stopping retransmission on
'N2ExYWUwNGJkODAzZGMyYjBmYzQwMTY1YzgwMGQ5MWM.' of Response 1: Match Found
13:37:40 chan_sip.c:  Received INVITE (5) - Command in SIP INVITE
13:37:40 chan_sip.c: Setting NAT on RTP to Off
13:37:40 chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw)
13:37:40 chan_sip.c: Checking SIP call limits for device 1000
13:37:40 chan_sip.c: Updating call counter for incoming call
13:37:40 devicestate.c: No provider found, checking channel drivers for SIP
- 1000
13:37:40 chan_sip.c: Checking device state for peer 1000
13:37:40 devicestate.c: Changing state for SIP/1000 - state 2 (In use)
13:37:40 devicestate.c: device 'SIP/1000' state '2'
13:37:40 app_queue.c: Device 'SIP/1000' changed to state '2' (In use) but we
don't care because they're not a member of any queue.
13:37:40 chan_sip.c: *** Our native formats are 0x4 (ulaw)
13:37:40 chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw)
13:37:40 chan_sip.c: *** Our capabilities are 0xe (gsm|ulaw|alaw)
13:37:40 chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
13:37:40 chan_sip.c: This channel will not be able to handle video.
13:37:40 chan_sip.c: build_route: Contact hop: 
13:37:40 chan_sip.c: SIP/1000-0021a568: New call is still down Trying...

13:37:40 chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for
192.168.0.84:27928
13:37:40 devicestate.c: No provider found, checking channel drivers for SIP
- 1000
13:37:40 chan_sip.c: Checking device state for peer 1000
13:37:40 devicestate.c: Changing state for SIP/1000 - state 2 (In use)
13:37:40 devicestate.c: device 'SIP/1000' state '2'
13:37:40 app_queue.c: Device 'SIP/1000' changed to state '2' (In use) but we
don't care because they're not a member of any queue.
13:37:40 pbx.c: Launching 'Answer'
13:37:40 ] pbx.c: -- Executing [1...@from-internal:1]
Answer("SIP/1000-0021a568", "") in new stack
13:37:40 devicestate.c: No provider found, checking channel drivers for SIP
- 1000
13:37:40 chan_sip.c: Checking device state for peer 1000
13:37:40 devicestate.c: Changing state for SIP/1000 - state 2 (In use)
13:37:40 devicestate.c: device 'SIP/1000' state '2'
13:37:40 app_queue.c: Device 'SIP/1000' changed to state '2' (In use) but we
don't care because they're not a member of any queue.
13:37:40 chan_sip.c: SIP answering channel: SIP/1000-0021a568
13:37:40 chan_sip.c: Setting framing from config on incoming call
13:37:40 chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True
Text flag: True
13:37:40 chan_sip.c: ** Our prefcodec: 0x0 (nothing)
13:37:40 chan_sip.c: -- Done with adding codecs to SDP
13:37:40 channel.c: Internal timing is disabled (option_internal_timing=0
chan->timingfd=28)
13:37:40 chan_sip.c: Done building SDP. Settling with this capability: 0xc
(ulaw|alaw)
13:37:40 chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for
192.168.0.84:27928
13:37:40 rtp.c: Got RTCP report of 132 bytes
13:37:40 pbx.c: Launching 'Dial'
13:37:40 ] pbx.c: -- Executing [1...@from-internal:2]
Dial("SIP/1000-0021a568", "Mobile/Carlos/909037079681") in new stack
13:37:40 rtp.c: Channel 'Mobile/Carlos-0213' has no RTP, not doing anything
13:37:40 channel.c: Not copying variable DIALEDTIME.
13:37:40 channel.c: Not copying variable ANSWEREDTIME.
13:37:40 channel.c: Not 

Re: [asterisk-users] Bandwidth management and ADSL router

[James A. Shigley]

A lot of ISP adsl modems aren't capable. But most should be. Just log in to it 
give it a private IP for your lan(192.X.X.X or whatever your using) then have 
all your computers use that local IP as their gateway address.

If you have an ADSL modem which doesn't then simple get a router (hell a 
Linksys/Dlink $50 cheapy from wallmart would work) and have the ADSL plug into 
the router and all the stations use the router for their gateway.

If you have a spare server or virtual server space you can use Vyatta 
(Vyatta.com) it is a free open source router/firewall/vpn/few other things. 
I've never used it in a virtual environment, but I see no reason why it 
wouldn't work that way. Also note that it requires almost nothing to run so you 
can put it on an old < 1Ghz machine and It would still operate just fine.

James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.5,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
 
CONFIDENTIALITY NOTICE: This email, including any attachments, contains 
information which may be confidential or privileged. The information is 
intended to be for the use of the individual or entity named above. If you are 
not the intended recipient, be aware that any disclosure, copying, distribution 
or use of the contents of this information is prohibited. If you have received 
this email in error, please notify the sender immediately by "reply to sender 
only" message and destroy all electronic and hard copies of the communication, 
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"Common sense is the collection of prejudices acquired by age eighteen." -- 
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"Once you can accept the universe as matter expanding into nothing that is 
something,wearing stripes with plaid comes easy." -- Albert Einstein
"Theory is when you know something, but it doesn't work. Practice is when
something works, but you don't know why. Programmers combine theory and
practice: Nothing works and they don't know why.-Anonymous Developer"

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Tuesday, May 26, 2009 11:55 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Bandwidth management and ADSL router


Thanks for all.

But what all gave me was a software need to be installed on PC, but I am 
looking for a router (ADSL router) that can does this, because usually the ADSL 
router is the default gateway where all the traffic goes out and in. 

Any ADSL router device can do this?

About Draytek, as I understand that control can be done only at upload traffic 
and not download traffic, while 90% of the problem are coming from download 
traffic, so this is not the needed.

Any advise in that direction?

Regards
Bilal


  


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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Jeff LaCoursiere]

On Tue, 26 May 2009, Danny Nicholas wrote:

> The best a native cat5 can run is 100 meters.  Unless you like paying your
> telco huge bucks, you should go for some kind of SIP connection to your box.
>

He was asking about an analog telco connection - not an ethernet drop.

j

>
>
>  _
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> asterisk-us...@rogg.is
> Sent: Tuesday, May 26, 2009 9:09 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] Maximum cable length for analog phone from FXS
> port
>
>
>
> Hello.
>
>
>
> I am looking for details of the maximum allowed/usable/effective wire/cable
> length of the connection from a FXS port of Digium analog cards to the
> analog telephone handset.
>
>
>
> To clarify my intention, I need to have an analog telephone connection to my
> asterisk box that is 3000 meters (3km) away at least. If you have any
> details of ATA boxes or other similar devices that I could use to do this,
> I'd appreciate your input. It must be able to use a regular analog telephone
> handset on the far end.
>
>
>
> I've searched high and low and either I'm not clever enough in using the
> right terms for this or it is rarely documented?
>
>
>
> Any details much appreciated.
>
>
>
> Thank you!
>
> Baldvin
>
>
>
>

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[asterisk-users] SIP over VPN

[Marco Sambo]

Hi all,
I have a question. I have a VPN and I want to use a SIP softphone on my
notebook using with asterisk. But I have some problem with firewall and
port.
Someone knows which ports I should open on my firewall??? I can't connect
...

Thanks all.

Marco
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Steve Johnson]

For long distances, a wireless point-to-point might be more economical
than trenching.

e.g: Carlson Wideband CDMA Spread Spectrum Phone Line Extender
http://www.oksolar.com/communications/phone_line_ext.htm

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Re: [asterisk-users] Bandwidth management and ADSL router

[bilal ghayyad]

Thanks for all.

But what all gave me was a software need to be installed on PC, but I am 
looking for a router (ADSL router) that can does this, because usually the ADSL 
router is the default gateway where all the traffic goes out and in. 

Any ADSL router device can do this?

About Draytek, as I understand that control can be done only at upload traffic 
and not download traffic, while 90% of the problem are coming from download 
traffic, so this is not the needed.

Any advise in that direction?

Regards
Bilal


  


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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Steve Underwood]

Tzafrir Cohen wrote:
> On Tue, May 26, 2009 at 05:39:46PM +0200, randulo wrote:
>   
>> On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas  wrote:
>> 
>>> I run my analog telco over cat5, but that's in-house and definitely not 
>>> 3km. That sounds really far for current loop stuff.
>>>   
>> I was doing that too. I asked this same question a few years ago and
>> the answer was 100-200 meters. This is just a quick rule of thumb, but
>> it seems about right. 3km, I doubt that would work, but it depends, as
>> someone said, totally depending on ohm's law :)
>> 
>
> If it were to depend solely on Ohm's law, than 3km would be marginal but
> probably within reach. Not exactly sure what other factors are there to 
> count.
>   
The copper planning limit (or whatever local term your telco uses) is 
typically 10km.

Steve


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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Tzafrir Cohen]

On Tue, May 26, 2009 at 12:44:26PM -0400, Jon Pounder wrote:
> Wilton Helm wrote:
> 
> one thing I missed mentioning about fxs devices - the linksys/sipura 
> ones actually allow you to set line characteristics on the slic inside 
> it. you can vary from the 600ohm default, and tweak gains a bit. Some 
> mix of a capacitive line or different resistance may help. never tried 
> myself but there are a ton of things you can play with.

Any of those are actually important?

Which?

-- 
   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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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Danny Nicholas]

I run my analog telco over cat5, but that's in-house and definitely not 3Km.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Tuesday, May 26, 2009 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: bald...@rogg.is
Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS
port



On Tue, 26 May 2009, Danny Nicholas wrote:

> The best a native cat5 can run is 100 meters.  Unless you like paying your
> telco huge bucks, you should go for some kind of SIP connection to your
box.
>

He was asking about an analog telco connection - not an ethernet drop.

j

>
>
>  _
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> asterisk-us...@rogg.is
> Sent: Tuesday, May 26, 2009 9:09 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] Maximum cable length for analog phone from FXS
> port
>
>
>
> Hello.
>
>
>
> I am looking for details of the maximum allowed/usable/effective
wire/cable
> length of the connection from a FXS port of Digium analog cards to the
> analog telephone handset.
>
>
>
> To clarify my intention, I need to have an analog telephone connection to
my
> asterisk box that is 3000 meters (3km) away at least. If you have any
> details of ATA boxes or other similar devices that I could use to do this,
> I'd appreciate your input. It must be able to use a regular analog
telephone
> handset on the far end.
>
>
>
> I've searched high and low and either I'm not clever enough in using the
> right terms for this or it is rarely documented?
>
>
>
> Any details much appreciated.
>
>
>
> Thank you!
>
> Baldvin
>
>
>
>

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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Jon Pounder]

Wilton Helm wrote:
>
> You are exactly right. Cat 5 had no advantage over cheaper wire for 
> voice, and the length limitations are meaningless. Consider that Cat 5 
> is typically use with signals that extent to 30 MHz or beyond. A voice 
> grade analog circuit must go to 4 KHz (1/10,000 as much). At 4 KHz, 
> the wire generally doesn’t even act like a controlled impedance.
>

I completely agree, but that said I still use cat5/e for everything 
anyway, not worth having more than one kind of wire, and lets you change 
your mind later on the usage. Once you are using outside plant 
facilities though, you live with what you get, and don't expect much.

> Wilton
>
> *From:* asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David 
> Gibbons
> *Sent:* Tuesday, May 26, 2009 8:50 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Maximum cable length for analog phone 
> from FXS port
>
> I could be wrong but I don’t think the cat5 limit of 100 meters 
> applies to any analog signaling over that copper. I believe it only 
> applies to Ethernet signaling.
>
> -Dave
>
> 
>
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Re: [asterisk-users] FXS

[Jon Pounder]

Diogo Saad wrote:
> how do I configure my SIP account information? I mean, sip proxy and etc.

you need just a couple pieces of information
server (put this in any setting that says proxy or host etc, all set the 
same)
account (the extension in asterisk, put anywhere that sounds like a 
non-display only field)
password (secret, key, password etc., should be one field that takes 
this in the config)
register = yes

basically that's it.


you mean need to disable feature codes etc, but the above will get most 
any sip device working with asterisk once you setup an extension for it.
>
> On Tue, May 26, 2009 at 1:19 PM, Jon Pounder  > wrote:
>
> Diogo Saad wrote:
> > Using an ATA, Do I still need a softphone or it´s embedded in the
> > hardware?
>
> plain old walmart phone plugs in the ata (with or without callerid,
> adsi, cordless, etc)
>
> >
> > On Tue, May 26, 2009 at 12:09 PM, Steve Edwards  
> > @sedwards.com 
> > wrote:
> >
> > On Tue, 26 May 2009, Diogo Saad wrote:
> >
> > > What is the easiest way to connect my "black phone" to a
> PC running
> > > asterisk?
> > >
> > > I don't need multiple extensions, I've got just 1 phone. Is
> > there any
> > > USB FXS adapter?
> >
> > An Ethernet based ATA would be more versatile. I like Digium's
> > discontinued IAXy. Dead simple to configure, easy to travel
> with,
> > no NAT
> > headaches.
> >
> > Used on ebay should set you back about US$30.
> >
> > Thanks in advance,
> >
> 
> > Steve Edwards  sedwa...@sedwards.com
> 
> >  >  Voice: +1-760-468-3867 PST
> > Newline Fax:
> > +1-760-731-3000
> >
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> >
> >
> >
> >
> > --
> > Diogo Saad
> >
> >
> 
> >
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>
> 
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Wilton Helm]

You are exactly right.  Cat 5 had no advantage over cheaper wire for voice,
and the length limitations are meaningless.  Consider that Cat 5 is
typically use with signals that extent to 30 MHz or beyond.  A voice grade
analog circuit must go to 4 KHz (1/10,000 as much).  At 4 KHz, the wire
generally doesn't even act like a controlled impedance.

 

Wilton

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Tuesday, May 26, 2009 8:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS
port

 

I could be wrong but I don't think the cat5 limit of 100 meters applies to
any analog signaling over that copper. I believe it only applies to Ethernet
signaling.

 

-Dave

 

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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Jon Pounder]

Wilton Helm wrote:

one thing I missed mentioning about fxs devices - the linksys/sipura 
ones actually allow you to set line characteristics on the slic inside 
it. you can vary from the 600ohm default, and tweak gains a bit. Some 
mix of a capacitive line or different resistance may help. never tried 
myself but there are a ton of things you can play with.


> There are a lot of factors that impact this. First, CAT 5, while 
> usable is overkill. Cat 3 (otherwise known as I/O wire) works equally 
> well for voice grade lines. That being said, for that long a run, a 
> heavier gauge wire would be better. I believe telcos use 18 – 22 guage 
> (Cat 5 and Cat 3 are both 26 awg). This has less resistive loss.
>
> Most FXS or ATA devices use 24 volts or less for “battery”. That works 
> fine for short loops, but limits the range. A central office POTS port 
> normally uses 48 VDC which works well to several KM. If the customer 
> is at the end of a long run in a rural area, they use a “long line” 
> card which uses 75 volts. (In rural communities, they often place the 
> line cards in a roadside “remote terminal” and use statistically 
> multiplexed T1s to make it appear to the switch as a part of it.
>
> That addresses the DC characteristics, which can be reduced to ohms 
> law. A phone needs around 8 V @ .02 A. The wire resistance determine 
> the drop (E = IR) and the source voltage determines whether there will 
> be enough left. The A.C. characteristics are more complicated. The FXS 
> must do a 2 wire to 4 wire conversion, which involves matching the 
> impedance of the line. The FXS is generally designed for relatively 
> short lines, so might not be able to match either the resistance or 
> capacitance found in a long run. Heavier wire will minimize this. In 
> addition to that, the transmit side of the 2 wire to 4 wire circuit 
> must be able to drive the load it sees, and again it may not be 
> designed with a long run in mind. Finally, COs line cards have the 
> ability to adjust receive and transmit gain to compensate for sound 
> level losses in long lines. While this isn’t routinely done on simple 
> circuits, it is an option an FXS doesn’t generally have. In addition, 
> the more gain that is inserted, the harder it is to balance to 2 wire 
> to 4 wire circuit, and the more complex it has to be in order to 
> support this.
>
> Wilton
>
> 
>
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[asterisk-users] Suggest good calling service for London

[Kashif Naeem]

Hello All,

We are setting up call center of 10 agents and expecting its growth till 30
agents. Mainly calling is within UK. Please suggest some good service for UK
dialing with London DID.

Regards,
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com

Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766

Email: kas...@haditelecom.com
MSN: kashif__na...@hotmail.com
Gmail: meet.kas...@gmail.com
Skype: kashif.naeem

302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Wilton Helm]

 

There are a lot of factors that impact this.  First, CAT 5, while usable is
overkill.  Cat 3 (otherwise known as I/O wire) works equally well for voice
grade lines.  That being said, for that long a run, a heavier gauge wire
would be better.  I believe telcos use 18 - 22 guage (Cat 5 and Cat 3 are
both 26 awg).  This has less resistive loss.  

 

Most FXS or ATA devices use 24 volts or less for "battery".  That works fine
for short loops, but limits the range.  A central office POTS port normally
uses 48 VDC which works well to several KM.  If the customer is at the end
of a long run in a rural area, they use a "long line" card which uses 75
volts.  (In rural communities, they often place the line cards in a roadside
"remote terminal" and use statistically multiplexed T1s to make it appear to
the switch as a part of it.

 

That addresses the DC characteristics, which can be reduced to ohms law.  A
phone needs around 8 V @ .02 A.  The wire resistance determine the drop (E =
IR) and the source voltage determines whether there will be enough left.
The A.C. characteristics are more complicated.  The FXS must do a 2 wire to
4 wire conversion, which involves matching the impedance of the line.  The
FXS is generally designed for relatively short lines, so might not be able
to match either the resistance or capacitance found in a long run.  Heavier
wire will minimize this.  In addition to that, the transmit side of the 2
wire to 4 wire circuit must be able to drive the load it sees, and again it
may not be designed with a long run in mind.  Finally, COs line cards have
the ability to adjust receive and transmit gain to compensate for sound
level losses in long lines.  While this isn't routinely done on simple
circuits, it is an option an FXS doesn't generally have.  In addition, the
more gain that is inserted, the harder it is to balance to 2 wire to 4 wire
circuit, and the more complex it has to be in order to support this.

 

Wilton

 

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Re: [asterisk-users] FXS

[Diogo Saad]

how do I configure my SIP account information? I mean, sip proxy and etc.

On Tue, May 26, 2009 at 1:19 PM, Jon Pounder  wrote:

> Diogo Saad wrote:
> > Using an ATA, Do I still need a softphone or it´s embedded in the
> > hardware?
>
> plain old walmart phone plugs in the ata (with or without callerid,
> adsi, cordless, etc)
>
> >
> > On Tue, May 26, 2009 at 12:09 PM, Steve Edwards  > @sedwards.com > wrote:
> >
> > On Tue, 26 May 2009, Diogo Saad wrote:
> >
> > > What is the easiest way to connect my "black phone" to a PC running
> > > asterisk?
> > >
> > > I don't need multiple extensions, I've got just 1 phone. Is
> > there any
> > > USB FXS adapter?
> >
> > An Ethernet based ATA would be more versatile. I like Digium's
> > discontinued IAXy. Dead simple to configure, easy to travel with,
> > no NAT
> > headaches.
> >
> > Used on ebay should set you back about US$30.
> >
> > 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 mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
> > --
> > Diogo Saad
> >
> > 
> >
> > ___
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> >
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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>



-- 
Diogo Saad
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Cary Fitch]

Excellent analysis of the real world.  Start with this, and work out the
issues, or go to VOIP.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wilton Helm
Sent: Tuesday, May 26, 2009 11:33 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Maximum cable length for analog phone fromFXS
port

 

 

There are a lot of factors that impact this.  First, CAT 5, while usable is
overkill.  Cat 3 (otherwise known as I/O wire) works equally well for voice
grade lines.  That being said, for that long a run, a heavier gauge wire
would be better.  I believe telcos use 18 - 22 guage (Cat 5 and Cat 3 are
both 26 awg).  This has less resistive loss.  

 

Most FXS or ATA devices use 24 volts or less for "battery".  That works fine
for short loops, but limits the range.  A central office POTS port normally
uses 48 VDC which works well to several KM.  If the customer is at the end
of a long run in a rural area, they use a "long line" card which uses 75
volts.  (In rural communities, they often place the line cards in a roadside
"remote terminal" and use statistically multiplexed T1s to make it appear to
the switch as a part of it.

 

That addresses the DC characteristics, which can be reduced to ohms law.  A
phone needs around 8 V @ .02 A.  The wire resistance determine the drop (E =
IR) and the source voltage determines whether there will be enough left.
The A.C. characteristics are more complicated.  The FXS must do a 2 wire to
4 wire conversion, which involves matching the impedance of the line.  The
FXS is generally designed for relatively short lines, so might not be able
to match either the resistance or capacitance found in a long run.  Heavier
wire will minimize this.  In addition to that, the transmit side of the 2
wire to 4 wire circuit must be able to drive the load it sees, and again it
may not be designed with a long run in mind.  Finally, COs line cards have
the ability to adjust receive and transmit gain to compensate for sound
level losses in long lines.  While this isn't routinely done on simple
circuits, it is an option an FXS doesn't generally have.  In addition, the
more gain that is inserted, the harder it is to balance to 2 wire to 4 wire
circuit, and the more complex it has to be in order to support this.

 

Wilton

 

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Re: [asterisk-users] SIP over VPN

[David Gibbons]

Assuming you mean the firewall in front of the client, you don't need to open 
any ports as long as the VPN client is tunneling all traffic to and from the 
Asterisk server.

I  set NAT=yes in the config file for the extensions behind a VPN.

-Dave

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Sambo
Sent: Tuesday, May 26, 2009 11:21 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP over VPN

Hi all,
I have a question. I have a VPN and I want to use a SIP softphone on my 
notebook using with asterisk. But I have some problem with firewall and port.
Someone knows which ports I should open on my firewall??? I can't connect ...

Thanks all.

Marco
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Jon Pounder]

Miguel Molina wrote:
> randulo escribió:
>> On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas  wrote:
>>   
>>> I run my analog telco over cat5, but that's in-house and definitely not 
>>> 3km. That sounds really far for current loop stuff.
>>> 
>>
>> I was doing that too. I asked this same question a few years ago and
>> the answer was 100-200 meters. This is just a quick rule of thumb, but
>> it seems about right. 3km, I doubt that would work, but it depends, as
>> someone said, totally depending on ohm's law :)
>>   
> What I think about this is, the length of the copper cable between the 
> central office and home is usually several km, but definitely helped 
> by the central office circuitry (current source instead of voltage 
> source, that guarantees a minimum ringing voltage on the far end). 
> What I don't know is, a FXS port behaves the same as a central office, 
> electrically speaking? If that is so, you could extend your 3km of 
> cable without problems, but I think you can have some noise problems 
> depending on what places the cable has to go through.

from the ringing point of view, the CO ring generator is usually truly a 
sine wave and this propagates well through a cable. The cheap fxs ports 
are mostly square waves and lower voltages with limited current sourcing 
(check the REN numbers they are capable of ringing for a comparison if 
its listed) some of the cheap ones have trouble ringing a phone plugged 
in with a short line cord. So in addition to being frequency choked by 
the long run the square wave will get reduced in amplitude, and it may 
well have been marginal amplitude to begin with.

so depending what fxs hardware you have driving it and the load from the 
phone, results will range from works perfectly to does not work at all.




>
>> r
>>
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>
>
> -- 
> Ing. Miguel Molina
> Grupo de Tecnología
> Millenium Phone Center
> PBX: (+57 1)6500800 ext. 1201
> Fax: (+57 1)6500816
> Móvil: (+57)3138873587 
>   
> 
>
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[John Novack]

You are correct.
Telcos normally supply dial tone to business and residence for miles if 
there is no DSL
Loading coils are used to offset the capacitance of cables, and precise 
spacing of these is required, and are engineered for different types of 
cable.

John Novack


David Gibbons wrote:
>
> I could be wrong but I don't think the cat5 limit of 100 meters 
> applies to any analog signaling over that copper. I believe it only 
> applies to Ethernet signaling.
>
> -Dave
>
> *From:* asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny 
> Nicholas
> *Sent:* Tuesday, May 26, 2009 10:41 AM
> *To:* bald...@rogg.is; 'Asterisk Users Mailing List - Non-Commercial 
> Discussion'
> *Subject:* Re: [asterisk-users] Maximum cable length for analog phone 
> from FXS port
>
> The best a native cat5 can run is 100 meters. Unless you like paying 
> your telco huge bucks, you should go for some kind of SIP connection 
> to your box.
>
> 
>
> *From:* asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
> *asterisk-us...@rogg.is
> *Sent:* Tuesday, May 26, 2009 9:09 AM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* [asterisk-users] Maximum cable length for analog phone from 
> FXS port
>
> Hello.
>
> I am looking for details of the maximum allowed/usable/effective 
> wire/cable length of the connection from a FXS port of Digium analog 
> cards to the analog telephone handset.
>
> To clarify my intention, I need to have an analog telephone connection 
> to my asterisk box that is 3000 meters (3km) away at least. If you 
> have any details of ATA boxes or other similar devices that I could 
> use to do this, I'd appreciate your input. It must be able to use a 
> regular analog telephone handset on the far end.
>
> I've searched high and low and either I'm not clever enough in using 
> the right terms for this or it is rarely documented?
>
> Any details much appreciated.
>
> Thank you!
>
> Baldvin
>
> 
>
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Dog is my co-pilot


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Re: [asterisk-users] FXS

[Jon Pounder]

Diogo Saad wrote:
> Using an ATA, Do I still need a softphone or it´s embedded in the 
> hardware?

plain old walmart phone plugs in the ata (with or without callerid, 
adsi, cordless, etc)

>
> On Tue, May 26, 2009 at 12:09 PM, Steve Edwards  @sedwards.com > wrote:
>
> On Tue, 26 May 2009, Diogo Saad wrote:
>
> > What is the easiest way to connect my "black phone" to a PC running
> > asterisk?
> >
> > I don't need multiple extensions, I've got just 1 phone. Is
> there any
> > USB FXS adapter?
>
> An Ethernet based ATA would be more versatile. I like Digium's
> discontinued IAXy. Dead simple to configure, easy to travel with,
> no NAT
> headaches.
>
> Used on ebay should set you back about US$30.
>
> Thanks in advance,
> 
> Steve Edwards  sedwa...@sedwards.com
>   Voice: +1-760-468-3867 PST
> Newline Fax:
> +1-760-731-3000
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> -- 
> Diogo Saad
>
> 
>
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Re: [asterisk-users] FXS

[Diogo Saad]

Using an ATA, Do I still need a softphone or it´s embedded in the hardware?

On Tue, May 26, 2009 at 12:09 PM, Steve Edwards
wrote:

> On Tue, 26 May 2009, Diogo Saad wrote:
>
> > What is the easiest way to connect my "black phone" to a PC running
> > asterisk?
> >
> > I don't need multiple extensions, I've got just 1 phone. Is there any
> > USB FXS adapter?
>
> An Ethernet based ATA would be more versatile. I like Digium's
> discontinued IAXy. Dead simple to configure, easy to travel with, no NAT
> headaches.
>
> Used on ebay should set you back about US$30.
>
> Thanks in advance,
> 
> Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
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-- 
Diogo Saad
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Re: [asterisk-users] FXS

[Lee Spenadel]

How about a low cost ATA?   Just plug the ATA into the network, configure it
- along with a SIP definition within sip.conf and you're ready to go.

 

Lee

 

From: Diogo Saad [mailto:diogos...@gmail.com] 
Sent: Tuesday, May 26, 2009 10:41 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] FXS

 

What is the easiest way to connect my "black phone" to a PC running
asterisk?

I don't need multiple extensions, I've got just 1 phone. Is there any USB
FXS adapter?

Thanks

-- 
Diogo Saad

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[asterisk-users] CDR after SIP blind transfer.

[Chris Maciejewski]

Hi,

I can't get Asterisk to save CDRs for calls transferred via SIP blind transfer.

My extensions.conf:
[globals]
__TRANSFER_CONTEXT = transfer

[common]
exten => 123,1,Playback(demo-congrats)
exten => 123,n,Hangup()

exten => _0X.,1,Dial(SIP/${ext...@pstn-gw,60)
exten => _0X.,n,Hangup()

exten => i,1,Hangup()
exten => h,1,Hangup()
exten => t,1,Hangup()

[transfer]
exten => 123,1,Goto(common,${EXTEN},1)

Scenario A:
SIP Phone dials 123 and hangs up after 10 seconds.
CDR is recorded just fine.

Scenario B:
SIP Phone dials 02088441234 which is routed to the external peer.
After 10 seconds call is transferred (blindly) to extension 123. After
another 10 seconds external peer hangs up.

Problem: there is only one CDR recorded for the first 10 seconds long
call. Second part of the call, after 02088441234 was transferred to
123 is NOT recorded.

Is there any way to force Asterisk to record CDR in scenario B
(without using LOCAL channel)?

Regards,
Chris

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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Cary Fitch]

Also be wary of the loop you get.  

 

Depending on the Telco you are dealing with, and the type of loop you get,
"Alarm circuit, etc." they may , and have the right to, put in  a low pass
circuit to limit bandwidth to 15 Hz.  That keeps people from using cheap
alarm circuits for voice.  It is not likely they will go to the trouble.
But, they can do it.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Tuesday, May 26, 2009 9:26 AM
To: bald...@rogg.is; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS
port

 

That is a pretty long run.
The type of analog phone can be an issue. How LITTLE loop current will it
operate on? Most need more than 20 Ma to signal properly, and the voltage
output of the ATA needs to be known
Type of signaling? DTMF? pulse?
Interconnection cable wire size and capacitance will affect high frequency
response, loop current, inductive pickup and pulse shaping to name just a
few. The ATA requirements need to be known. A total loop resistance of 500
ohms should work, but go out to 1200 and most will fail
Do you really have control over this or will you be renting a pair from the
local telco?
Protection should be applied on both ends for safety of the user(s) and
devices.
There MUST be a better way???

asterisk-us...@rogg.is wrote: 

Hello.

 

I am looking for details of the maximum allowed/usable/effective wire/cable
length of the connection from a FXS port of Digium analog cards to the
analog telephone handset.

 

To clarify my intention, I need to have an analog telephone connection to my
asterisk box that is 3000 meters (3km) away at least. If you have any
details of ATA boxes or other similar devices that I could use to do this,
I'd appreciate your input. It must be able to use a regular analog telephone
handset on the far end.

 

I've searched high and low and either I'm not clever enough in using the
right terms for this or it is rarely documented?

 

Any details much appreciated.

 

Thank you!

Baldvin

 

 





  _  



 
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Heath Roberts]

On Tue, May 26, 2009 at 10:09 AM,  wrote:

>  I am looking for details of the maximum allowed/usable/effective
> wire/cable length of the connection from a FXS port of Digium analog cards
> to the analog telephone handset.
>
>
>
> To clarify my intention, I need to have an analog telephone connection to
> my asterisk box that is 3000 meters (3km) away at least. If you have any
> details of ATA boxes or other similar devices that I could use to do this,
> I‘d appreciate your input. It must be able to use a regular analog telephone
> handset on the far end.
>
>
>
> I‘ve searched high and low and either I‘m not clever enough in using the
> right terms for this or it is rarely documented?
>
I've not seen a high-current ATA, but you could probably add a KIT8L from
here: http://www.sandman.com/longloop.html to a regular terminal adapter to
boost the loop voltage/current.

--
Heath Roberts
htrobe...@gmail.com
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Miguel Molina]

randulo escribió:

On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas  wrote:
  

I run my analog telco over cat5, but that's in-house and definitely not 3km. 
That sounds really far for current loop stuff.



I was doing that too. I asked this same question a few years ago and
the answer was 100-200 meters. This is just a quick rule of thumb, but
it seems about right. 3km, I doubt that would work, but it depends, as
someone said, totally depending on ohm's law :)
  
What I think about this is, the length of the copper cable between the 
central office and home is usually several km, but definitely helped by 
the central office circuitry (current source instead of voltage source, 
that guarantees a minimum ringing voltage on the far end). What I don't 
know is, a FXS port behaves the same as a central office, electrically 
speaking? If that is so, you could extend your 3km of cable without 
problems, but I think you can have some noise problems depending on what 
places the cable has to go through.



r

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--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
PBX: (+57 1)6500800 ext. 1201
Fax: (+57 1)6500816
Móvil: (+57)3138873587 

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[asterisk-users] How to register with TCP transport ?

[Olivier]

Hi,

In my sip.conf, I've got :
[general](+)
;   
register=>tcp://trunk4ipbx:passw...@192.168.100.129

register=>trunk4ipbx:passw...@192.168.100.129

When I'm using the TCP line instead of the other, I've got :
[May 26 17:58:42] NOTICE[2859]: chan_sip.c:20169 sip_parse_host: '/' is not
a valid port number on line 25 of sip.conf. using default.
[May 26 17:58:42] WARNING[2859]: chan_sip.c:6560 sip_register: Format for
registration is
[transport://]user[:secret[:authuse...@domain[:port][/extension][~expiry] at
line 25


Is this 
"register=>tcp://trunk4ipbx:passw...@192.168.100.129"
statement correct ?

Regards
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[asterisk-users] FXS

[Diogo Saad]

What is the easiest way to connect my "black phone" to a PC running
asterisk?

I don't need multiple extensions, I've got just 1 phone. Is there any USB
FXS adapter?

Thanks

-- 
Diogo Saad
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Re: [asterisk-users] Maximum cable length for analog phone from FXSport

[Cary Fitch]

Sigh, lets repeal Ohm's law.
;-)

In practice the controlling rules are:

Murphy's Law:  "If anything can go wrong it will."

O'Toole's corollary to Murphy's law:  "And, it will produce the worst
possible results."

Cary Fitch

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Tuesday, May 26, 2009 10:56 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Maximum cable length for analog phone from
FXSport

On Tue, May 26, 2009 at 05:39:46PM +0200, randulo wrote:
> On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas  wrote:
> > I run my analog telco over cat5, but that's in-house and definitely not
3km. That sounds really far for current loop stuff.
> 
> I was doing that too. I asked this same question a few years ago and
> the answer was 100-200 meters. This is just a quick rule of thumb, but
> it seems about right. 3km, I doubt that would work, but it depends, as
> someone said, totally depending on ohm's law :)

If it were to depend solely on Ohm's law, than 3km would be marginal but
probably within reach. Not exactly sure what other factors are there to 
count.

-- 
   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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Re: [asterisk-users] FXS

[Steve Edwards]

On Tue, 26 May 2009, Diogo Saad wrote:

> What is the easiest way to connect my "black phone" to a PC running 
> asterisk?
>
> I don't need multiple extensions, I've got just 1 phone. Is there any 
> USB FXS adapter?

An Ethernet based ATA would be more versatile. I like Digium's 
discontinued IAXy. Dead simple to configure, easy to travel with, no NAT 
headaches.

Used on ebay should set you back about US$30.

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] Maximum cable length for analog phone from FXS port

[Tzafrir Cohen]

On Tue, May 26, 2009 at 05:39:46PM +0200, randulo wrote:
> On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas  wrote:
> > I run my analog telco over cat5, but that's in-house and definitely not 
> > 3km. That sounds really far for current loop stuff.
> 
> I was doing that too. I asked this same question a few years ago and
> the answer was 100-200 meters. This is just a quick rule of thumb, but
> it seems about right. 3km, I doubt that would work, but it depends, as
> someone said, totally depending on ohm's law :)

If it were to depend solely on Ohm's law, than 3km would be marginal but
probably within reach. Not exactly sure what other factors are there to 
count.

-- 
   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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Re: [asterisk-users] Converting Cisco 7961 to SIP

[David Gibbons]

Cory,

Precisely what do you mean by 'Anything other than Callmanager will essentially 
be a "hack"'?

I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP 
image. They're not 'hacked', they're set up properly against the Cisco provided 
SIP image and are rock-solid stable. I would pit them against any of the 
cheaper model SIP phones any time, any place, any day.

I've written scripts to do nearly everything that call manager can do without 
paying hundreds of dollars per user for the call manager software. Just about 
the only thing they can't do at the moment is BLF because they require SIP over 
TCP to handle SIP messages about BLF status, something that I'm not willing to 
implement just yet.

In the past, Cisco phones have had a bad rap as not being usable outside of a 
call manager environment. That's just not the case.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews
Sent: Tuesday, May 26, 2009 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP

Darrin,

The files you are using are consistent with SIP for Cisco Call Manager. 
Anything other than Callmanager will essentially be a "hack". I am not sure how 
proprietary the Avaya system is in regards to registration and "open-SIP" 
support. Asterisk and any iteration of it will support it, but Cisco hasn't 
really designed a load compatible with it yet. I can tell you that I haven't 
really found any configuration file generation tools for these files. The 
reason being is that these loads are mainly used for SCCP and SIP Cisco 
systems. There is a well known tutorial on how to "Hack to the CP-7970 to 
trixbox CE located here:

http://www.asterisktutorials.com/cisco-7970-ip-phone/

This may help get you jump started and pointed in the right direction. The only 
problem that may arise is that in the tutorial, the use a specific SIP load 
(8.0.3) which may not be available for the 7961G.

Cory J. Andrews
Director New Market Initiatives

Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
candr...@sayersmedia.com


Have I exceeded your expectations?  Please share your experience with my boss,  
Benjamin P. Sayers, CEO

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-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin Henshaw
Sent: Tuesday, May 26, 2009 7:40 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Converting Cisco 7961 to SIP

As part of a project to move a clients Cisco phones to SIP to work with an 
Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk 
setup. Now, I've gotten the firmware files from the site, the latest version is 
8.4 which contains the following files:

apps41.8-4-3-16.sbn
cnu41.8-4-3-16.sbn
cvm41sip.8-4-3-16.sbn
dsp41.8-4-3-16.sbn
jar41sip.8-4-3-16.sbn
SIP41.8-4-4S.loads
term41.default.loads
term61.default.loads

Now I've read over loads of documentation on it, but am getting tripped up. 
Most of what I've seen talks about the older firmware versions usually 7.4. I 
have a feeling I'm still missing a lot of stuff. Anyone have any recent links 
or information?

Also, anyone know of a decent way to generate the config files? I'd hate to 
have to go through all of it manually? Thanks.

Cheers,

Darrin Henshaw

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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[John Novack]

That is a pretty long run.
The type of analog phone can be an issue. How LITTLE loop current will 
it operate on? Most need more than 20 Ma to signal properly, and the 
voltage output of the ATA needs to be known

Type of signaling? DTMF? pulse?
Interconnection cable wire size and capacitance will affect high 
frequency response, loop current, inductive pickup and pulse shaping to 
name just a few. The ATA requirements need to be known. A total loop 
resistance of 500 ohms should work, but go out to 1200 and most will fail
Do you really have control over this or will you be renting a pair from 
the local telco?
Protection should be applied on both ends for safety of the user(s) and 
devices.

There MUST be a better way???

asterisk-us...@rogg.is wrote:


Hello.

 

I am looking for details of the maximum allowed/usable/effective 
wire/cable length of the connection from a FXS port of Digium analog 
cards to the analog telephone handset.


 

To clarify my intention, I need to have an analog telephone connection 
to my asterisk box that is 3000 meters (3km) away at least. If you 
have any details of ATA boxes or other similar devices that I could 
use to do this, I'd appreciate your input. It must be able to use a 
regular analog telephone handset on the far end.


 

I've searched high and low and either I'm not clever enough in using 
the right terms for this or it is rarely documented?


 


Any details much appreciated.

 


Thank you!

Baldvin

 




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Re: [asterisk-users] howto store local exchange prefixes ?

[Barry L. Kline]

sean darcy wrote:

> Maybe I've not explained this correctly. I know, or can look up, the 40+ 
> local exchanges that are local. I can parse the dial EXTEN to determine 
> the exchange. I can check the exchange against a DB. I want to determine 
> which exchanges are "local". I do not want to store an exchange dialed 
> by a user.

I didn't explain myself very well.

My Asterisk system sits between the PSTN and a legacy PBX.  Asterisk
answers the call and among other things, prompts for an extension
number.   I needed to know if the extension entered is valid before
sending the call on to the old PBX.  I simply have a lookup subroutine
to validate the extension.

My code for looking up the validity of their entry is:

exten => _[123]XX,1,Verbose(1,${CALLERID(all)} requested extension
${EXTEN});
exten => _[123]XX,n,Gosub(validate-extension,s,1(${EXTEN}));
exten => _[123]XX,n,Goto(extension-${GOSUB_RETVAL});
exten => _[123]XX,n(extension-FOUND),Verbose(1,${CALLERID(all)} xfer to
${DB(${DB_IWATSU_EXTENSIONS}/${EXTEN})} at extension ${EXTEN});
exten => _[123]XX,n,macro(bridge-to-iwatsu,7${EXTEN});
exten => _[123]XX,n(extension-NOTFOUND),background(pbx-invalid);
exten => _[123]XX,n,WaitExten(5);



The lookup, which will initialize the AsteriskDB if necessary, is:

;
; This subroutine's purpose is to check the validity of an extension.
;
; Parameters:
;  ARG1 = Extension to check
; Returns:
;  FOUND or NOTFOUND
;
[validate-extension]
exten => s,1,Verbose(1,Checking validity of extension ${ARG1});
;
; Let's check to ensure that the database is loaded.  We'll do
; that by looking for extension 399, the Iwatsu master phone.
;
exten => s,n,GotoIf(${DB_EXISTS(${DB_IWATSU_EXTENSIONS}/399)}?search:load)
exten => s,n(load),DBdeltree(${DB_IWATSU_EXTENSIONS}); Clear all
existing entries
exten => s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/120)='Rikki')
exten => s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/121)='Terri')
exten => s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/122)='CorpConf')
exten => s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/123)='Linda')
exten => s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/124)='Kim')
exten => s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/125)='Nancy B')
exten => s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/126)='Wayne')
...
;
; Extension 399 is the master extension for the Iwatsu
; and should always show up. It is used for testing
; the validity of the database in the dialplan.
;
exten => s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/399)='MASTER')
;
; Search here
;
exten =>
s,n(search),GotoIf(${DB_EXISTS(${DB_IWATSU_EXTENSIONS}/${ARG1})}?found:notfound)
exten => s,n(found),Return(FOUND);
exten => s,n(notfound),Return(NOTFOUND);








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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Jeff LaCoursiere]

On Tue, 26 May 2009, randulo wrote:

> On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas  wrote:
>> I run my analog telco over cat5, but that's in-house and definitely not 3km. 
>> That sounds really far for current loop stuff.
>
> I was doing that too. I asked this same question a few years ago and
> the answer was 100-200 meters. This is just a quick rule of thumb, but
> it seems about right. 3km, I doubt that would work, but it depends, as
> someone said, totally depending on ohm's law :)
>

Egad, this is just not true.  100 - 200 meters is for ETHERNET, not analog 
voice.  I have many runs over 1K meters that work just fine, and several 
that are close to 3Km that honestly do NOT.  Think about high rise 
buildings - many strung with CAT3 cable for voice from the basement. 
Many of those runs may be well over 1000m.

A better question is why is he stuck using a 3Km leased circuit?  Like 
another poster said "there must be a better way".

j

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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Steve Howes]

On 26 May 2009, at 16:39, Jeff LaCoursiere wrote:
> YMMV

I think thats the problem :D sorry couldn't resist..

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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Jeff LaCoursiere]

On Tue, 26 May 2009, Steve Howes wrote:

> On 26 May 2009, at 16:39, Jeff LaCoursiere wrote:
>> YMMV
>
> I think thats the problem :D sorry couldn't resist..
>

I did kind of mean that tounge-in-cheek :):)


j

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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[randulo]

On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas  wrote:
> I run my analog telco over cat5, but that's in-house and definitely not 3km. 
> That sounds really far for current loop stuff.

I was doing that too. I asked this same question a few years ago and
the answer was 100-200 meters. This is just a quick rule of thumb, but
it seems about right. 3km, I doubt that would work, but it depends, as
someone said, totally depending on ohm's law :)

r

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Re: [asterisk-users] FXS

[Diogo Saad]

what I want to do is to answers to mobile calls using a regular phone.

Is a "usb fxs" all I need? Does this "u100" have smooth integration with
Asterisk ?


On Tue, May 26, 2009 at 11:55 AM, Geraint Lee  wrote:

> There is indeed... well i was about to say there was, but it turns out the
> one i've got is an fxo adapter, have a look and see if sangoma have any fxs
> adapters in the series, it seems to be called the usbfxo u100
>
> 2009/5/26 Diogo Saad 
>
>> What is the easiest way to connect my "black phone" to a PC running
>> asterisk?
>>
>> I don't need multiple extensions, I've got just 1 phone. Is there any USB
>> FXS adapter?
>>
>> Thanks
>>
>> --
>> Diogo Saad
>>
>>
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-- 
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Jeff LaCoursiere]

On Tue, 26 May 2009, Danny Nicholas wrote:

> I run my analog telco over cat5, but that's in-house and definitely not 3Km.

Of course - and that is just fine.  If you were running ethernet 
signalling over that CAT5 than your 100m limit would apply.  If you were 
running gigabit over that same cable its more like 80 feet.  He isn't 
asking about ethernet signalling.  As many posts have shown this morning, 
the actual length limit for running analog voice and DTMF signalling over 
the cable depends on many things that the OP probably has no control over.
YMMV, like most things.

j

>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
> LaCoursiere
> Sent: Tuesday, May 26, 2009 10:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: bald...@rogg.is
> Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS
> port
>
>
>
> On Tue, 26 May 2009, Danny Nicholas wrote:
>
>> The best a native cat5 can run is 100 meters.  Unless you like paying your
>> telco huge bucks, you should go for some kind of SIP connection to your
> box.
>>
>
> He was asking about an analog telco connection - not an ethernet drop.
>
> j
>
>>
>>
>>  _
>>
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
>> asterisk-us...@rogg.is
>> Sent: Tuesday, May 26, 2009 9:09 AM
>> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>> Subject: [asterisk-users] Maximum cable length for analog phone from FXS
>> port
>>
>>
>>
>> Hello.
>>
>>
>>
>> I am looking for details of the maximum allowed/usable/effective
> wire/cable
>> length of the connection from a FXS port of Digium analog cards to the
>> analog telephone handset.
>>
>>
>>
>> To clarify my intention, I need to have an analog telephone connection to
> my
>> asterisk box that is 3000 meters (3km) away at least. If you have any
>> details of ATA boxes or other similar devices that I could use to do this,
>> I'd appreciate your input. It must be able to use a regular analog
> telephone
>> handset on the far end.
>>
>>
>>
>> I've searched high and low and either I'm not clever enough in using the
>> right terms for this or it is rarely documented?
>>
>>
>>
>> Any details much appreciated.
>>
>>
>>
>> Thank you!
>>
>> Baldvin
>>
>>
>>
>>
>
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Re: [asterisk-users] A problem in playing sound files

[Steve Edwards]

On Tue, 26 May 2009, abdelkader wrote:

> I have 8 DID: 7 from a provider1 and 1 from provider2.
>
> Each time a customer calls one of the DID, the system plays a message.
>
> The problem is that the message is played normally for all the DIDs from the
> provider1 and is not played (not heard) for the DID from provider2.

Codecs would be a good place to start. Maybe NAT issues.

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 Cisco 7961 to SIP

[Cory Andrews]

Please do!

Cory J. Andrews
Director New Market Initiatives
 
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
candr...@sayersmedia.com


Have I exceeded your expectations?  Please share your experience with my boss,  
Benjamin P. Sayers, CEO

NOTICE: The information contained in this email and any document attached 
hereto is intended only for the named recipient(s). It is the property of the 
VoIP Supply, LLC and shall not be used, disclosed or reproduced without the 
express written consent of VoIP Supply, LLC. If you are not the intended 
recipient, nor the employee or agent responsible for delivering this message in 
confidence to the intended recipient(s), you are hereby notified that you have 
received this transmittal in error, and any review, dissemination, distribution 
or copying of this transmittal or its attachments is strictly prohibited. If 
you have received this transmittal and/or attachments in error, please notify 
me immediately by reply e-mail or telephone and then delete this message, 
including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 
14225 USA. 



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Tuesday, May 26, 2009 10:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP

Ahh I see.

In response to your other question about the auto-provisioning of Cisco phones, 
I wrote some scripts that work against an active directory and setup the phones 
automagically. I'll send the link your way if you'd like.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews
Sent: Tuesday, May 26, 2009 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP

Did not mean to infer they don't perform wonderfully with Asterisk.  By "hack" 
I meant that Cisco does not offer any official support for them on Asterisk.

Cory J. Andrews
Director New Market Initiatives

Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
candr...@sayersmedia.com


Have I exceeded your expectations?  Please share your experience with my boss,  
Benjamin P. Sayers, CEO

NOTICE: The information contained in this email and any document attached 
hereto is intended only for the named recipient(s). It is the property of the 
VoIP Supply, LLC and shall not be used, disclosed or reproduced without the 
express written consent of VoIP Supply, LLC. If you are not the intended 
recipient, nor the employee or agent responsible for delivering this message in 
confidence to the intended recipient(s), you are hereby notified that you have 
received this transmittal in error, and any review, dissemination, distribution 
or copying of this transmittal or its attachments is strictly prohibited. If 
you have received this transmittal and/or attachments in error, please notify 
me immediately by reply e-mail or telephone and then delete this message, 
including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 
14225 USA.



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Tuesday, May 26, 2009 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP

Cory,

Precisely what do you mean by 'Anything other than Callmanager will essentially 
be a "hack"'?

I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP 
image. They're not 'hacked', they're set up properly against the Cisco provided 
SIP image and are rock-solid stable. I would pit them against any of the 
cheaper model SIP phones any time, any place, any day.

I've written scripts to do nearly everything that call manager can do without 
paying hundreds of dollars per user for the call manager software. Just about 
the only thing they can't do at the moment is BLF because they require SIP over 
TCP to handle SIP messages about BLF status, something that I'm not willing to 
implement just yet.

In the past, Cisco phones have had a bad rap as not being usable outside of a 
call manager environment. That's just not the case.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews
Sent: Tuesday, May 26, 2009 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP

Darrin,

The files you are using are consistent with SIP for Cisco Call Manager. 
Anything other than Callmanager will esse

Re: [asterisk-users] FXS

[Geraint Lee]

There is indeed... well i was about to say there was, but it turns out the
one i've got is an fxo adapter, have a look and see if sangoma have any fxs
adapters in the series, it seems to be called the usbfxo u100

2009/5/26 Diogo Saad 

> What is the easiest way to connect my "black phone" to a PC running
> asterisk?
>
> I don't need multiple extensions, I've got just 1 phone. Is there any USB
> FXS adapter?
>
> Thanks
>
> --
> Diogo Saad
>
>
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Re: [asterisk-users] asterisk-addon 1.6.1 problem

[Tilghman Lesher]

On Tuesday 26 May 2009 02:52:18 Rilawich Ango wrote:
> Hi all,
>   I download asterisk-addon 1.6.1 but the VoIP phone failed to
> register to the system with the message below.
>
> [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317
> realtime_mysql: MySQL RealTime: Invalid database specified: asterisk
> [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317
> realtime_mysql: MySQL RealTime: Invalid database specified: asterisk
> sip
>
> I use the same configuration file (res_mysql.conf & extconfig.conf) in
> 1.6.0 but failed.  Any big change in 1.6.1?

Please read UPGRADE.txt in the asterisk-addons directory.

-- 
Tilghman

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Re: [asterisk-users] Problem running Dahdi

[Tzafrir Cohen]

On Mon, May 25, 2009 at 10:27:22AM -0400, Mike wrote:
> I did run make install, probably 3-4 times before I ended up asking that
> question in the mailing list.
> 
> Here is the required output: to the first one, "could not find module
> dahdi".
> 
> To the second, it found dahdi in /lib/modules/2.6.18-128.1.10.el5/dahdi

Either you have not run depmod (which is strange, it is part of 'make
install') or '2.6.18-128.1.10.el5' is not your kernel version.

>  - What kernel version?
> 2.6.18-128.1.10.el5

What is the output of:

  uname -r

If it is exactly '2.6.18-128.1.10.el5' , then try:

  depmod
  modinfo dahdi

-- 
   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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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Lyle Giese]

asterisk-us...@rogg.is wrote:
>
> Hello.
>
>  
>
> I am looking for details of the maximum allowed/usable/effective
> wire/cable length of the connection from a FXS port of Digium analog
> cards to the analog telephone handset.
>
>  
>
> To clarify my intention, I need to have an analog telephone connection
> to my asterisk box that is 3000 meters (3km) away at least. If you
> have any details of ATA boxes or other similar devices that I could
> use to do this, I'd appreciate your input. It must be able to use a
> regular analog telephone handset on the far end.
>
>  
>
> I've searched high and low and either I'm not clever enough in using
> the right terms for this or it is rarely documented?
>
>  
>
> Any details much appreciated.
>
>  
>
> Thank you!
>
> Baldvin
>
>  
>
It's not expressed in distance.  They will supply the current & voltage
output and you need to apply ohm's law.  That requires knowing the
resistance of the cable which is dependent on length and gauge.

Lyle Giese
LCR Computer Services, Inc.

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Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[Danny Nicholas]

The best a native cat5 can run is 100 meters.  Unless you like paying your
telco huge bucks, you should go for some kind of SIP connection to your box.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
asterisk-us...@rogg.is
Sent: Tuesday, May 26, 2009 9:09 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Maximum cable length for analog phone from FXS
port

 

Hello.

 

I am looking for details of the maximum allowed/usable/effective wire/cable
length of the connection from a FXS port of Digium analog cards to the
analog telephone handset.

 

To clarify my intention, I need to have an analog telephone connection to my
asterisk box that is 3000 meters (3km) away at least. If you have any
details of ATA boxes or other similar devices that I could use to do this,
I'd appreciate your input. It must be able to use a regular analog telephone
handset on the far end.

 

I've searched high and low and either I'm not clever enough in using the
right terms for this or it is rarely documented?

 

Any details much appreciated.

 

Thank you!

Baldvin

 

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Re: [asterisk-users] Converting Cisco 7961 to SIP

[Cory Andrews]

Did not mean to infer they don't perform wonderfully with Asterisk.  By "hack" 
I meant that Cisco does not offer any official support for them on Asterisk.  

Cory J. Andrews
Director New Market Initiatives
 
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
candr...@sayersmedia.com


Have I exceeded your expectations?  Please share your experience with my boss,  
Benjamin P. Sayers, CEO

NOTICE: The information contained in this email and any document attached 
hereto is intended only for the named recipient(s). It is the property of the 
VoIP Supply, LLC and shall not be used, disclosed or reproduced without the 
express written consent of VoIP Supply, LLC. If you are not the intended 
recipient, nor the employee or agent responsible for delivering this message in 
confidence to the intended recipient(s), you are hereby notified that you have 
received this transmittal in error, and any review, dissemination, distribution 
or copying of this transmittal or its attachments is strictly prohibited. If 
you have received this transmittal and/or attachments in error, please notify 
me immediately by reply e-mail or telephone and then delete this message, 
including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 
14225 USA. 



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Tuesday, May 26, 2009 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP

Cory,

Precisely what do you mean by 'Anything other than Callmanager will essentially 
be a "hack"'?

I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP 
image. They're not 'hacked', they're set up properly against the Cisco provided 
SIP image and are rock-solid stable. I would pit them against any of the 
cheaper model SIP phones any time, any place, any day.

I've written scripts to do nearly everything that call manager can do without 
paying hundreds of dollars per user for the call manager software. Just about 
the only thing they can't do at the moment is BLF because they require SIP over 
TCP to handle SIP messages about BLF status, something that I'm not willing to 
implement just yet.

In the past, Cisco phones have had a bad rap as not being usable outside of a 
call manager environment. That's just not the case.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews
Sent: Tuesday, May 26, 2009 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP

Darrin,

The files you are using are consistent with SIP for Cisco Call Manager. 
Anything other than Callmanager will essentially be a "hack". I am not sure how 
proprietary the Avaya system is in regards to registration and "open-SIP" 
support. Asterisk and any iteration of it will support it, but Cisco hasn't 
really designed a load compatible with it yet. I can tell you that I haven't 
really found any configuration file generation tools for these files. The 
reason being is that these loads are mainly used for SCCP and SIP Cisco 
systems. There is a well known tutorial on how to "Hack to the CP-7970 to 
trixbox CE located here:

http://www.asterisktutorials.com/cisco-7970-ip-phone/

This may help get you jump started and pointed in the right direction. The only 
problem that may arise is that in the tutorial, the use a specific SIP load 
(8.0.3) which may not be available for the 7961G.

Cory J. Andrews
Director New Market Initiatives

Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
candr...@sayersmedia.com


Have I exceeded your expectations?  Please share your experience with my boss,  
Benjamin P. Sayers, CEO

NOTICE: The information contained in this email and any document attached 
hereto is intended only for the named recipient(s). It is the property of the 
VoIP Supply, LLC and shall not be used, disclosed or reproduced without the 
express written consent of VoIP Supply, LLC. If you are not the intended 
recipient, nor the employee or agent responsible for delivering this message in 
confidence to the intended recipient(s), you are hereby notified that you have 
received this transmittal in error, and any review, dissemination, distribution 
or copying of this transmittal or its attachments is strictly prohibited. If 
you have received this transmittal and/or attachments in error, please notify 
me immediately by reply e-mail or telephone and then delete this message, 
including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 
14225 USA.



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailt

Re: [asterisk-users] Converting Cisco 7961 to SIP

[David Gibbons]

Ahh I see.

In response to your other question about the auto-provisioning of Cisco phones, 
I wrote some scripts that work against an active directory and setup the phones 
automagically. I'll send the link your way if you'd like.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews
Sent: Tuesday, May 26, 2009 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP

Did not mean to infer they don't perform wonderfully with Asterisk.  By "hack" 
I meant that Cisco does not offer any official support for them on Asterisk.

Cory J. Andrews
Director New Market Initiatives

Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
candr...@sayersmedia.com


Have I exceeded your expectations?  Please share your experience with my boss,  
Benjamin P. Sayers, CEO

NOTICE: The information contained in this email and any document attached 
hereto is intended only for the named recipient(s). It is the property of the 
VoIP Supply, LLC and shall not be used, disclosed or reproduced without the 
express written consent of VoIP Supply, LLC. If you are not the intended 
recipient, nor the employee or agent responsible for delivering this message in 
confidence to the intended recipient(s), you are hereby notified that you have 
received this transmittal in error, and any review, dissemination, distribution 
or copying of this transmittal or its attachments is strictly prohibited. If 
you have received this transmittal and/or attachments in error, please notify 
me immediately by reply e-mail or telephone and then delete this message, 
including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 
14225 USA.



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Tuesday, May 26, 2009 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP

Cory,

Precisely what do you mean by 'Anything other than Callmanager will essentially 
be a "hack"'?

I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP 
image. They're not 'hacked', they're set up properly against the Cisco provided 
SIP image and are rock-solid stable. I would pit them against any of the 
cheaper model SIP phones any time, any place, any day.

I've written scripts to do nearly everything that call manager can do without 
paying hundreds of dollars per user for the call manager software. Just about 
the only thing they can't do at the moment is BLF because they require SIP over 
TCP to handle SIP messages about BLF status, something that I'm not willing to 
implement just yet.

In the past, Cisco phones have had a bad rap as not being usable outside of a 
call manager environment. That's just not the case.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cory Andrews
Sent: Tuesday, May 26, 2009 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Converting Cisco 7961 to SIP

Darrin,

The files you are using are consistent with SIP for Cisco Call Manager. 
Anything other than Callmanager will essentially be a "hack". I am not sure how 
proprietary the Avaya system is in regards to registration and "open-SIP" 
support. Asterisk and any iteration of it will support it, but Cisco hasn't 
really designed a load compatible with it yet. I can tell you that I haven't 
really found any configuration file generation tools for these files. The 
reason being is that these loads are mainly used for SCCP and SIP Cisco 
systems. There is a well known tutorial on how to "Hack to the CP-7970 to 
trixbox CE located here:

http://www.asterisktutorials.com/cisco-7970-ip-phone/

This may help get you jump started and pointed in the right direction. The only 
problem that may arise is that in the tutorial, the use a specific SIP load 
(8.0.3) which may not be available for the 7961G.

Cory J. Andrews
Director New Market Initiatives

Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
candr...@sayersmedia.com


Have I exceeded your expectations?  Please share your experience with my boss,  
Benjamin P. Sayers, CEO

NOTICE: The information contained in this email and any document attached 
hereto is intended only for the named recipient(s). It is the property of the 
VoIP Supply, LLC and shall not be used, disclosed or reproduced without the 
express written consent of VoIP Supply, LLC. If you are not the intended 
recipient, nor the employee or agent responsible for delivering this message in 
confidence to the intended

Re: [asterisk-users] Maximum cable length for analog phone from FXS port

[David Gibbons]

I could be wrong but I don't think the cat5 limit of 100 meters applies to any 
analog signaling over that copper. I believe it only applies to Ethernet 
signaling.

-Dave

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, May 26, 2009 10:41 AM
To: bald...@rogg.is; 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Maximum cable length for analog phone from FXS 
port

The best a native cat5 can run is 100 meters.  Unless you like paying your 
telco huge bucks, you should go for some kind of SIP connection to your box.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
asterisk-us...@rogg.is
Sent: Tuesday, May 26, 2009 9:09 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Maximum cable length for analog phone from FXS port

Hello.

I am looking for details of the maximum allowed/usable/effective wire/cable 
length of the connection from a FXS port of Digium analog cards to the analog 
telephone handset.

To clarify my intention, I need to have an analog telephone connection to my 
asterisk box that is 3000 meters (3km) away at least. If you have any details 
of ATA boxes or other similar devices that I could use to do this, I'd 
appreciate your input. It must be able to use a regular analog telephone 
handset on the far end.

I've searched high and low and either I'm not clever enough in using the right 
terms for this or it is rarely documented?

Any details much appreciated.

Thank you!
Baldvin

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Re: [asterisk-users] h extension and channel variables

[Thomas Kenyon]

On 5/26/2009 14:08, Marco Sambo wrote:
> I set a variable CalledID to ${EXTEN} before dial it. So in h extension
> I can use ${CalledID}.
>
Thanks for the response.

In that case if there is an intervening call that is shorter, then the 
$calledID will be wrong.

I found a better approach than using the h, extensions.

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Re: [asterisk-users] Problem running Dahdi

[Mike]

Thanks for taking the time to answer.

I've played with the server a lot in the past few days, and I am not sure
what did it, but for futur reference this is my best guess: I think I had
32-bit code or RPMs installed on a 64-bit machine (specifically: HP-hardware
specific RPMs for hardware monitoring).

Things seemed well on the surface, but weren't  going too great under the
hood.

Fixed now.  Thanks for those who took the time to try and help.

Mike


> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Danny Nicholas
> Sent: Tuesday, May 26, 2009 10:04
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Problem running Dahdi
> 
> It is my experience that /e/i/dahdi doesn't always work correctly
(opensuse
> 11.0).  For whatever reason, it doesn't do the required modprobe to get
the
> dadhi module activated.
> 
> Try doing modprobe wctdm
> Then
> Dahdi_cfg -vv
> 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
> Sent: Monday, May 25, 2009 9:27 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Problem running Dahdi
> 
> I did run make install, probably 3-4 times before I ended up asking that
> question in the mailing list.
> 
> Here is the required output: to the first one, "could not find module
> dahdi".
> 
> To the second, it found dahdi in /lib/modules/2.6.18-128.1.10.el5/dahdi
> 
> As for the other questions:
> 
> What do you do?
> I simply try starting /etc/init.d/dahdi restart
> 
> What would you expect to happen?
> No Red warnings, for one.  I have another system that I configured awhile
> ago, and that starts fine.  I understand I have no hardware loaded, but
all
> modules load with a green OK.
> 
> What actually happens?
> 
> FATAL: Module Dahdi not found
> 
> [snip] all modules listed as not found [/snip]
> 
> Error: missing /dev/dahdi!
> 
> What system is this on?
>  - What versions of dahdi-linux and dahdi-tools?
> Latest as found on asterisk.org, that would be
> DAHDI Linux 2.1.0.4
> DAHDI Tools 2.1.0.2
> 
>  - What distribution? What version?
> CentOS, 5.3.  I tried updating all packages before trying again, same
> result.
> 
>  - What kernel version?
> 2.6.18-128.1.10.el5
> 
> 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> > boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
> > Sent: Monday, May 25, 2009 9:53
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] Problem running Dahdi
> >
> > On Mon, May 25, 2009 at 09:32:07AM -0400, Mike wrote:
> > > Sorry, it seems to have disappeared from my original email!
> > >
> > > FATAL: Module Dahdi not found
> > >
> > > [snip] all modules listed as not found [/snip]
> > >
> > > Error: missing /dev/dahdi!
> >
> > Your description makes me suspect you have not run 'make install' in
> > dahdi-linux.
> >
> > What is the output of:
> >
> >   modinfo dahdi
> >   find /lib/modules -name dahdi
> >
> > --
> >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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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
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[asterisk-users] Maximum cable length for analog phone from FXS port

[asterisk-users]

Hello.

 

I am looking for details of the maximum allowed/usable/effective wire/cable
length of the connection from a FXS port of Digium analog cards to the
analog telephone handset.

 

To clarify my intention, I need to have an analog telephone connection to my
asterisk box that is 3000 meters (3km) away at least. If you have any
details of ATA boxes or other similar devices that I could use to do this,
I'd appreciate your input. It must be able to use a regular analog telephone
handset on the far end.

 

I've searched high and low and either I'm not clever enough in using the
right terms for this or it is rarely documented?

 

Any details much appreciated.

 

Thank you!

Baldvin

 

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[asterisk-users] A problem in playing sound files

[abdelkader]

Hello,

I have 8 DID: 7 from a provider1 and 1 from provider2.

Each time a customer calls one of the DID, the system plays a message.

The problem is that the message is played normally for all the DIDs from the
provider1 and is not played (not heard) for the DID from provider2.

My question is: What can be the cause of this problem.

Thanks.
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Re: [asterisk-users] Converting Cisco 7961 to SIP

[Darrin Henshaw]

Ok, ignore what I said below. I've got it working now, thanks a million for 
this link: 
http://www.greenwireit.com/blog/2009/04/reflash-your-cisco-7940-7941-7960-or-7961-phone-to-sip/.

However, now I'm wondering about the dialplan.xml, can it handle regular 
expressions like 9[2-9]..? Thanks.

Cheers,


Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC
Ignition Support Center | www.ignition.bm
Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288
Atlanta | Bermuda | Cayman | Halifax


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin Henshaw
Sent: Tuesday, May 26, 2009 08:40
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Converting Cisco 7961 to SIP

As part of a project to move a clients Cisco phones to SIP to work with an 
Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk 
setup. Now, I've gotten the firmware files from the site, the latest version is 
8.4 which contains the following files:

apps41.8-4-3-16.sbn
cnu41.8-4-3-16.sbn
cvm41sip.8-4-3-16.sbn
dsp41.8-4-3-16.sbn
jar41sip.8-4-3-16.sbn
SIP41.8-4-4S.loads
term41.default.loads
term61.default.loads

Now I've read over loads of documentation on it, but am getting tripped up. 
Most of what I've seen talks about the older firmware versions usually 7.4. I 
have a feeling I'm still missing a lot of stuff. Anyone have any recent links 
or information?

Also, anyone know of a decent way to generate the config files? I'd hate to 
have to go through all of it manually? Thanks.

Cheers,

Darrin Henshaw

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solely those of the author and do not necessarily represent those of Ignition.  
If you are not the intended recipient of this email and its attachments, you 
must take no action based upon them, nor must you copy or show them to anyone.  
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Re: [asterisk-users] Unable to make outbound calls

[Kal Feher]

Ok I've solved the problem. I do not think it was as switchtype issue after
all as it is now working with a national2 configuration.

I need to sort out some of the changes and I'll post back for reference.
However it appears to be some form of parsing order issue between all the
locations that define dahdi trunk groups. What is odd is that this appears
anecdotally to be different between 1.4.22 and 1.4.24. But I'll confirm and
reply.


On 26/5/09 3:46 PM, "Kal Feher"  wrote:

> My thoughts exactly. I've tried National2, 4ess and now ni1
> ni1 just worked on Asterisk 1.4.22. (failover box I downgraded). So I'm
> swapping back to 1.4.24 to test that now.
> 
> 
> On 26/5/09 3:34 PM, "Danny Nicholas"  wrote:
> 
>> Based on this link -
>> http://www.trixbox.org/forums/vendor-forums-certified/sangoma/a101dx-hangup-
>> cause-code-90-outbound-calls
>> 
>> I'd check my polarity settings in dahdi.conf.  Maybe signaling?
>> 
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher
>> Sent: Tuesday, May 26, 2009 8:22 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Unable to make outbound calls
>> 
>> Sorry. I don't get many opportunities to test this system as its live. Here
>> are the results:
>> 
>>-- Executing [...@dlpn_dialplan1:1] Dial("SIP/19722-b650fb80", "DAHDI/1")
>> in new stack
>> -- Requested transfer capability: 0x00 - SPEECH
>> -- Called 1
>> -- Channel 0/1, span 1 got hangup, cause 90
>>  WARNING[6454]: app_dial.c:765 wait_for_answer: Unable to forward voice or
>> dtmf
>> -- Hungup 'DAHDI/1-1'
>>   == Everyone is busy/congested at this time (1:0/0/1)
>> 
>> 
>> On 20/5/09 3:52 PM, "Danny Nicholas"  wrote:
>> 
>>> This all looks ok.  What happens if you try to access the DAHDI channel
>>> outside of Asterisk control:
>>> In dialplan 
>>> Exten => 9,1,Dial(DAHDI/1)
>>> 
>>> Dial 9
>>> Get dialtone
>>> Dial number
>>> 
>>> 
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com
>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher
>>> Sent: Wednesday, May 20, 2009 2:55 AM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: Re: [asterisk-users] Unable to make outbound calls
>>> 
>>> I attached the show channels in my first post, but removed it to reduce
>> the
>>> length of replies. Here it is again along with show status.
>>> Note that there is only 1 PRI currently attached.
>>> 
>>> geriatrix*CLI> dahdi show status
>>> Description  Alarms IRQbpviol
>>> CRC4  
>>> T2XXP (PCI) Card 0 Span 1OK 0  0
>> 0
>>> T2XXP (PCI) Card 0 Span 2RED0  0
>> 0
>>>
>>> geriatrix*CLI> dahdi show channels
>>>Chan Extension  Context Language   MOH Interpret
>>>  pseudodefaultdefault
>>>   1DID_span_1 default
>>>   2DID_span_1 default
>>>   3DID_span_1 default
>>>   4DID_span_1 default
>>>   5DID_span_1 default
>>>   6DID_span_1 default
>>>   7DID_span_1 default
>>>   8DID_span_1 default
>>>   9DID_span_1 default
>>>  10DID_span_1 default
>>>  11DID_span_1 default
>>>  12DID_span_1 default
>>>  13DID_span_1 default
>>>  14DID_span_1 default
>>>  15DID_span_1 default
>>>  16DID_span_1 default
>>>  17DID_span_1 default
>>>  18DID_span_1 default
>>>  19DID_span_1 default
>>>  20DID_span_1 default
>>>  21DID_span_1 default
>>>  22DID_span_1 default
>>>  23DID_span_1 default
>>>  25DID_span_2 default
>>>  26DID_span_2 default
>>>  27DID_span_2 default
>>>  28DID_span_2 default
>>>  29DID_span_2 default
>>>  30DID_span_2 default
>>>  31DID_span_2 default
>>>  32DID_span_2 default
>>>  33DID_span_2 default
>>>  34DID_span_2 default
>>>  35DID_span_2 default
>>>  36  

Re: [asterisk-users] Problem running Dahdi

[Danny Nicholas]

It is my experience that /e/i/dahdi doesn't always work correctly (opensuse
11.0).  For whatever reason, it doesn't do the required modprobe to get the
dadhi module activated. 

Try doing modprobe wctdm
Then
Dahdi_cfg -vv


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Monday, May 25, 2009 9:27 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Problem running Dahdi

I did run make install, probably 3-4 times before I ended up asking that
question in the mailing list.

Here is the required output: to the first one, "could not find module
dahdi".

To the second, it found dahdi in /lib/modules/2.6.18-128.1.10.el5/dahdi

As for the other questions:

What do you do? 
I simply try starting /etc/init.d/dahdi restart

What would you expect to happen?
No Red warnings, for one.  I have another system that I configured awhile
ago, and that starts fine.  I understand I have no hardware loaded, but all
modules load with a green OK.

What actually happens?

FATAL: Module Dahdi not found

[snip] all modules listed as not found [/snip]

Error: missing /dev/dahdi!

What system is this on?
 - What versions of dahdi-linux and dahdi-tools?
Latest as found on asterisk.org, that would be 
DAHDI Linux 2.1.0.4
DAHDI Tools 2.1.0.2

 - What distribution? What version?
CentOS, 5.3.  I tried updating all packages before trying again, same
result.

 - What kernel version?
2.6.18-128.1.10.el5


> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
> Sent: Monday, May 25, 2009 9:53
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Problem running Dahdi
> 
> On Mon, May 25, 2009 at 09:32:07AM -0400, Mike wrote:
> > Sorry, it seems to have disappeared from my original email!
> >
> > FATAL: Module Dahdi not found
> >
> > [snip] all modules listed as not found [/snip]
> >
> > Error: missing /dev/dahdi!
> 
> Your description makes me suspect you have not run 'make install' in
> dahdi-linux.
> 
> What is the output of:
> 
>   modinfo dahdi
>   find /lib/modules -name dahdi
> 
> --
>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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Re: [asterisk-users] howto store local exchange prefixes ?

[Danny Nicholas]

Now that I've slogged through everyone else's reply and got to the original
post, here's an idea.  You seem to have the dialplan part worked out; why
not do a simple HTML interface to do the Berkley maint using asterisk -rx to
do the CLI reads/pokes? With asterisk -rx you can automate 90+ percent of
CLI functions.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Monday, May 25, 2009 4:29 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] howto store local exchange prefixes ?

The local telco is now going 10 digit dialing even for local (free) 
calls which used to be 7 digit. For a while no problem, everyone will 
continue to dial 7 digits, and I'll add the area code. But pretty soon 
everyone will become used to 10 digits.

There are about 40 3 digit local exchanges. I'd like to store the 
exchanges in a database, and use the dialplan to check them. I can 
figure that out.

I've looked at the Berkeley DB. That works pretty well, if the exchanges 
are all stored. But it looks like the exchanges have to be entered 1 by 
1 from the CLI. And can only be reviewed, corrected, or deleted from the 
CLI. I haven't found any simple frontend for the DB.

I'd also consider sqlite3, but from the sqlite3 .conf.sample, it's only 
for CDR. In any event, I couldn't find a simple frontend. I'd prefer not 
to go into mysql etc for such a simple project.

sean


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Re: [asterisk-users] Unable to make outbound calls

[Kal Feher]

My thoughts exactly. I've tried National2, 4ess and now ni1
ni1 just worked on Asterisk 1.4.22. (failover box I downgraded). So I'm
swapping back to 1.4.24 to test that now.


On 26/5/09 3:34 PM, "Danny Nicholas"  wrote:

> Based on this link -
> http://www.trixbox.org/forums/vendor-forums-certified/sangoma/a101dx-hangup-
> cause-code-90-outbound-calls
> 
> I'd check my polarity settings in dahdi.conf.  Maybe signaling?
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher
> Sent: Tuesday, May 26, 2009 8:22 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Unable to make outbound calls
> 
> Sorry. I don't get many opportunities to test this system as its live. Here
> are the results:
> 
>-- Executing [...@dlpn_dialplan1:1] Dial("SIP/19722-b650fb80", "DAHDI/1")
> in new stack
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called 1
> -- Channel 0/1, span 1 got hangup, cause 90
>  WARNING[6454]: app_dial.c:765 wait_for_answer: Unable to forward voice or
> dtmf
> -- Hungup 'DAHDI/1-1'
>   == Everyone is busy/congested at this time (1:0/0/1)
> 
> 
> On 20/5/09 3:52 PM, "Danny Nicholas"  wrote:
> 
>> This all looks ok.  What happens if you try to access the DAHDI channel
>> outside of Asterisk control:
>> In dialplan 
>> Exten => 9,1,Dial(DAHDI/1)
>> 
>> Dial 9
>> Get dialtone
>> Dial number
>> 
>> 
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher
>> Sent: Wednesday, May 20, 2009 2:55 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Unable to make outbound calls
>> 
>> I attached the show channels in my first post, but removed it to reduce
> the
>> length of replies. Here it is again along with show status.
>> Note that there is only 1 PRI currently attached.
>> 
>> geriatrix*CLI> dahdi show status
>> Description  Alarms IRQbpviol
>> CRC4  
>> T2XXP (PCI) Card 0 Span 1OK 0  0
> 0
>> T2XXP (PCI) Card 0 Span 2RED0  0
> 0
>>
>> geriatrix*CLI> dahdi show channels
>>Chan Extension  Context Language   MOH Interpret
>>  pseudodefaultdefault
>>   1DID_span_1 default
>>   2DID_span_1 default
>>   3DID_span_1 default
>>   4DID_span_1 default
>>   5DID_span_1 default
>>   6DID_span_1 default
>>   7DID_span_1 default
>>   8DID_span_1 default
>>   9DID_span_1 default
>>  10DID_span_1 default
>>  11DID_span_1 default
>>  12DID_span_1 default
>>  13DID_span_1 default
>>  14DID_span_1 default
>>  15DID_span_1 default
>>  16DID_span_1 default
>>  17DID_span_1 default
>>  18DID_span_1 default
>>  19DID_span_1 default
>>  20DID_span_1 default
>>  21DID_span_1 default
>>  22DID_span_1 default
>>  23DID_span_1 default
>>  25DID_span_2 default
>>  26DID_span_2 default
>>  27DID_span_2 default
>>  28DID_span_2 default
>>  29DID_span_2 default
>>  30DID_span_2 default
>>  31DID_span_2 default
>>  32DID_span_2 default
>>  33DID_span_2 default
>>  34DID_span_2 default
>>  35DID_span_2 default
>>  36DID_span_2 default
>>  37DID_span_2 default
>>  38DID_span_2 default
>>  39DID_span_2 default
>>  40DID_span_2 default
>>  41DID_span_2 default
>>  42DID_span_2 default
>>  43DID_span_2 default
>>  44DID_span_2 default
>>  45DID_span_2 default
>>  46DID_span_2   

Re: [asterisk-users] 1.6.0.9 sip.c: "Serious Network Trouble" ??

[Tilghman Lesher]

On Saturday 23 May 2009 11:03:13 sean darcy wrote:
> I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 over this weekend.
>
> I'm getting:
>
> [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit:
> Serious Network Trouble; __sip_xmit returns error for pkt data
> [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit:
> Serious Network Trouble; __sip_xmit returns error for pkt data
> [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit:
> Serious Network Trouble; __sip_xmit returns error for pkt data
> .
>
> What does this mean? What do i do about it?
>
> sip worked fine in 1.4.24.1.

The difference is that 1.6.0 reports network errors, whereas 1.4 did not.

-- 
Tilghman

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Re: [asterisk-users] Asterisk and Data Modem

[Danny Nicholas]

Install nv_faxdetect.  This will make asterisk not attempt to process the
modem call for a specified period of time.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Morgan
Sent: Tuesday, May 26, 2009 5:06 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Asterisk and Data Modem

Hi All,

We have an Asterisk 1.4.21.2 box which uses a 2 port Digium card to bridge
calls, as follows:

ISDN Provider <---> Span 1(pri_cpe) <---> Span 2(pri_net) <> Phone
System 

The company that looks after our internal phone system can no longer dial in
using their data modem in order to maintain the internal phone system.  Is
there any way we can configure our asterisk to allow them to dial in using
their modem?

Regards,

Jon.


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Re: [asterisk-users] Unable to make outbound calls

[Danny Nicholas]

Based on this link -
http://www.trixbox.org/forums/vendor-forums-certified/sangoma/a101dx-hangup-
cause-code-90-outbound-calls

I'd check my polarity settings in dahdi.conf.  Maybe signaling?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher
Sent: Tuesday, May 26, 2009 8:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unable to make outbound calls

Sorry. I don't get many opportunities to test this system as its live. Here
are the results:

   -- Executing [...@dlpn_dialplan1:1] Dial("SIP/19722-b650fb80", "DAHDI/1")
in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called 1
-- Channel 0/1, span 1 got hangup, cause 90
 WARNING[6454]: app_dial.c:765 wait_for_answer: Unable to forward voice or
dtmf
-- Hungup 'DAHDI/1-1'
  == Everyone is busy/congested at this time (1:0/0/1)


On 20/5/09 3:52 PM, "Danny Nicholas"  wrote:

> This all looks ok.  What happens if you try to access the DAHDI channel
> outside of Asterisk control:
> In dialplan 
> Exten => 9,1,Dial(DAHDI/1)
> 
> Dial 9
> Get dialtone
> Dial number
> 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kal Feher
> Sent: Wednesday, May 20, 2009 2:55 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Unable to make outbound calls
> 
> I attached the show channels in my first post, but removed it to reduce
the
> length of replies. Here it is again along with show status.
> Note that there is only 1 PRI currently attached.
> 
> geriatrix*CLI> dahdi show status
> Description  Alarms IRQbpviol
> CRC4  
> T2XXP (PCI) Card 0 Span 1OK 0  0
0
> T2XXP (PCI) Card 0 Span 2RED0  0
0
>
> geriatrix*CLI> dahdi show channels
>Chan Extension  Context Language   MOH Interpret
>  pseudodefaultdefault
>   1DID_span_1 default
>   2DID_span_1 default
>   3DID_span_1 default
>   4DID_span_1 default
>   5DID_span_1 default
>   6DID_span_1 default
>   7DID_span_1 default
>   8DID_span_1 default
>   9DID_span_1 default
>  10DID_span_1 default
>  11DID_span_1 default
>  12DID_span_1 default
>  13DID_span_1 default
>  14DID_span_1 default
>  15DID_span_1 default
>  16DID_span_1 default
>  17DID_span_1 default
>  18DID_span_1 default
>  19DID_span_1 default
>  20DID_span_1 default
>  21DID_span_1 default
>  22DID_span_1 default
>  23DID_span_1 default
>  25DID_span_2 default
>  26DID_span_2 default
>  27DID_span_2 default
>  28DID_span_2 default
>  29DID_span_2 default
>  30DID_span_2 default
>  31DID_span_2 default
>  32DID_span_2 default
>  33DID_span_2 default
>  34DID_span_2 default
>  35DID_span_2 default
>  36DID_span_2 default
>  37DID_span_2 default
>  38DID_span_2 default
>  39DID_span_2 default
>  40DID_span_2 default
>  41DID_span_2 default
>  42DID_span_2 default
>  43DID_span_2 default
>  44DID_span_2 default
>  45DID_span_2 default
>  46DID_span_2 default
>  47DID_span_2 default
> 
> 
> 
> On 19/5/09 6:31 PM, "Danny Nicholas"  wrote:
> 
>> Please post your CLI output from dahdi show status and dahdi show
> channels.
>> 
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of

Re: [asterisk-users] Converting Cisco 7961 to SIP

[Cory Andrews]

Darrin,

The files you are using are consistent with SIP for Cisco Call Manager. 
Anything other than Callmanager will essentially be a "hack". I am not sure how 
proprietary the Avaya system is in regards to registration and "open-SIP" 
support. Asterisk and any iteration of it will support it, but Cisco hasn't 
really designed a load compatible with it yet. I can tell you that I haven't 
really found any configuration file generation tools for these files. The 
reason being is that these loads are mainly used for SCCP and SIP Cisco 
systems. There is a well known tutorial on how to "Hack to the CP-7970 to 
trixbox CE located here:

http://www.asterisktutorials.com/cisco-7970-ip-phone/ 

This may help get you jump started and pointed in the right direction. The only 
problem that may arise is that in the tutorial, the use a specific SIP load 
(8.0.3) which may not be available for the 7961G.

Cory J. Andrews
Director New Market Initiatives
 
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
candr...@sayersmedia.com


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-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin Henshaw
Sent: Tuesday, May 26, 2009 7:40 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Converting Cisco 7961 to SIP

As part of a project to move a clients Cisco phones to SIP to work with an 
Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk 
setup. Now, I've gotten the firmware files from the site, the latest version is 
8.4 which contains the following files:

apps41.8-4-3-16.sbn
cnu41.8-4-3-16.sbn
cvm41sip.8-4-3-16.sbn
dsp41.8-4-3-16.sbn
jar41sip.8-4-3-16.sbn
SIP41.8-4-4S.loads
term41.default.loads
term61.default.loads

Now I've read over loads of documentation on it, but am getting tripped up. 
Most of what I've seen talks about the older firmware versions usually 7.4. I 
have a feeling I'm still missing a lot of stuff. Anyone have any recent links 
or information?

Also, anyone know of a decent way to generate the config files? I'd hate to 
have to go through all of it manually? Thanks.

Cheers,

Darrin Henshaw

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