RE: [asterisk-users] Microsoft launches first PABX

[Anselm Martin Hoffmeister]

Am Donnerstag, den 22.03.2007, 22:17 -0700 schrieb shadowym:
> As far as I can tell, the phone system does not run on a Desktop/Server OS
> on a standard PC.  Just the config clients run on the desktop.
> 
> Then again they are using Dlink as one of the 3 manufacturers of the Phone
> Server so I wouldn't expect commercial grade.

Let us wait for the actual implementation before ranting too much - I
have certain, not the best, expectations of any new MS products, but
they might one day be proven untrue.

I have to admin though that the combination of D-Link and MS does not
exactly stand for highest quality, reliable, bugfree products. But they
might once produce a great product.

The most interesting of all this for me is which protocols they will
use, e.g. wether they will talk SIP, or rather "MS OpenIAX" or "Skype2.0
protocol", or something completely new and not just slightly
uncompatible.

Let us see the facts: Telephone systems with more than a handful
telephones and more than just the ability to call (be it voicemail,
conferencing, queues, agents...) are complicated, and in most cases need
to be tailored to the customers' needs. As long as the "customer" is not
an IT-ish company, they will hopefully understand that getting all the
knowledge about this internally costs work hours (and thus, money) the
same - and experience is something that can not be learned in a few
hours of document study and point-and-clicking. High-quality solutions
need professional hands, pals, possibly yours.

This will by no means be the death of the technical consulting around
telephone PABXs.

BR
Anselm

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Re: [asterisk-users] Microsoft launches first PABX

[Christopher Chan]

Anselm Martin Hoffmeister wrote:

Am Donnerstag, den 22.03.2007, 22:17 -0700 schrieb shadowym:

As far as I can tell, the phone system does not run on a Desktop/Server OS
on a standard PC.  Just the config clients run on the desktop.

Then again they are using Dlink as one of the 3 manufacturers of the Phone
Server so I wouldn't expect commercial grade.


Let us wait for the actual implementation before ranting too much - I
have certain, not the best, expectations of any new MS products, but
they might one day be proven untrue.


I have yet to see anything that does turn into a mess from M$.



I have to admin though that the combination of D-Link and MS does not
exactly stand for highest quality, reliable, bugfree products. But they
might once produce a great product.


On the Mac OS X side of things...



The most interesting of all this for me is which protocols they will
use, e.g. wether they will talk SIP, or rather "MS OpenIAX" or "Skype2.0
protocol", or something completely new and not just slightly
uncompatible.


Given M$ history, it will probably be another embrace, extend and 
extinguish the old.




Let us see the facts: Telephone systems with more than a handful
telephones and more than just the ability to call (be it voicemail,
conferencing, queues, agents...) are complicated, and in most cases need
to be tailored to the customers' needs. As long as the "customer" is not
an IT-ish company, they will hopefully understand that getting all the
knowledge about this internally costs work hours (and thus, money) the
same - and experience is something that can not be learned in a few
hours of document study and point-and-clicking. High-quality solutions
need professional hands, pals, possibly yours.

This will by no means be the death of the technical consulting around
telephone PABXs.


Er...is not this what asterisk is about? telephone PABX guys sniff at 
computer guys moving in their space.

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Re: [asterisk-users] Re: About Pickup Grandstream

[Tzafrir Cohen]

On Thu, Mar 22, 2007 at 09:42:56PM -0500, Lacy Moore - Aspendora wrote:
> On 3/22/07, Lukas <[EMAIL PROTECTED]> wrote:
> >First of all, thanks a lot.
> >
> >Believe me that if I'm writing down here it's due that i cannot find the
> >problem out. Maybe it's a bug, but either of IAX or mISDN couldn't get
> >pickup calls.
> >Could be the GrandStream?
> >
> Forgive my lack of knowledge on this, but does mISDN use BRIstuff?

No.

-- 
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+972-50-7952406   mailto:[EMAIL PROTECTED]   
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Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1

[Richard Klingler]

Hmm..interestingly no one answered if chan_skinny works with 7970G
on * 1.4.x (o;

I know that CIsco phones are bad with NAT and SIP...old story (o;
THat's why I use local Cisco phones with SIP and local * which then
connects to outside * vis IAX...


cheers
rick


Hermann Wecke schrieb:

Richard Klingler wrote:

Has any1 got their 7970 to work with * 1.4.x ?


Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2
without problems (Asterisk 1.2.16). Just remember that 7970 only will
register if your Asterisk is at the same network - no NAT between them - 
check http://preview.tinyurl.com/345fmj

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Re: [asterisk-users] Re: wct4xxp problem

[Steve Totaro]

Tomislav Parcina wrote:

Steve Totaro wrote:
All providers will do this if a line is in alarm for a couple days.  
The FIRST thing to do is call them and make sure it is turned up.  
Once and only once was I called prior to the CO/NOC turning down a 
circuit that had been in alarm for two days straight.


Yes, and before they turn down the line they should call you and check 
what's wrong. What if that was your backup line and you haven't 
noticed that it's in red alarm? Then one day when you need that line 
it doesn't work.





They "should" but as I said, I have only had the call once out of 
several dozens of instances.  What the telco "should" do is more often 
not what is done. 

If any of your lines are in red alarm long enough to be turned down, and 
you have not noticed (including a backup link) then YOU need to do a 
better job.  Is it the job of the telco to monitor your CPE and alert 
you to your own problems?  They call it a Demarc for a reason and what 
is beyond it is your responsibility. 

I would suggest using some sort of monitoring system if you are too lazy 
or just don't have the time to log into a few boxes and check the 
status.  I take a few minutes in the morning to look around the data 
center for LEDs, log into a few boxes to check for status, and glance 
over logs. 

At any rate, a simple call and they turn it back up. 


Thanks,
Steve
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Re: [asterisk-users] Outbound SIP call from asterisk extension

[Jaswinder Singh]

[outgoing]
exten => 1234,1,Dial(SIP/[EMAIL PROTECTED])


Whats the dialplan number to ring to userA on server (ONDO) ?
if u know that try
exten => 1234,1,Dial(SIP/[EMAIL PROTECTED]
)

since ur sip.conf has [*192.xxx.xxx.xxx*-out].
i am not sure why you use Sip/test to call to userA . Is that how server
ONDO is configured ?
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Re: [asterisk-users] Microsoft launches first PABX

[Steve Totaro]

Christopher Chan wrote:

Anselm Martin Hoffmeister wrote:

Am Donnerstag, den 22.03.2007, 22:17 -0700 schrieb shadowym:
As far as I can tell, the phone system does not run on a 
Desktop/Server OS

on a standard PC.  Just the config clients run on the desktop.

Then again they are using Dlink as one of the 3 manufacturers of the 
Phone

Server so I wouldn't expect commercial grade.


Let us wait for the actual implementation before ranting too much - I
have certain, not the best, expectations of any new MS products, but
they might one day be proven untrue.


I have yet to see anything that does turn into a mess from M$.



I have to admin though that the combination of D-Link and MS does not
exactly stand for highest quality, reliable, bugfree products. But they
might once produce a great product.


On the Mac OS X side of things...



The most interesting of all this for me is which protocols they will
use, e.g. wether they will talk SIP, or rather "MS OpenIAX" or "Skype2.0
protocol", or something completely new and not just slightly
uncompatible.


Given M$ history, it will probably be another embrace, extend and 
extinguish the old.




Let us see the facts: Telephone systems with more than a handful
telephones and more than just the ability to call (be it voicemail,
conferencing, queues, agents...) are complicated, and in most cases need
to be tailored to the customers' needs. As long as the "customer" is not
an IT-ish company, they will hopefully understand that getting all the
knowledge about this internally costs work hours (and thus, money) the
same - and experience is something that can not be learned in a few
hours of document study and point-and-clicking. High-quality solutions
need professional hands, pals, possibly yours.

This will by no means be the death of the technical consulting around
telephone PABXs.


Er...is not this what asterisk is about? telephone PABX guys sniff at 
computer guys moving in their space.




By far the easiest "turnkey" system I have dealt with is the 3com 
V3000/NBX.  No IP addresses to worry about except for the NCP (typically 
the phones are MAC or layer two devices by default).  You just plug in a 
phone and it downloads the latest firmware from the NBX and gets 
assigned the next extension that is open.  The GUI is very simple.  A 
V3000 is about the price of a decent Asterisk box (off the shelf that 
is, it comes in a 1u chassis, four FXOs, one FXS, MOH minijack port for 
about $3,000).  The phones are great and have advanced button mappings 
that are done in the PBX.


That is for an entry level four FXO system.  If you add a T1 or need 
more analog FXS ports, you need to buy an NBX 100 chassis to populate 
with cards, the chassis is about $500 but the T1 boards are in the 
$3,000 - $5,000 range and only have one port.


It has been a while since I have was certified and worked on these 
systems, so pricing may have changed. 

I do not see MS coming out with a product that can beat the 3com system 
for ease of installation and 3com NBX (started by students from MIT who 
formed a company called "NBX Corporation" that was purchased by 3com 
~1997 or 1998, so they have had long enough to iron out bugs and 
implement advanced feature sets.  3com is large enough to resist MS and 
I seriously doubt they can "extinguish" asterisk or Cisco.


Obviously, Asterisk may be in the same "Market Space" of all other IP 
PBXs, but in reality it is apples to oranges.  Being able to explain and 
sell this concept to the customer is paramount in making the sale and 
often times, it makes more sense for 3com system or possibly an MS system.


Thanks,
Steve

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RE: [asterisk-users] Re: About Pickup Grandstream

[LKS GMAIL]

Excuse me. I couldn't understand you.

Saludos, Lukassky.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Tzafrir Cohen
Enviado el: viernes, 23 de marzo de 2007 11:04
Para: asterisk-users@lists.digium.com
Asunto: Re: [asterisk-users] Re: About Pickup Grandstream

On Thu, Mar 22, 2007 at 09:42:56PM -0500, Lacy Moore - Aspendora wrote:
> On 3/22/07, Lukas <[EMAIL PROTECTED]> wrote:
> >First of all, thanks a lot.
> >
> >Believe me that if I'm writing down here it's due that i cannot find the
> >problem out. Maybe it's a bug, but either of IAX or mISDN couldn't get
> >pickup calls.
> >Could be the GrandStream?
> >
> Forgive my lack of knowledge on this, but does mISDN use BRIstuff?

No.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] minimal asterisk for iax2 bridge

[Francisco Pérez Botella]

Hello:

I'm building asterisk for a bcm96348GW board, wich has a usb capable device to 
timming ztdummy, kernel is 2.6.8.1.

This board will be serve as a prototype for an IAX2 trunk "bridge" in the form

SIP/IAX2<--->IAX2trunk<--Inet-->IAX2trunk<-->LucentCS/SIP<-->SS7/POTS

The parameters are known.. in the sense that only g729.a will be used, the two 
IAX2trunks are not going to do transcode at all indeed, just repacking and 
protocol conversion.

so in this scenario, do I need to ever compile and load codecs modules in the 
firmware built ?? space is limited in the board.

Do i need to compile the "format" modules too ??

must I put allow g729 in the confs?

on the other side I need to generate CDRs, one in the board for " sale" rates 
in the callshop an other in the Lucent compact switch for "cost" rates, 
question is. I need to answer in the board before dial iax2 because of 
sendtext parameters, then in the dial iax2.. Do I need to reset cdrs in order 
to do not count time between "answer" and far end answer ??

At the other side of the bridge the machine just dial sip reintroducing call 
name (or number) for authenticate in front of Lucent CS, I think I have not 
to "answer" incomming calls to iax2 bridge, because it will generate billable 
time in the board cdr engine. Am I wrong ??

Please send any comments, thanks

-- 
Francisco J. Pérez Botella
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[asterisk-users] ChanSpy and MeetMe

[GDrayer]

I have been successful using ChanSpy on a standard Dial call but when
attempting to ChanSpy on an incoming call that has been added to a
MeetMe conference (attempting to coach an agent that is speaking to a
conference of callers) it seems to fail to connect to the channel.
Here's the console dump:

-- Accepting call from '2154182700' to '3399' on channel 0/18, span
4
-- Executing [EMAIL PROTECTED]:1] Answer("Zap/90-1", "") in new
stack
-- Executing [EMAIL PROTECTED]:2] Read("Zap/90-1", "GOTDTMF|demo
instruct|1||1|1") in new stack
-- Accepting a maximum of 1 digits.
-- Playing 'demo-instruct' (language 'en')
-- User entered '5'
-- Executing [EMAIL PROTECTED]:3] GotoIf("Zap/90-1", "5?9") in new
stack
-- Goto (from-internal,3399,9)
-- Executing [EMAIL PROTECTED]:9] AGI("Zap/90-1",
"simpleconf.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/simpleconf.agi
-- Playing 'digits/5' (language 'en')
-- AGI Script Executing Application: (CHANSPY) Options: (Zap/73|wbq)

I verified Zap/73 is the correct channel of the caller currently in the
conference I am attempting to ChanSpy on.  Has anyone done this before?
I apologize in advance if my question lacks the necessary information,
I'm new to Asterisk.

-George
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Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1

[Julio Arruda]

Richard Klingler wrote:

Hmm..interestingly no one answered if chan_skinny works with 7970G
on * 1.4.x (o;

I know that CIsco phones are bad with NAT and SIP...old story (o;
THat's why I use local Cisco phones with SIP and local * which then
connects to outside * vis IAX...



I've a 7912G running with 1.4.x and chan_skinny, and seems to be working 
 just fine (better than 1.2 anyway, the 7912G is not the 'heavy usage' 
phone at home, but still..)


I tried twice to acquire the proper license to upgrade the 7912G to SIP, 
but the order got 'dropped' by the reseller after 2 weeks of 'shipping' 
:-), since 1.4 seems to be handling it just fine, I've moved this to the 
lower priority TODO list.




Hermann Wecke schrieb:

Richard Klingler wrote:

Has any1 got their 7970 to work with * 1.4.x ?


Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2
without problems (Asterisk 1.2.16). Just remember that 7970 only will
register if your Asterisk is at the same network - no NAT between them 
- check http://preview.tinyurl.com/345fmj

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Re: [asterisk-users] Microsoft launches first PABX

[Jon Pounder]

This will by no means be the death of the technical consulting around
telephone PABXs.


Er...is not this what asterisk is about? telephone PABX guys sniff at 
computer guys moving in their space.


at least in the asterisk case its all about open standards, interoperability,
reliability, scalability etc., cost savings does usually factor in as well but
its not the primary driver for going with asterisk above another solution in a
lot of cases.






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Jon Pounder

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


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
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www.opayc.com


This message was sent using IMP, the Internet Messaging Program.


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Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1

[Pavel Jezek]

If you have 7970 right configured to point to asterisk server, you 
should be able to see some skinny debug on console, or look what report 
"skinny show devices"
I haven't any 7970, so can't help so much, I'm using only 7920 wifi 
phone with chan_skinny and 1.4trunk, it's usable, basic functionality is 
working, but don't expect too much,
btw, if you have money to buy this highend phone with proprietary 
signaling, why don't connect to callmanager?
asterisk will never support all features available in proprierary system 
as good as "original" ;-)

PJ





Richard Klingler wrote:

Hmm..interestingly no one answered if chan_skinny works with 7970G
on * 1.4.x (o;

I know that CIsco phones are bad with NAT and SIP...old story (o;
THat's why I use local Cisco phones with SIP and local * which then
connects to outside * vis IAX...


cheers
rick


Hermann Wecke schrieb:

Richard Klingler wrote:

Has any1 got their 7970 to work with * 1.4.x ?


Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2
without problems (Asterisk 1.2.16). Just remember that 7970 only will
register if your Asterisk is at the same network - no NAT between 
them - check http://preview.tinyurl.com/345fmj

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[asterisk-users] No Audio when integrating openSER and Asterisk , in NAT

[raviprakash sunkara]

Hello Users

openSER is sip proxy and registrar ,
Asterisk is as PBX, Conference and Voicemail servers,
openSER and Asterisk are  in the Same N/w
Where As the UAC are in Behind the NAT,
When Astetrisk is not integrated ,  UAC are in Behind the NAT is working,

openSER is 192.168.2.5
Asterisk is 192.168.2.6

I'm just use rewritehost to asterisk server,

UAC > openSER  -  - - -> Asterisk (voicemail sserver ).


Please help me

--
Thanks and Regards
Ravi Prakash Sunkara
[EMAIL PROTECTED]
M:+91 9985077535
www.hyperion-tech.com
Client and Parent company :- www.august-networks.com
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Re: [asterisk-users] polycom random reboots

[Louis-David Mitterrand]

On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote:
> Yes, I recently saw this with a 501, in my case the network drop was
> the problem. If you have a good tester then run it on the connection.
> I had another drop near by and just swicthed to it.

Was that phone using POE ?
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Re: [asterisk-users] Microsoft launches first PABX

[Christopher Chan]

Obviously, Asterisk may be in the same "Market Space" of all other IP 
PBXs, but in reality it is apples to oranges.  Being able to explain and 
sell this concept to the customer is paramount in making the sale and 
often times, it makes more sense for 3com system or possibly an MS system.



Surely asterisk is in more than the ip PBX 'market space'! But yes, if 
3com or MS can take away the 'sorcery' of a PBX, it will probably make 
sense for some.

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Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1

[Richard Klingler]

I was able to register to * 1.4.1 via skinny...and it showed up
on the lines and devices show output..

On the phone, however, no lines were displayed nor could it
phone out or receive any calls...


Anyone able to share some snippets of their skinny.conf?
I just used the examples and modified the MAC line and
extension line config...but seems something else is
missing...


cheers
rick



Pavel Jezek schrieb:
If you have 7970 right configured to point to asterisk server, you 
should be able to see some skinny debug on console, or look what report 
"skinny show devices"
I haven't any 7970, so can't help so much, I'm using only 7920 wifi 
phone with chan_skinny and 1.4trunk, it's usable, basic functionality is 
working, but don't expect too much,
btw, if you have money to buy this highend phone with proprietary 
signaling, why don't connect to callmanager?
asterisk will never support all features available in proprierary system 
as good as "original" ;-)

PJ





Richard Klingler wrote:

Hmm..interestingly no one answered if chan_skinny works with 7970G
on * 1.4.x (o;

I know that CIsco phones are bad with NAT and SIP...old story (o;
THat's why I use local Cisco phones with SIP and local * which then
connects to outside * vis IAX...


cheers
rick


Hermann Wecke schrieb:

Richard Klingler wrote:

Has any1 got their 7970 to work with * 1.4.x ?


Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2
without problems (Asterisk 1.2.16). Just remember that 7970 only will
register if your Asterisk is at the same network - no NAT between 
them - check http://preview.tinyurl.com/345fmj

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Re: [asterisk-users] Microsoft launches first PABX

[Christopher Chan]

Jon Pounder wrote:




This will by no means be the death of the technical consulting around
telephone PABXs.


Er...is not this what asterisk is about? telephone PABX guys sniff at 
computer guys moving in their space.


at least in the asterisk case its all about open standards, 
interoperability,
reliability, scalability etc., cost savings does usually factor in as 
well but
its not the primary driver for going with asterisk above another 
solution in a

lot of cases.


:). When the office location is changed, I am going to try and rip out 
the arcane Nitsuko KTS and replace it with asterisk + VoIP phones and 
add voice mail and what not :D.

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Re: [asterisk-users] Dial(Local/[EMAIL PROTECTED])?

[Rizwan Hisham]

OK, but y would i want to use it. i mean y not use goto and y this? and what
dialout files are you talking about?

On 3/20/07, Marco Mouta <[EMAIL PROTECTED]> wrote:


Hi,

This is a "tool" that allows you at any time and any place of your
Dialplan or Dialout Call file to dial a specific extension at a specific
context, even if you are not currently in the specific context.

example:

you are at [from-internal] context and you can say:

[from-internal]
exten=> 1234,1,Dial(Local/[EMAIL PROTECTED]) ; Dialing in VIP services,
another context, without a GOTO or something else...
exten=> 1234,2,hangup

This is just a simple example, pls take a look on dialout call files and
for sure you will notice how useful this "local" could be.

Hope it helps,

Best regards,
MoutaPT


On 3/19/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote:
>
> Rizwan Hisham wrote:
>
> > I dont understand the syntax of the dial application when used like
> this:
> >
> > Dial(Local/[EMAIL PROTECTED])
> >
> > i want to know what is this "Local" doing instead of Tech like SIP,
> IAX,
> > H323?
>
> SIP/200 would dial a device (the SIP user 200) whereas
> Local/200 dials the extension 200 in your dialplan (in
> context longdistance).
>
>
> Regards,
>   Philipp
>
> --
> amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
>  Let's use IT to solve problems and not to create new ones.
>Asterisk? -> http://www.das-asterisk-buch.de
>
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Re: [asterisk-users] Re: 302 Moved temporarely

[Eric \"ManxPower\" Wieling]

Tomislav Parcina wrote:

Eric "ManxPower" Wieling wrote:
I believe that if Asterisk receives a 302 back from a device when 
sending a call to it (i.e. the SIP device has Call Forwarding set) 
Asterisk will do the right thing.  This may only work for redirects to 
locations on the same Asterisk server.  This is NOT the case with 
registrations, only call setup.


Well, this happens when I try to make outgoing call through Voipbuster 
SIP provider. I register fine, but sometimes when I try to call someone 
I receive that message (probably they redirect me to some server with 
lower usage) and I can't establish phone call.


But, as far as I have understand you, this should work. Right?


Yes.  My only experience with 302s are when I enable call forwarding on 
a Polycom or Cisco phone.  The phone sends Asterisk a 302 and Asterisk 
then uses chan_local to process the call thru the dialplan as though the 
call was dialed locally.  I vaguely recall Olle adding the 302 support 
several years ago and mentioning that there were security implications 
with supporting 302s to destinations outside of Asterisk.  look>  Look at this in sip.conf.sample:


;promiscredir = no  ; If yes, allows 302 or REDIR to 
non-local SIP address
; Note that promiscredir when redirects 
are made to the
; local system will cause loops since 
SIP is incapable

; of performing a "hairpin" call.
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Re: [asterisk-users] Microsoft launches first PABX

[Jon Pounder]

Quoting Christopher Chan <[EMAIL PROTECTED]>:



Obviously, Asterisk may be in the same "Market Space" of all other 
IP PBXs, but in reality it is apples to oranges.  Being able to 
explain and sell this concept to the customer is paramount in making 
the sale and often times, it makes more sense for 3com system or 
possibly an MS system.



Surely asterisk is in more than the ip PBX 'market space'! But yes, 
if 3com or MS can take away the 'sorcery' of a PBX, it will probably 
make sense for some.


I don't think anything these days should be considered sorcery - one 
quick stop

at google is usually enough to explain the operating principles behind any
system if the end user cares to know. Its more that naive users have this
misplaced trust for M$ even though they demonstrate over and over again they
are not trustworthy either for marketing techniques or quality.

The real acceptance push should be for linux on the desktop, and then it would
not seem so strange that the same operating system runs the phone switch,
coffee pot, router, etc., it would be oh, that's what it is, I know about that
since I use it every day at work.






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Jon Pounder

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


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


This message was sent using IMP, the Internet Messaging Program.


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[asterisk-users] AsteriskNow Beta 4 with T1 Cards?

[Brian K. Alexander, Jr. \(Vision Point Systems\)]

I would like to use AsteriskNow with a T1 PRI. I know that AsteriskNow will
not auto-detect the digital cards but is there a way to manually configure
AsteriskNow so it will be able to see and use the trunks? 

 

Thanks,

-Brian

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Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1

[Pavel Jezek]

my simple, but working config for 7920...

Dial(Skinny/[EMAIL PROTECTED])


[general]
bindaddr=193.179.38.20; Address to bind to
bindport=2000   ; Port to bind to, default tcp/2000
dateformat=D-M-Y; M,D,Y in any order (5 chars max)
keepalive=30
disallow=all
allow=alaw

[PJ]
device=SEPxxx
linelabel="xxx"
context = xxx
nat=1
callwaiting=1
transfer=1
threewaycalling=1
line => 324




Richard Klingler wrote:

I was able to register to * 1.4.1 via skinny...and it showed up
on the lines and devices show output..

On the phone, however, no lines were displayed nor could it
phone out or receive any calls...


Anyone able to share some snippets of their skinny.conf?
I just used the examples and modified the MAC line and
extension line config...but seems something else is
missing...


cheers
rick



Pavel Jezek schrieb:
If you have 7970 right configured to point to asterisk server, you 
should be able to see some skinny debug on console, or look what 
report "skinny show devices"
I haven't any 7970, so can't help so much, I'm using only 7920 wifi 
phone with chan_skinny and 1.4trunk, it's usable, basic functionality 
is working, but don't expect too much,
btw, if you have money to buy this highend phone with proprietary 
signaling, why don't connect to callmanager?
asterisk will never support all features available in proprierary 
system as good as "original" ;-)

PJ





Richard Klingler wrote:

Hmm..interestingly no one answered if chan_skinny works with 7970G
on * 1.4.x (o;

I know that CIsco phones are bad with NAT and SIP...old story (o;
THat's why I use local Cisco phones with SIP and local * which then
connects to outside * vis IAX...


cheers
rick


Hermann Wecke schrieb:

Richard Klingler wrote:

Has any1 got their 7970 to work with * 1.4.x ?


Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2
without problems (Asterisk 1.2.16). Just remember that 7970 only will
register if your Asterisk is at the same network - no NAT between 
them - check http://preview.tinyurl.com/345fmj

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.



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RE: [asterisk-users] AsteriskNow Beta 4 with T1 Cards?

[Brian K. Alexander, Jr. \(Vision Point Systems\)]

Thanks for the quick response but getting the cards to work with Asterisk is
not the problem - what I want is for them to show up in asterisk-gui.

For example, I have a little test system sitting here. It has Fedora6,
Zaptel/Libpri/Asterisk 1.4 and asterisk-gui. It has an analog card installed
but the gui can not detect the analog ports. So I can not use those ports
from asterisk-gui. I can manually configure them just fine and they work -
but I can not access them in the asterisk-gui. I know this is not a digital
card but it is the same problem - asterisk-gui can not detect the card so I
need to somehow force the card into the configuration.

Is there a way to manually configure the system so that ports that the gui
normally could not detect would be available?

Thanks again,
-Brian

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: Friday, March 23, 2007 10:45 AM
To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] AsteriskNow Beta 4 with T1 Cards?

brian,
try these links:

http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf
http://www.voip-info.org/wiki/view/Asterisk+PRI
daveC


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[asterisk-users] SRTP testers needed

[marek cervenka]

please look at
http://www.voip-info.org/wiki/view/Asterisk+SRTP

and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle, 
...)


---
Marek Cervenka
Centrum Vypocetni Techniky
CVT - http://cvt.fpf.slu.cz
FPF SLU OPAVA   - http://www.fpf.slu.cz
LCNA- http://lcna.slu.cz
===

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Re: [asterisk-users] How to access Voicemail Password in Asteriskwithout using V

[Julian Lyndon-Smith]

Ah. I was wanting to test and try this cool tool.

I was trying to download the minivoicemail apps by themselves so that I 
could patch my patched 1.4, but see that it is a whole branch in it's 
own right ;)


Does the Minivm require other changes in the 1.4 tree, or could I just 
download the minivm app itself, rather than the whole branch ?


TIA

Julian.

Olle E Johansson wrote:


11 mar 2007 kl. 10.04 skrev Julian Lyndon-Smith:


Olle,

A couple of questions:

A) If it works for 1.2, I presume that it would work for 1.4 ?

There is a branch called minivoicemail-1.4.

Modules for 1.2 doesn't run without changes in 1.4, since we changed the
module loader interface.

When this is tested, it will be merged to trunk.

B) We currently use realtime for voicemail. Is there a realtime engine 
for Minivoicemail ?
Minivoicemail uses the realtime engine. It's covered in the 
documentation, like

many other things :-)

Oops, just realized while checking the accuracy of that cocky statement 
that Minivm
was using the same realtime handle as voicemail(). Not god. It's now 
"minivm".


Thanks for pointing me in that direction.

/O



Julian

Olle E Johansson wrote:

Just to tease users to test Mini-Voicemail:
*CLI> show function MINIVMACCOUNT
  -= Info about function 'MINIVMACCOUNT' =-
[Syntax]
MINIVMACCOUNT(:item)
[Synopsis]
Gets MiniVoicemail account information
[Description]
Valid items are:
- path   Path to account mailbox (if account exists, 
otherwise temporary mailbox)

- hasaccount 1 if static Minivm account exists, 0 otherwise
- fullname   Full name of account owner
- email  Email address used for account
- etemplate  E-mail template for account (default template if 
none is configured)
- ptemplate  Pager template for account (default template if none 
is configured)

- accountcodeAccount code for voicemail account
- pincodePin code for voicemail account
- timezone   Time zone for voicemail account
- language   Language for voicemail account
-  Channel variable value (set in 
configuration for account)

-
Please test Mini-Voicemail for Asterisk 1.2!
http://www.voip-forum.com for more information.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk SIP Masterclass, Stockholm, May 7-11 - Register now!
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* Olle E Johansson - [EMAIL PROTECTED]
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden



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[asterisk-users] no incoming dad with mISDN 1.1.1 and asterisk?

[Louis-David Mitterrand]

Hello,

After upgrading my kernel to mISDN-1.1.1 while keeping asterisk-1.2.16 I 
no longer match any extension. Apparently the "dad" is empty. However I 
can see the number just before it (146472130):

P[ 4] I IND :SETUP oad:!?145201798p
¡146472130 dad:
¡146472130 pid:2 state:none
P[ 4] EXPORT_PID: pid:2
Mar 23 09:35:28 WARNING[6725]: chan_misdn.c:4750 chan_misdn_log: 
Extension can never match, so disconnecting
P[ 4] I SEND:RELEASE oad:!?145201798p
¡146472130 dad:
¡146472130 pid:2
P[ 4]  --> bc_state:BCHAN_CLEANED
P[ 4] I IND :RELEASE_COMPLETE oad: dad: pid:2 state:EXTCANTMATCH
P[ 4] hangup_chan
P[ 4] -> hangup
P[ 4] * IND : HANGUPpid:2 ctx:default dad:
¡146472130 oad:!?145201798p
¡146472130 State:EXTCANTMATCH
P[ 4]  --> cause:2
P[ 4]  --> out_cause:2
P[ 4]  --> state:EXTCANTMATCH
P[ 4] Channel: mISDN/4-u0 hanguped new state:CLEANING
P[ 4] release_chan: bc with l3id: 40001


With mISDN-1.0.4 and the same asterisk it works fine:

P[ 4] I IND :SETUP oad:145201798 dad:146472130 pid:2 state:none
P[ 4] EXPORT_PID: pid:2
P[ 4] I SEND:PROCEEDING oad:0145201798 dad:0146472130 pid:2
P[ 4]  --> bc_state:BCHAN_CLEANED
-- Executing Goto("mISDN/4-1", "2130|1") in new stack
-- Goto (default,2130,1)
-- Executing NoOp("mISDN/4-1", "") in new stack
-- Executing Macro("mISDN/4-1", "queue") in new stack
-- Executing NoOp("mISDN/4-1", "0145201798") in new stack
-- Executing Monitor("mISDN/4-1", 
"gsm|20070323-093814-0145201798-2130|mb") in new stack
-- Executing Queue("mISDN/4-1", "2130|rntT|||10") in new stack
P[ 4] * IND : Indication [3] from s
P[ 4]  --> * IND :  ringing pid:2
P[ 4] I SEND:ALERTING oad:0145201798 dad:0146472130 pid:2
P[ 4]  --> bc_state:BCHAN_CLEANED
P[ 4]  --> * SEND: State Ring pid:2
P[ 4]  --> incoming_early_audio off
-- Called SIP/0146472130
-- Called SIP/ekiga
-- SIP/0146472130-08199d18 is ringing

I didn't touch to the mISDN installation other than upgrade the kernel 
and its modules (compiled on another machine). Should I also upgrade 
mISDNuser to 1.1.1 on that server?

Thanks,
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Re: [asterisk-users] SRTP testers needed

[Patrick]

On Fri, 2007-03-23 at 16:12 +0100, marek cervenka wrote:
> please look at
> http://www.voip-info.org/wiki/view/Asterisk+SRTP
> 
> and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle, 
> ...)

Does this work on 1.2 or 1.4 too or is it trunk only?

Regards,
Patrick

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[asterisk-users] cause 127

[Thomas Stein]

Hello.

Someone knows what cause 127 mean. The phone that i'm calling rings once and 
than the connection interrupts:

P[ 5]  --> l3id:10040
P[ 5]  --> cause:127
P[ 5]  --> out_cause:127
P[ 5]  --> state:ALERTING
P[ 5]  --> Channel: mISDN/5-1 hanguped new state:CLEANING
P[ 5] $$$ CLEANUP CALLED pid:3

best regards
-- 
Thomas Stein
knowledgeTools®  damit Sie sehen, was Sie wissen!
-
knowledgeTools International GmbH
Wallstraße 15 / 15 a
10179 Berlin

Fon: +49 30 726169 090
Fax: +49 30 726169 249

[EMAIL PROTECTED]
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Geschäftsführer: Oliver Seyboldt, Reinhard Kunz, Friedhelm Laschütza
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[asterisk-users] Semi-OT: Use T.38 ATAs to Extend fax lines

[Dave Fullerton]

Greetings.

I have a scenario I would like some advice on. I have a 100,000 square 
foot building that we will be moving some work crews into. It has 
offices on each end of the building and a fiber line between them. I 
currently have an asterisk 1.2 system in place and about 30 phones. My 
problem is they want a few fax machines out in the warehouse area where 
I currently have no wiring for POTS lines. So my question is this: If my 
D-Mark is at one end of the building, can I use some T.38 ATA's and 
Asterisk 1.4 to hook up my fax machines and save me the hassle of 
running copper all over just for fax machines? Or, am I better off just 
running the copper. I've done a few hours research and here's what I was 
thinking:



FAX A <-> GS-HT286 <-> |--|
   |  | || <-> POTS A
FAX B <-> GS-HT286 <-> | Asterisk | | GS | <-> POTS B
   |   1.4.2  | <-> |GXW-4104| <-> POTS C
FAX C <-> GS-HT286 <-> | to route | || <-> POTS D
   |   calls  |
FAX D <-> GS-HT286 <-> |--|


What I need is if FAX A calls out it goes over POTS A and if POTS A 
rings FAX A rings, etc. I'm confident I can set up asterisk, my concern 
is the reliability of the equipment and the calls. Is this a sound plan?


Thanks in advance.

-Dave
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Re: [asterisk-users] cause 127

[Doug Lytle]

Thomas Stein wrote:

Hello.

Someone knows what cause 127 mean. The phone that i'm calling rings once and 
  



A quick Google turned up the following link:

http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf

Doug


--

Ben Franklin quote:

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


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[asterisk-users] Debian Asterisk and MeetMe

[Alan Chandler]

I am trying to set up a simple conference call capability with asterisk

My meetme.conf

[general]
[rooms]
conf => 61
conf => 62
conf => 63
conf => 63


My extensions.conf
exten => 60,1,Answer()
exten => 60,2,MeetMe(,EMxp) 



When I enter extension 60 I enter a conference - I get repeated 

"you are entering conference 6 1 that is not a valid conference number 
you are entering conference 6 1" etc

CPU load rises to near 100% from mpeg123 and asterisk has exited.  This 
seems to be the same if I take the M of the options (ie no music on 
hold).

It appears to be something related to the the zap device

Mar 23 15:55:49 WARNING[19849]: chan_zap.c:913 zt_open: Unable to 
open '/dev/zap/pseudo': No such file or directory
Mar 23 15:55:49 ERROR[19849]: chan_zap.c:7518 chandup: Unable to dup 
channel: No such file or directory
Mar 23 15:55:49 WARNING[19849]: app_meetme.c:465 build_conf: Unable to 
open pseudo channel - trying device
Mar 23 15:55:49 WARNING[19849]: app_meetme.c:468 build_conf: Unable to 
open pseudo device


I presume, being the Debian binary package, that I don't have the 
ztdummy driver installed.  On the otherhand I read in viop-info.org 
that that wasn't necessary with Linux 2.6 (I am running 2.6.18).

Where do I go from here?

-- 
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [asterisk-users] cause 127

[Bruce Ferrell]

a quick google search for isdn cause codes gives this:

http://72.14.253.104/search?q=cache:mTG1cVFR5hIJ:www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf+isdn+cause+codes&hl=en&ct=clnk&cd=3&gl=us

Used to report a protocol error event only when no other cause in the 
protocol error class applies.


127
7F

Interworking, Unspecified.
Indicates that there has been interworking with a network which does not 
provide causes for actions it
takes. Thus, the precise cause for a message which is being sent cannot 
be ascertained.



Thomas Stein wrote:

Hello.

Someone knows what cause 127 mean. The phone that i'm calling rings once and 
than the connection interrupts:


P[ 5]  --> l3id:10040
P[ 5]  --> cause:127
P[ 5]  --> out_cause:127
P[ 5]  --> state:ALERTING
P[ 5]  --> Channel: mISDN/5-1 hanguped new state:CLEANING
P[ 5] $$$ CLEANUP CALLED pid:3

best regards

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Re: [asterisk-users] chan_capi and only one B channel usable?

[Karsten Wemheuer]

On 03/22/2007, Torge Szczepanek wrote:
> Hello list!
> 
> I have a Asterisk 1.2.10 running using the package from Backports.org
> for Debian Sarge.
> 
> I have setup chan_capi (0.6.5 from Backports) and it seems that I am
> only able to use on B-Channel.
> 
> When trying to place the second call I get:
> 
> CAPI INFO 0x34a2: No circuit / channel available
> 
> capi info shows:
> Contr1: 2 B channels total, 1 B channels free
> 
> And capi.conf is:
> 
> [ISDN1]
> msn=
> isdnmode=msn
> incomingmsn=*
> controller=1
> group=1
> softdtmf=on
> accountcode=
> context=remote
> echosquelch=1
> echocancel=yes
> echotail=64
> callgroup=1
> devices=2
> 
> Any ideas?

If You don't have any active calls from * using ISDN, there may be other
software using ISDN via the capi stack in Your box. The message
> capi info shows:
> Contr1: 2 B channels total, 1 B channels free
means, that there is one B-Channel used by CAPI (not neccessarily *).

HTH,
Karsten

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Re: [asterisk-users] cause 127

[Thomas Stein]

On Friday 23 March 2007, Doug Lytle wrote:
> Thomas Stein wrote:
> > Hello.
> >
> > Someone knows what cause 127 mean. The phone that i'm calling rings once
> > and
>
> A quick Google turned up the following link:
>
> http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code
>_List.pdf

Thank you Doug. But i'm afraid this doesn't get me any further. I think i have 
to spend more time on this rig.

best regards
-- 
Thomas Stein
knowledgeTools®  damit Sie sehen, was Sie wissen!
-
knowledgeTools International GmbH
Wallstraße 15 / 15 a
10179 Berlin

Fon: +49 30 726169 090
Fax: +49 30 726169 249

[EMAIL PROTECTED]
www.knowledgetools.de

Sitz  Berlin
AG  Berlin-Charlottenburg
HRB 86378

Geschäftsführer: Oliver Seyboldt, Reinhard Kunz, Friedhelm Laschütza
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[asterisk-users] Noob question regarding PCI 2.x & TDM400P Card

[Barton Fisher]

I have some old PC's I want to build as a test box - It's up and running 
OK now.  Now I installed a TDM400P and there is nothing I can do to get 
the card to come up.  My guess is the box is not PCI 2.2 compliant or 
does it need to be to see the card?


Thanks, Bart

Here's what I know:

Processors  1
Model   Pentium III (Katmai)
CPU Speed   551.37 MHz
Cache Size  512 KB
System Bogomips 1103.57
PCI Devices 
-   Bridge: Intel Corporation 82371AB/EB/MB PIIX4 ACPI
-   Ethernet controller: Intel Corporation 82557/8/9 [Ethernet Pro 100]
-   Host bridge: Intel Corporation 440BX/ZX/DX - 82443BX/ZX/DX Host bridge
-   IDE interface: Intel Corporation 82371AB/EB/MB PIIX4 IDE
-   ISA bridge: Intel Corporation 82371AB/EB/MB PIIX4 ISA
-   PCI bridge: Intel Corporation 440BX/ZX/DX - 82443BX/ZX/DX AGP bridge
-   USB Controller: Intel Corporation 82371AB/EB/MB PIIX4 USB
-   VGA compatible controller: Chips and Technologies F69000 HiQVideo


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[asterisk-users] SIP/IAX peers UNREACHABLE and audio loss

[Edoardo Serra]

Hi all,
	I'm having a problem with some Asterisk servers interconnected with 
each other using IAX (I also tried with SIP without solving the problem)


Sometimes, with apparently no reason, some peers become UNREACHABLE
(I have qualify=yes in iax.conf) and REACHABLE again as soon as
another qualify test is made.

Our users are also complaining about audio loss during their calls,
apparently randomly, everything goes ok for days and bad for another few 
days.


I strongly believe the 2 problems are strictly related because in the 
logs I see REACHABLE / UNREACHABLE messages only for certains days

without regularity.
The days in wich i see a lot of messages are exactly the days with
most of complaint about audio loss

I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE)
are quite always during business hours, this makes me think at somewhat 
related to load (cpu load, badwidth load, calls load, etc...)


But, looking at hardware specs of our lan, servers and average load I 
don't think they are over-stressed.


Our servers are all:
2 x Intel(R) Xeon(TM) CPU 3.20GHz
1 GB RAM
2 x IDE HDDs Software RAID 1
Asterisk 1.2.13 with res_perl
Gentoo Linux
Some of them has a Sangoma card connected with an E1

Most ot these are on the same LAN, interconnected with a 1 GB switch
(I don't think it should be a bandwidth problem).

Load averages of these server is varying from 0.5 to 1.0
(I guess it should be ok)

On each server we don't have more than 50 concurrent calls
(bridged SIP <-> IAX2 or IAX2 <-> ZAP)

Used codec is mostly G729

Sometimes on asterisk cli i see some messages like
"Avoided initial deadlock for '0x9fd130', 10 retries!"
I don't know if it could be somehow related.

Someone of you can point me in the right direction ?

Tnx in advance

Regards

Ing. Edoardo Serra
WeBRainstorm S.r.l.

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[asterisk-users] HUD Lite server on Debian

[Giorgio Incantalupo]

Hi,
anybody knows where to find a HUD Lite Server package for Debian  or a 
tar.gz to install?

(I tried to use alien on a  .rpm without success)

Thx!

Giorgio
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[asterisk-users] PC / Phone Combo

[Chris Gamble]

Is there a phone in the sub-400$ department that has at least a 10 inch display 
and a nearly modern web browser with keyboard hook-up that is either sip or iax 
based? I dont want to do video, just have a good web browser built into the 
phone.
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Re: [asterisk-users] Microsoft launches first PABX

[Anselm Martin Hoffmeister]

Am Freitag, den 23.03.2007, 17:09 +0800 schrieb Christopher Chan:
> Anselm Martin Hoffmeister wrote:
> > Am Donnerstag, den 22.03.2007, 22:17 -0700 schrieb shadowym:
> > Let us see the facts: Telephone systems with more than a handful
> > telephones and more than just the ability to call (be it voicemail,
> > conferencing, queues, agents...) are complicated, and in most cases need
> > to be tailored to the customers' needs. As long as the "customer" is not
> > an IT-ish company, they will hopefully understand that getting all the
> > knowledge about this internally costs work hours (and thus, money) the
> > same - and experience is something that can not be learned in a few
> > hours of document study and point-and-clicking. High-quality solutions
> > need professional hands, pals, possibly yours.
> > 
> > This will by no means be the death of the technical consulting around
> > telephone PABXs.
> 
> Er...is not this what asterisk is about? telephone PABX guys sniff at 
> computer guys moving in their space.

But still, the need for experts is there; a PABX (be it from Nortel,
Avaya or an Asterisk-based one) is not a teaspoon but a Swiss army
knife.

Go and ask Mr. Joe Average what all those tools on a proper Swiss army
knife are for. If he manages to get all the blades and tools out at
all ;-)

I do not see reason to worry about a decrementing need for engineer-type
people who know their job and their toolbox, that was what I wanted to
express. Campains for better, more widespread personal computing have
not removed the need for operational knowledge or "the neighbour" for
fixing a broken .DLL; customers may have hoped so for decades.

Once again, let Microsoft's marketing soap bubbles burst and see if
anything else comes out than wet air.

Regards
Anselm

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RE: [asterisk-users] PC / Phone Combo

[Dean Collins]

Hi Chris,
What are you hoping to do on the browser? I don't have a solution for
you but curious about what your intentions are.

Could a wireless display or similar with a softphone application provide
what you are looking to achieve?

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Chris Gamble
> Sent: Friday, 23 March 2007 1:42 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] PC / Phone Combo
> 
> 
> Is there a phone in the sub-400$ department that has at least a 10
inch display and
> a nearly modern web browser with keyboard hook-up that is either sip
or iax
> based? I dont want to do video, just have a good web browser built
into the phone.
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RE: [asterisk-users] Noob question regarding PCI 2.x & TDM400P Card

[Yuan LIU]

From: Barton Fisher <[EMAIL PROTECTED]>
Date: Fri, 23 Mar 2007 09:26:44 -0800

I have some old PC's I want to build as a test box - It's up and running OK 
now.  Now I installed a TDM400P and there is nothing I can do to get the 
card to come up.  My guess is the box is not PCI 2.2 compliant or does it 
need to be to see the card?


I had a similar situation.  What I found was: the CMOS setup program had an 
option to turn PCI 2.2 on or off - default was off.  Later motherboards no 
longer have this.


Yuan Liu


Thanks, Bart

Here's what I know:

Processors  1
Model   Pentium III (Katmai)
CPU Speed   551.37 MHz
Cache Size  512 KB
System Bogomips 1103.57
PCI Devices
-   Bridge: Intel Corporation 82371AB/EB/MB PIIX4 ACPI
-   Ethernet controller: Intel Corporation 82557/8/9 [Ethernet Pro 100]
-   Host bridge: Intel Corporation 440BX/ZX/DX - 82443BX/ZX/DX Host bridge
-   IDE interface: Intel Corporation 82371AB/EB/MB PIIX4 IDE
-   ISA bridge: Intel Corporation 82371AB/EB/MB PIIX4 ISA
-   PCI bridge: Intel Corporation 440BX/ZX/DX - 82443BX/ZX/DX AGP bridge
-   USB Controller: Intel Corporation 82371AB/EB/MB PIIX4 USB
-   VGA compatible controller: Chips and Technologies F69000 HiQVideo






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Re: [asterisk-users] Cisco 30VIP Phone

[Chris Nighswonger]

On 3/22/07, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:

On 3/22/07, Chris Nighswonger <[EMAIL PROTECTED]> wrote:
> 1.4.1
>
I've got one of those at home and a test system running 1.4.2.  I'll
take a look tonight and see if there is anything obvious.  I'm not a
developer, though.  I know one of the guys working on chan_skinny uses
30VIPs, so I would have thought it worked.

I do know when I tried the 30VIP on chan_sccp, it was doing some weird things.


Wondering if you had a chance to look at your 30VIP?

I have three registering with * and having basic functionality. I am
at a loss to know how to program the buttons (other than dtmf, hold,
mute, spkr). Here is what the * console shows when one of the phones
registers:

   -- Starting Skinny session from 192.168.0.70
   -- Device 'SEP000196C00CDC' successfully registered
Device capability set to '12'
Adding button: 9, 1
Adding button: 1, 0
Adding button: 15, 0
Adding button: 126, 0
Adding button: 5, 0
Adding button: 125, 0

It appears that * is setting up some buttons. But where it is getting
the config info, I don't know.

Thanks,
Chris
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[asterisk-users] Sendmail and exchange for voicemail integration

[Jordan Novak]

I am having real trouble getting Asterisk to send to exchange. They are
on the same LAN. Does anyone know of a walkthrough for this setup. I
have gotten it to work before, but that was to a hotmail account.
 
Jordan Novak
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Re: [asterisk-users] AsteriskNow Beta 4 with T1 Cards?

[Tzafrir Cohen]

On Fri, Mar 23, 2007 at 11:09:54AM -0400, Brian K. Alexander,  Jr. (Vision 
Point Systems) wrote:
> Thanks for the quick response but getting the cards to work with Asterisk is
> not the problem - what I want is for them to show up in asterisk-gui.
> 
> For example, I have a little test system sitting here. It has Fedora6,
> Zaptel/Libpri/Asterisk 1.4 and asterisk-gui. It has an analog card installed
> but the gui can not detect the analog ports. So I can not use those ports
> from asterisk-gui. I can manually configure them just fine and they work -
> but I can not access them in the asterisk-gui. I know this is not a digital
> card but it is the same problem - asterisk-gui can not detect the card so I
> need to somehow force the card into the configuration.
> 
> Is there a way to manually configure the system so that ports that the gui
> normally could not detect would be available?

The asterisk-gui uses /etc/asterisk/zapscan.conf, which is detected at
boot time (by zapscan which is run from /etc/init.d/zaptel)

As genzaptelconf is now able to replace the zapscan utility of the
asterisk-gui, and also to detect digital spans and give them resonable
values in many cases, I wonder if it can do the job for you.

Currently, genzaptelconf -z will still pront nothing for digital spans.
Is there anything that you can add to /etc/asterisk/zapscan.conf to make
the digital span available?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Cisco 30VIP Phone

[Chris Nighswonger]

On 3/23/07, Chris Nighswonger <[EMAIL PROTECTED]> wrote:


I have three registering with * and having basic functionality. I am
at a loss to know how to program the buttons (other than dtmf, hold,
mute, spkr). Here is what the * console shows when one of the phones
registers:

-- Starting Skinny session from 192.168.0.70
-- Device 'SEP000196C00CDC' successfully registered
Device capability set to '12'
Adding button: 9, 1
Adding button: 1, 0
Adding button: 15, 0
Adding button: 126, 0
Adding button: 5, 0
Adding button: 125, 0

It appears that * is setting up some buttons. But where it is getting
the config info, I don't know.


Sorry for answering my own post, however it may help someone else:

Soft button configuration is set in skinny.c

I'm still looking for some explaination of the logic and sytax of setting them.

Chris
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Re: [asterisk-users] Sendmail and exchange for voicemail integration

[Bruce Reeves]

Jordan,

Assuming that the voicemail users are email users on the domain for exchange
then your DNS entries for MX will take care of most of the work. Sendmail on
the Centos installs I have done has required no changes to the default
config to work with our exchange servers. You probably will want to make
sure that the SMTP protocol on Exchange allows the Sendmail server to relay.

On 3/23/07, Jordan Novak <[EMAIL PROTECTED]> wrote:


 I am having real trouble getting Asterisk to send to exchange. They are
on the same LAN. Does anyone know of a walkthrough for this setup. I have
gotten it to work before, but that was to a hotmail account.

 Jordan Novak

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--
Bruce Reeves
Nortex Networks
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Re: [asterisk-users] PC / Phone Combo

[Bruce Reeves]

Sounds like your looking for a laptop with a good softphone. You take an
ultralight laptop and you could probably meet all these requirements plus
some.

On 3/23/07, Dean Collins <[EMAIL PROTECTED]> wrote:


Hi Chris,
What are you hoping to do on the browser? I don't have a solution for
you but curious about what your intentions are.

Could a wireless display or similar with a softphone application provide
what you are looking to achieve?



Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Chris Gamble
> Sent: Friday, 23 March 2007 1:42 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] PC / Phone Combo
>
>
> Is there a phone in the sub-400$ department that has at least a 10
inch display and
> a nearly modern web browser with keyboard hook-up that is either sip
or iax
> based? I dont want to do video, just have a good web browser built
into the phone.
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--
Bruce Reeves
Nortex Networks
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RE: [asterisk-users] Sendmail and exchange for voicemail integration

[Michael Collins]

Jordan,

 

I don't know if you've down this step before, but my network admin sent
me these instructions a few months ago.  It allows you to tell your
Exchange Server's SMTP to allow relays from specific domains, hosts, or
subnets.  Hope it helps.  (Works for Exch 2000 and 2003.)

-MC

1.  Go to Exchange System Manager

2.  Drill down to Servers, (your Exchange server), Protocols, SMTP,
Default SMTP Virtual server.

3.  On Default Virtual Server, right click on it, select properties.
Select the Access tab on the top, then select the "Relay" button.

4.  On the "Relay Restrictions" window, make sure the "Only the list
below" button is selected.  

5.  Add an allowed IP, subnet or domain name

6.  When done, hit OK 3 times and that's it.

 

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Friday, March 23, 2007 10:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sendmail and exchange for voicemail
integration

 

Jordan,

Assuming that the voicemail users are email users on the domain for
exchange then your DNS entries for MX will take care of most of the
work. Sendmail on the Centos installs I have done has required no
changes to the default config to work with our exchange servers. You
probably will want to make sure that the SMTP protocol on Exchange
allows the Sendmail server to relay. 

On 3/23/07, Jordan Novak <[EMAIL PROTECTED]> wrote:

I am having real trouble getting Asterisk to send to exchange. They are
on the same LAN. Does anyone know of a walkthrough for this setup. I
have gotten it to work before, but that was to a hotmail account.

 

Jordan Novak


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-- 
Bruce Reeves
Nortex Networks 

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[asterisk-users] Doorphone vs. Grandstream BT101

[Jay Milk]

I've done all the googling I can on this, and have come to the 
conclusion that a Grandstream BT101 can be abused to be a door phone.  
Could someone with access to one, confirm that the following is possible?


Researched:
1. When set to auto-answer, dialing the phone will result in a short 
beep and instant speaker-phone connection.
2. When pressing the "message" button while on-hook, the phone will 
activate speaker-phone and dial the number configured for voice mail 
retrieval.


Assumptions:
3. Pressing the "message" button additional times will simply be ignored 
by the phone.
4. Hanging up the other end of the call will deactivate the speaker 
phone and cause the phone to go on-hook. (This is the behavior I see on 
a Polycom 430).


If the researched functions and my assumptions are correct, this phone 
would make an ideal door-phone;  The message button becomes the 
call-button, which rings every phone in the house until answered (for 
intercom).  It could even take messages.  Listen-in on the door works 
through the auto-answer feature.


Could someone confirm this for me?

Thanks!
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[asterisk-users] IAX2 certificate based (RSA) user auth in 1.4.x

[Kai-Uwe Jensen]

I have a working 1.2.17 installation, where calls are successfully
passed from a "slave
server" to a "master server" via IAX2. Authentication between the
slave and master server
is set up with RSA certificates (inkeys and outkey, auth=rsa). When
invoking the dialplan
(Dial(IAX2/master/${EXTEN:1},30)), everything "just works". The master
will accept an
authenticated call from slave.

Migrating the master to 1.4.2 (and 1.4.1) will break things. The call
does not get accepted
by the master anymore, however, it also does not get rejected
outright. Instead it will
time out, with repeated AUTHREQ requests made from the master to the slave.

Replacing the 1.2.17 slave with a 1.4.x slave (and using the exact
same config) will have
the call succeed against a 1.2.x master.

This appears to be an issue of not being able to match the user
(slave) of an incoming call
request on the master to a valid "friend" definition in iax.conf in
1.4.x. Can anyone confirm
that RSA certificate-based authentication has been observed working in
1.4.x? If so, is
there a secret to make it work? I have not found anything documented I deemed
relevant in this context, but I may have overlooked things.

Thanks!
--
"I am Dyslexic of Borg. Fusistance is retile. Your ass will be laminated!"
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[asterisk-users] RE: PC / Phone Combo

[Chris Gamble]

Its for a collections type application. Accounts passed due will show up on the 
screen in either an automated call fashion, or as a result of an incoming call. 
I need the physical interface to be somewhat simple since the end users are not 
exactly computer friendly. I had considered using the polycom 601 series, but 
the web browser screen is just a tad too small, and too limited in 
functionality.


>
> Hi Chris,
> What are you hoping to do on the browser? I don't have a solution for
> you but curious about what your intentions are.
>
> Could a wireless display or similar with a softphone application provide
> what you are looking to achieve?
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Re: [asterisk-users] Doorphone vs. Grandstream BT101

[Dave Fullerton]

Jay Milk wrote:
I've done all the googling I can on this, and have come to the 
conclusion that a Grandstream BT101 can be abused to be a door phone.  
Could someone with access to one, confirm that the following is possible?


Researched:
1. When set to auto-answer, dialing the phone will result in a short 
beep and instant speaker-phone connection.
2. When pressing the "message" button while on-hook, the phone will 
activate speaker-phone and dial the number configured for voice mail 
retrieval.


Assumptions:
3. Pressing the "message" button additional times will simply be ignored 
by the phone.
4. Hanging up the other end of the call will deactivate the speaker 
phone and cause the phone to go on-hook. (This is the behavior I see on 
a Polycom 430).


If the researched functions and my assumptions are correct, this phone 
would make an ideal door-phone;  The message button becomes the 
call-button, which rings every phone in the house until answered (for 
intercom).  It could even take messages.  Listen-in on the door works 
through the auto-answer feature.


Jay,

If my memory serves me your assumptions are correct. When the BT is set 
to auto-answer through the config you get a short warble-like sound and 
the phone goes into speaker mode (I'm using one for paging in that exact 
fasion). I'm pretty sure that if you are on-hook, pressing the message 
button will instantly place the call and go into speaker mode (if noone 
verifies this I'll try it this weekend). I don't know about #3. #4 is 
correct when the phone is set to auto-answer. If it is not on 
auto-answer the phone will play a busy signal when the other party hangs 
up which I always thought was kind of dumb.


-Dave

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Re: [asterisk-users] Doorphone vs. Grandstream BT101

[Doug Lytle]

Jay Milk wrote:
I've done all the googling I can on this, and have come to the 
conclusion that a Grandstream BT101 can be abused to be a door phone.  
Could someone with access to one, confirm that the following is possible?


Researched:
1. When set to auto-answer, dialing the phone will result in a short 
beep and instant speaker-phone connection.
I have this setup now, but don't recall the short beep.  It may be 
configurable.


2. When pressing the "message" button while on-hook, the phone will 
activate speaker-phone and dial the number configured for voice mail 
retrieval.




Correct.


Assumptions:
3. Pressing the "message" button additional times will simply be 
ignored by the phone.


I have several, I can check this weekend.

4. Hanging up the other end of the call will deactivate the speaker 
phone and cause the phone to go on-hook. (This is the behavior I see 
on a Polycom 430).




I would have to say correct as well, since I'm using it as a paging unit 
and it does hang up after playing back the audio file.



Something to consider, the BT101's speak phone has no Echo cancellation 
whatsoever and sounds just awful in a two way conversation.


Doug


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[asterisk-users] Problem with busy and unavailable

[Stefan Guenther]

Hi,

although setting up voicemail for busy and unavailable should be easy, things 
aren't working the way they should in my configuration (asterisk 1.2.14 
bristuffed):

Here's the relevant part of the extensions.conf:

exten => 56830976,1,Answer()
exten => 56830976,2,Dial(SIP/hbaumgart,20,tr)
exten => 56830976,3,VoiceMail,u76
exten => 56830976,4,Hangup
exten => 56830976,103,VoiceMail,b76
exten => 56830976,104,Hangup

If not all documentations that I read are wrong, this configuration should be 
correct.

And here are the corresponding debugging messages:

-- SIP/hbaumgart-081a8f48 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
-- Executing VoiceMail("CAPI/contr1/56830976/-5", "u76") in new stack
-- Playing '/var/spool/asterisk/voicemail/default/76/unavail' 
(language 'en')
-- Setting up echo canceller (PLCI=0x501, function=1, options=2, tail=64)
   > sent FACILITY_REQ (PLCI=0x501)
   > sent FACILITY_REQ (PLCI=0x501)
-- Echo canceller successfully set up (PLCI=0x501)

Asterisk recognizes that the phone is busy, but why does it then play the 
unavailable message?

Thanks for any hint,

Stefan
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Re: [asterisk-users] Problem with busy and unavailable

[James FitzGibbon]

You might be reading out of date documentation.  In 1.2.14, if you don't
have "priorityjumping=yes" in extensions.conf and you don't activate
priority jumping for an individual application (for Dial(), you add "j" to
the options string), then the jumping behaviour you are expecting to happen
(n+101) doesn't happen.

If you don't want to do this via priority jumping, you can branch based on
the ${DIALSTATUS} variable:

exten   => s,n,Dial(${ARG2}|20)
exten   => s,n,Goto(s-${DIALSTATUS},1)
exten   => s-NOANSWER,1,Voicemail(${ARG1}|u)
exten   => s-NOANSWER,n,Hangup
exten   => s-BUSY,1,Voicemail(${ARG1}|b)
exten   => s-BUSY,n,Hangup
exten   => _s-.,1,Goto(s-NOANSWER,1)

This is part of a macro invoked as Macro(mailbox,channel), but you get the
general idea.

On 3/23/07, Stefan Guenther <[EMAIL PROTECTED]> wrote:



exten => 56830976,1,Answer()
exten => 56830976,2,Dial(SIP/hbaumgart,20,tr)
exten => 56830976,3,VoiceMail,u76
exten => 56830976,4,Hangup
exten => 56830976,103,VoiceMail,b76
exten => 56830976,104,Hangup

If not all documentations that I read are wrong, this configuration should
be
correct.

Asterisk recognizes that the phone is busy, but why does it then play the
unavailable message?





--
--
j.
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RE: [asterisk-users] RE: PC / Phone Combo

[Dean Collins]

Nah just pop the information to a desktop pc display, it'll work out a
lot better in the long run.

You're right the polycom displays wont provide you enough real estate
space for the information.

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Chris Gamble
> Sent: Friday, 23 March 2007 4:22 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] RE: PC / Phone Combo
> 
> Its for a collections type application. Accounts passed due will show
up on the
> screen in either an automated call fashion, or as a result of an
incoming call. I need
> the physical interface to be somewhat simple since the end users are
not exactly
> computer friendly. I had considered using the polycom 601 series, but
the web
> browser screen is just a tad too small, and too limited in
functionality.
> 
> 
> >
> > Hi Chris,
> > What are you hoping to do on the browser? I don't have a solution
for
> > you but curious about what your intentions are.
> >
> > Could a wireless display or similar with a softphone application
provide
> > what you are looking to achieve?
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[asterisk-users] Re: Dial(Local/[EMAIL PROTECTED])?

[Benny Amorsen]

> "RH" == Rizwan Hisham <[EMAIL PROTECTED]> writes:

RH> OK, but y would i want to use it. i mean y not use goto and y
RH> this? and what dialout files are you talking about?

You can't Goto in queues.conf


/Benny


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Re: [asterisk-users] SRTP testers needed

[marek cervenka]

On Fri, 2007-03-23 at 16:12 +0100, marek cervenka wrote:

please look at
http://www.voip-info.org/wiki/view/Asterisk+SRTP

and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle,
...)


Does this work on 1.2 or 1.4 too or is it trunk only?


trunk only ... now
no testers, no stable release

---
Marek Cervenka
Centrum Vypocetni Techniky
CVT - http://cvt.fpf.slu.cz
FPF SLU OPAVA   - http://www.fpf.slu.cz
LCNA- http://lcna.slu.cz
===

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Re: [asterisk-users] Doorphone vs. Grandstream BT101

[Jay Milk]

Doug Lytle wrote:

Jay Milk wrote:
I've done all the googling I can on this, and have come to the 
conclusion that a Grandstream BT101 can be abused to be a door 
phone.  Could someone with access to one, confirm that the following 
is possible?


Researched:
1. When set to auto-answer, dialing the phone will result in a short 
beep and instant speaker-phone connection.
I have this setup now, but don't recall the short beep.  It may be 
configurable.


2. When pressing the "message" button while on-hook, the phone will 
activate speaker-phone and dial the number configured for voice mail 
retrieval.




Correct.


Assumptions:
3. Pressing the "message" button additional times will simply be 
ignored by the phone.


I have several, I can check this weekend.

4. Hanging up the other end of the call will deactivate the speaker 
phone and cause the phone to go on-hook. (This is the behavior I see 
on a Polycom 430).




I would have to say correct as well, since I'm using it as a paging 
unit and it does hang up after playing back the audio file.



Something to consider, the BT101's speak phone has no Echo 
cancellation whatsoever and sounds just awful in a two way conversation.


Doug
Thanks to Dave and Doug for the quick responses.  I'm looking forward to 
hearing the response on #3, but I think I'll get get one of these 
devices to play with this weekend.  At worst, it'll be a usable garage 
or basement phone.


Doug, I didn't even consider audio-quality on this, as even with the 
most rudimentary speaker phone circuits, phones seem pretty usable these 
days.  I was planning to put this in a custom door-box anyway, along 
with a water-resistant speaker (plastic membrane).  Considering our 
wide-open porch and some physical separation of the mic/speaker, the 
echo may not be as much of an issue as protection from the elements. 

And contrary to what someone asked me in private, wiring isn't an issue 
-- I do have cat5 at the door bell :)


Thanks,
JM
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RE: [asterisk-users] Doorphone vs. Grandstream BT101

[Dean Collins]

> -Original Message-

> From: [EMAIL PROTECTED] [mailto:asterisk-users-

> [EMAIL PROTECTED] On Behalf Of Jay Milk

> Sent: Friday, 23 March 2007 5:58 PM

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

> Subject: Re: [asterisk-users] Doorphone vs. Grandstream BT101

> 

> 

> And contrary to what someone asked me in private, wiring isn't an
issue

> -- I do have cat5 at the door bell :)

> 

> Thanks,

> JM

> 

 

 

 

Like all good geeks should - correct Jay :-)


So did you run two lengths so that you have access to a IP Door camera
as well? Don't forget a few pairs for the electric strike to open the
door remotely from a web interface as well.

 

 

 

 

Regards,

Dean Collins
[EMAIL PROTECTED] 
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).

    
 

www.Mexuar.com  
Want to voice enable your website?
Use Corraleta to reach your customers in 10 seconds or less.

 



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[asterisk-users] switches

[phil . dawson]

anyone have experience with switches with QOS.  recommend makes / models?

I've experience of cisco, polycom and snom phones but how do they compare
to cheaper phones namely:

Aastra 9112I
Budgetone 200

any insight would be appreciated


Phil.

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Re: [asterisk-users] Refund from SellVoip?

[Tom Lynn]

Now I know where they've been spending my remaining balance...

On 3/21/07, Ira <[EMAIL PROTECTED]> wrote:


At 09:08 AM 3/21/2007, you wrote:
> > Does anybody know Jed Stafford?  As far as I can tell this ended up
> > being a one-man or two-man operation.  It's just sad.

I got a marketing email from them last week telling me about all
their cool new features.

Ira

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Re: [asterisk-users] Refund from SellVoip?

[Tom Lynn]

This is probably why they don't use PayPal anymore.  Now, there is no
resolution process that I can pursue, other than complaining to the Gov't.,
which I have.

On 3/20/07, Vicky <[EMAIL PROTECTED]> wrote:


I got money back around 6 months ago . It was a via paypal claim and hey
didn't reply till paypal's deadline so i got $30 back .

On 17/03/07, Ira <[EMAIL PROTECTED]> wrote:
>
> At 02:32 PM 3/16/2007, you wrote:
> >You were able to cancel service with Sellvoip?  That's impressive, that
>
> Actually, it's Voxee I tried to cancel and failed. I still use
> SellVOIP and it mostly works but support is a problem. I'm starting
> to use using Telasip more though as they work and have a POP only
> 19ms from here, a big advantage.
>
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Re: [asterisk-users] Semi-OT: Use T.38 ATAs to Extend fax lines

[Doug Lytle]

Dave Fullerton wrote:

Greetings.

I have a scenario I would like some advice on. I have a 100,000 square 
foot building that we will be moving some work crews into. It has 
offices on each end of the building and a fiber line between them. I 
currently have an asterisk 1.2 system in place and about 30 phones. My 
problem is they want a few fax machines out in the warehouse area 
where I currently have no wiring for POTS lines. So my question is 
this: If my D-Mark is at one end of the building, can I use some T.38 
ATA's and Asterisk 1.4 to hook up my 


If your CAT5 runs are from end to end, then you may be able to use some 
of the un-used pairs for your POTS lines.


Just a thought,

Doug


--
Ben Franklin quote:

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


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RE: [asterisk-users] Semi-OT: Use T.38 ATAs to Extend fax lines

[Dean Collins]

Hi Doug, Dave said "Fiber" not Cat5.

 

 

Hey Dave, You can get fiber to FXS solutions, not cheap though and I
don't know how far they scale down to (I used to sell Fujitsu carrier
grade FLX boxes into mining and similar spaces for similar 'large site'
problems like this).

 

 

Regards,

Dean Collins
[EMAIL PROTECTED] 
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).

    
 

www.Mexuar.com  
Want to voice enable your website?
Use Corraleta to reach your customers in 10 seconds or less.

 

> -Original Message-

> From: [EMAIL PROTECTED] [mailto:asterisk-users-

> [EMAIL PROTECTED] On Behalf Of Doug Lytle

> Sent: Friday, 23 March 2007 6:16 PM

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

> Subject: Re: [asterisk-users] Semi-OT: Use T.38 ATAs to Extend fax
lines

> 

> Dave Fullerton wrote:

> > Greetings.

> >

> > I have a scenario I would like some advice on. I have a 100,000
square

> > foot building that we will be moving some work crews into. It has

> > offices on each end of the building and a fiber line between them. I

> > currently have an asterisk 1.2 system in place and about 30 phones.
My

> > problem is they want a few fax machines out in the warehouse area

> > where I currently have no wiring for POTS lines. So my question is

> > this: If my D-Mark is at one end of the building, can I use some
T.38

> > ATA's and Asterisk 1.4 to hook up my

> 

> If your CAT5 runs are from end to end, then you may be able to use
some

> of the un-used pairs for your POTS lines.

> 

> Just a thought,

> 

> Doug

> 

> 

> --

> Ben Franklin quote:

> 

> "Those who would give up Essential Liberty to purchase a little
Temporary Safety,

> deserve neither Liberty nor Safety."

> 

> 

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Re: [asterisk-users] Semi-OT: Use T.38 ATAs to Extend fax lines

[Doug Lytle]

Dean Collins wrote:


Hi Doug, Dave said "Fiber" not Cat5.

Hey Dave, You can get fiber to FXS solutions, not cheap though and I 
don’t know how far they scale down to (I used to sell Fujitsu carrier 
grade FLX boxes into mining and similar spaces for similar ‘large 
site’ problems like this).





Oppps!

Doug


--
Ben Franklin quote:

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


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[asterisk-users] SER vs Asterisk?

[David Anderson]

We're going to be setting up Asterisk at our data center, as well as our
call center locations via an optical fiber point to point connection. Is it
best to have the servers communicate to eachother via SIP using SER, or just
use the Asterisk functions?

Thanks,
David
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Re: [asterisk-users] SER vs Asterisk?

[Marco Mouta]

Only with Asterisk you can handle it, but of course it depends on  your
requirements on scalability and redundancy needed.

How many agents? How many diferent locations? SIP trunk to your telco or
PSTN ? Remote Agents at home?

Post more details on your requirements and I believe there are so many
experienced users in this list all around the world that you will have good
tips here.





On 3/23/07, David Anderson <[EMAIL PROTECTED]> wrote:


We're going to be setting up Asterisk at our data center, as well as our
call center locations via an optical fiber point to point connection. Is it
best to have the servers communicate to eachother via SIP using SER, or just
use the Asterisk functions?

Thanks,
 David

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Re: [asterisk-users] SIP/IAX peers UNREACHABLE and audio loss

[Rajeev Natarajan]

Well, we have add similar issues - do you use a media gateway /.IP Phones /
softphones as your extensions?

We were running Audiocodes and for some reason (I suspect a poor ethernet
switch), when there are more than 15 people using the line, Audiocodes will
not respond to a qualify and asterisk will drop the call. Turned off qualify
(removed qualify=yes) and  things seem fine.

Rajeev

On 3/23/07, Edoardo Serra <[EMAIL PROTECTED]> wrote:


Hi all,
I'm having a problem with some Asterisk servers interconnected
with
each other using IAX (I also tried with SIP without solving the problem)

Sometimes, with apparently no reason, some peers become UNREACHABLE
(I have qualify=yes in iax.conf) and REACHABLE again as soon as
another qualify test is made.

Our users are also complaining about audio loss during their calls,
apparently randomly, everything goes ok for days and bad for another few
days.

I strongly believe the 2 problems are strictly related because in the
logs I see REACHABLE / UNREACHABLE messages only for certains days
without regularity.
The days in wich i see a lot of messages are exactly the days with
most of complaint about audio loss

I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE)
are quite always during business hours, this makes me think at somewhat
related to load (cpu load, badwidth load, calls load, etc...)

But, looking at hardware specs of our lan, servers and average load I
don't think they are over-stressed.

Our servers are all:
2 x Intel(R) Xeon(TM) CPU 3.20GHz
1 GB RAM
2 x IDE HDDs Software RAID 1
Asterisk 1.2.13 with res_perl
Gentoo Linux
Some of them has a Sangoma card connected with an E1

Most ot these are on the same LAN, interconnected with a 1 GB switch
(I don't think it should be a bandwidth problem).

Load averages of these server is varying from 0.5 to 1.0
(I guess it should be ok)

On each server we don't have more than 50 concurrent calls
(bridged SIP <-> IAX2 or IAX2 <-> ZAP)

Used codec is mostly G729

Sometimes on asterisk cli i see some messages like
"Avoided initial deadlock for '0x9fd130', 10 retries!"
I don't know if it could be somehow related.

Someone of you can point me in the right direction ?

Tnx in advance

Regards

Ing. Edoardo Serra
WeBRainstorm S.r.l.

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