Re: [Asterisk-Users] x100p/hangup detection issues? - SOLVED!

[Patrick Cantwell]

OK everybody, I have solved my problem!

The issue lies in how you handle the incoming call. I actually stumbled
across this while trying to find a better way of doing fax autodetection.
The trick is you *must* use Answer to pick up the call.  The *WRONG* way of
doing things is:

exten => s/,1,Zapateller
exten => s,1,NoOp
exten => s,2,PrivacyManager
exten => s,3,Goto(100,1); proceed to home phone
rules
exten => s,103,Hangup() ; In case caller doesn't
supply info correctly, hangup

(of course omit the zapateller/privacymanager stuff if you're not using it)

The *CORRECT* way of doing things is:

exten => s/,1,Zapateller   ; if CID is not present,
send telemarketer blast down the line
exten => s,1,NoOp ; if it is, don't do
anything :)
exten => s,2,PrivacyManager  ; check for CID.  If not
present, prompt caller for their number.
exten => s,3,Answer   ; pick up the call. this
allows asterisk/x100p to correctly detect phone company signalling!
exten => s,4,Ringing; generate some ringing
for the calling party
exten => s,5,Goto(100,1); proceed to home phone
rules
exten => s,103,Hangup() ; In case caller doesn't
supply info correctly, hangup

Note the rules 3 and 4, Answer and Ringing -- they do what you'd expect.
Answer takes the call from the PSTN and Ringing makes asterisk generate a
ringing tone while it handles the call internally.  This allows it to a) let
callers know the call is still in progress, and b) allows the fax rule to
pick up a fax machine during ringing!  I couldn't find this documented
anywhere, or I would have implemented it this way from the get-go.  Hope
this helps someone else! (Let me know if it does!)  Maybe this should be on
the wiki too? :)

Thanks,
Pat


- Original Message - 
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, December 05, 2003 1:22 AM
Subject: Re: [Asterisk-Users] x100p/hangup detection issues?


> On Thu, 4 Dec 2003, Jonathan Tew wrote:
>
> > We're testing with an X100P card.  When the caller on the POTS line
> > hangs up it never causes our IAX phones (DIAX in this case) to hang up.
> > Curious what you find out.
>
> a) try kewlstart if you're lucky enough to have it
> b) try BUSYDETECT
> c) alternatively in the US try callprogress
>
> - wasim
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>

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[Asterisk-Users] XBOX as and * Dedicated Server

[Miguel Cavazos]

Hello guys, i have been on this mailing list for some weeks now, and i
was wondering if someone here has installed linux on the XBOX and use it
as a dedicated server. Its a 200 USD computer and could make it perfect
to asterisk, its little and doesnt really take much space. My question
is could this make it for a stable server???

here are some links i found for linux on XBOX
http://xbox-linux.sourceforge.net/

some intresting screenshots found on that URL
http://xbox-linux.sourceforge.net/docs/screenshots.html

The only real thing that i dont know is where am i going to put the
X100p.

Miguel
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Re: [Asterisk-Users] Internet-to-phone gateway?

[wasim]

On Thu, 4 Dec 2003, Carl Youngblood wrote:

> Okay, I'm an asterisk newbie, so forgive me if this is really obvious, 
> but I'm wondering if there are services out that would let me simply 
> hook up an asterisk server to the internet and make as many outbound 
> calls as I wanted to for a monthly fee of some sort.  In other words, I 
> just have a server with an internet connection and someone else 
> provides the internet-to-phone gateway.  Come to think of it, would 
> asterisk even be needed for this kind of a solution?  Does anybody do 
> this?

www.nufone.net

 -wasim
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Re: [Asterisk-Users] Debian Testing / 2.4.22 / zaptel problems.

[Andrew Gillham]

Jeremy McNamara wrote:

Typical version skewTry linking to the kernel source that is 
actually running on the box.


Well as far as I can tell, the only version I have on the box is 2.4.22-1.
I certainly only have 'kernel-headers-2.4.22-1' installed and 'linux' 
symlinked
to that directory in /usr/src.

Are you saying my /usr/include would be skewed?  Since I thought that 
was from
the libc6-dev, not really kernel related?

I will try removing all of the -dev packages and re-installing them.

-Andrew

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[Asterisk-Users] Internet-to-phone gateway?

[Carl Youngblood]

Okay, I'm an asterisk newbie, so forgive me if this is really obvious, 
but I'm wondering if there are services out that would let me simply 
hook up an asterisk server to the internet and make as many outbound 
calls as I wanted to for a monthly fee of some sort.  In other words, I 
just have a server with an internet connection and someone else 
provides the internet-to-phone gateway.  Come to think of it, would 
asterisk even be needed for this kind of a solution?  Does anybody do 
this?

Thanks,
Carl Youngblood
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Re: [Asterisk-Users] x100p/hangup detection issues?

[wasim]

On Thu, 4 Dec 2003, Jonathan Tew wrote:

> We're testing with an X100P card.  When the caller on the POTS line 
> hangs up it never causes our IAX phones (DIAX in this case) to hang up.  
> Curious what you find out.

a) try kewlstart if you're lucky enough to have it
b) try BUSYDETECT
c) alternatively in the US try callprogress

- wasim
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RE: [Asterisk-Users] Channelbank Recomendation and GS102 question

[Ray Burkholder]

In what areas are you looking for the hybrid service?

Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunified.net
704 576 5101
> >
> >
> >>  We have an installation with 9 inbound voice channels (one is the 
> >>fax) and 768K data.  It is a Hybrid PRI.  It terminates into a
> >T100P.  It is working great!  The >cost was better than the 
> POTS plus data.
> >
> >Can I ask what Telephone/Internet service provider you are 
> getting this from?
> >Does anybody else have a setup like this?

> I, too, would be interested in hearing from what vendor you are 
> getting such a service.
> 
> JT


-- 
Scanned for viruses and dangerous content at 
http://www.oneunified.net and is believed to be clean.

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Re: [Asterisk-Users] x100p/hangup detection issues?

[Jonathan Tew]

Pat,

We're testing with an X100P card.  When the caller on the POTS line 
hangs up it never causes our IAX phones (DIAX in this case) to hang up.  
Curious what you find out.

Thanks,
Jonathan
Patrick Cantwell wrote:

Hi..
I've got an asterisk setup with an X100P card installed.. I'm 
noticing that upon hangup, it takes a good 3 to 5 seconds before 
asterisk realizes the line has been hung up and drops the call.. this 
causes my SIP phone to continue ringing, and occassional phantom voice 
mail messages to be left.. I'm located in good old standard North 
America, with a regular Verizon residential POTS line coming in.  I've 
checked polarity on the line (easy way, reversed it, then the card 
refused to hang up no matter what), and am using kewlstart on the FXO 
port.  I tried changing to loopstart with no change in results. I've 
poked around some and haven't found anything terribly helpful yet, 
except for another person asking about the same thing :)  Does anyone 
have any ideas?
Thanks,
Pat


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Re: [Asterisk-Users] voip-info.org is a great Resource ..BUT

[Jonathan Tew]

Being farely new to the Asterisk scene and searching for 
documentation I was wondering why the asterisk.org site didn't run a 
wiki.  There isn't anything as good as a wiki for collecting 
collaborative documentation.  Over time someone might want to convert 
the knowledge contained in the wiki to more updated formal 
documentation, but a wiki sure would be a great thing to have.

TC wrote:

Finally there is a place for us to contribute doc & info about * 
BUT it is realy beginning to feel the heat of its popularity
its takes me forever to edit pages on the site 
On IRC there are a number of service providers willing to donate 
CPU cycles & Bandwidth to support this resource ...

Does any one know who owns the site, and how we can help him move
to one of the generous * ISP willing to provide a faster home for it ?
It would also be nice for to resolve to asterisk-wiki.org :)

ps. if the owner wants to email off list i can provide contact info
for the generious donors ...


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Re: [Asterisk-Users] Debian Testing / 2.4.22 / zaptel problems.

[Jeremy McNamara]

Typical version skewTry linking to the kernel source that is 
actually running on the box.



Jeremy McNamara



Andrew Gillham wrote:

I'm having trouble getting zaptel to work on Debian Testing (Sarge) 
with a
2.4.22 kernel.

The errors I am seeing with 'insmod zaptel.o' are:
./zaptel.o: unresolved symbol devfs_unregister_R1c83d91a
./zaptel.o: unresolved symbol remove_wait_queue_R1bc53d4c
./zaptel.o: unresolved symbol __pollwait_R8ee1d6fc
./zaptel.o: unresolved symbol remove_proc_entry_R30e2f8d2
./zaptel.o: unresolved symbol devfs_register_chrdev_R69b94695
./zaptel.o: unresolved symbol proc_mkdir_R57411544
./zaptel.o: unresolved symbol devfs_register_Rbf40312b
./zaptel.o: unresolved symbol devfs_generate_path_R0c78ae56
./zaptel.o: unresolved symbol add_wait_queue_R498065b0
./zaptel.o: unresolved symbol create_proc_entry_Rebcd0c7f
./zaptel.o: unresolved symbol devfs_mk_symlink_R5b830122
./zaptel.o: unresolved symbol devfs_mk_dir_Rbf4f104c
This is with a stock install, with updates, and asterisk / zaptel from 
cvs.
Any ideas?

-Andrew

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[Asterisk-Users] Debian Testing / 2.4.22 / zaptel problems.

[Andrew Gillham]

I'm having trouble getting zaptel to work on Debian Testing (Sarge) with a
2.4.22 kernel.
The errors I am seeing with 'insmod zaptel.o' are:
./zaptel.o: unresolved symbol devfs_unregister_R1c83d91a
./zaptel.o: unresolved symbol remove_wait_queue_R1bc53d4c
./zaptel.o: unresolved symbol __pollwait_R8ee1d6fc
./zaptel.o: unresolved symbol remove_proc_entry_R30e2f8d2
./zaptel.o: unresolved symbol devfs_register_chrdev_R69b94695
./zaptel.o: unresolved symbol proc_mkdir_R57411544
./zaptel.o: unresolved symbol devfs_register_Rbf40312b
./zaptel.o: unresolved symbol devfs_generate_path_R0c78ae56
./zaptel.o: unresolved symbol add_wait_queue_R498065b0
./zaptel.o: unresolved symbol create_proc_entry_Rebcd0c7f
./zaptel.o: unresolved symbol devfs_mk_symlink_R5b830122
./zaptel.o: unresolved symbol devfs_mk_dir_Rbf4f104c
This is with a stock install, with updates, and asterisk / zaptel from cvs.
Any ideas?
-Andrew

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RE: [Asterisk-Users] Echo problem on conferencing....no analog interfaces

[Tom Lowe]

Now that actually brings up an interesting aspect.   "My" echo canceller
(the one at my end) would prevent the far end from hearing echo...so if
I don't have a canceller at my end, they would hear echo!  And, on a
local call, there's really no need to have an echo canceller...so I
guess that makes sense.

It still doesn't explain why one of the other two callers (both
accessing the conference exactly the same way) has echo and the other
doesn't.  

I guess I just have to live with the fact that mixing VoIP calls and TDM
calls on the same conference will simply have problems.  I assume there
are no other echo suppression settings that anyone can suggest.  

Tom



Tom Lowe, President/CTO
Compro Technologies, Inc.
512 South Main Street
Forked River, NJ  08731
My Phone:  +1-212-904-0788
Main Phone:  +1-609-242-2211
Fax:  +1-609-242-2212
Email:  [EMAIL PROTECTED]
Web: www.comprotech.com
 


-Original Message-
From: Steve Brown [mailto:[EMAIL PROTECTED] 
Sent: Thursday, December 04, 2003 7:43 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Echo problem on conferencingno analog
interfaces


There are probably echo suppressors on both ends of the LD circuit 
between NJ and CA, but none on the local NJ circuit.

Steve

Tom Lowe wrote:

>Not a silly question.  I've given that thought.
>
>To be honest, I'm not sure what kind of phone the California or NJ 
>callers were using.  However, we've had numerous conferece calls using 
>many other services and have never had echo problems.
>
>The problem, in this case, more likely has something to do with the NJ 
>leg of the call (call 2), and possibly my leg (I was caller 1), since 
>the echo happened when it was just the two of us.  If anything, the 
>"problem phone" would be my phone.  This is a Lucent Partner phone 
>system and phone.  Now, if it was MY phone causing the problem, then 
>both caller's 2 and 3 should be hearing their echo.  But they weren't.
>
>I'm stumped.
>
>
>For anyone who's trying to figure out how I'm doing an E1 PRI here in 
>the US, it's working like this:
>
>Verizon T1/PRI  -->  Cisco VCO/4K (Programmable switch)  -->  E1/PRI 
>--> MyAsterisk
>
>The other 2 calls are going like this:
>
>888 number --> SS7 --> VCO/4K --> T1/PRI --> Asterisk_1 --> IAX --> 
>MyAsterisk
>
>Tom
>
>
>
>Tom Lowe, President/CTO
>Compro Technologies, Inc.
>512 South Main Street
>Forked River, NJ  08731
>My Phone:  +1-212-904-0788
>Main Phone:  +1-609-242-2211
>Fax:  +1-609-242-2212
>Email:  [EMAIL PROTECTED]
>Web: www.comprotech.com
> 
>
>
>  
>


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Re: [Asterisk-Users] Implementing a ringback test function for Zap channels

[Eric Wieling]

My site http://www.fnords.org/~eric/asterisk/ has a sample Perl AGI 
Callback script.

Steve Rodgers wrote:

I'd like to add a test extension to implement ringback so that I can test a 
phone's ringer without having to use another channel in another room. The way
I'd like to implement this is to dial a test extension, get a tone, hang up, 
then one second later, have the system call me back at that extension.

There is a way to do this which is mentioned in the Asterisk white paper,
but it uses the old qcall mechanism which is depreciated. Is there any way
to do this with extension logic, or will it require writing a specialized app, 
or AGI script?

Steve.

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Re: [Asterisk-Users] Channelbank Recomendation and GS102 question

[Eric Wieling]

i-55.net which is a regional ISP for Louisiana, Mississippi and maybe 
one or two other nearby states provide us with a PRI (6 channels) and 
384K internet service over a single T-1 line.  They provide the service 
as a PRI and Ethernet Internet out of their Adtran channel bank.  We 
asked for the service to be provided this way, I'm sure they could have 
terminated both into our T100P if we had asked.  They are a small enough 
company that they actually care about their customers, but large enough 
to be an ISP and CLEC (their CLEC stuff is fairly recent) and provide 
good service.

We've had the service for a week or two.  This is a new office for us 
and it's not staffed yet.  There have been some annoying installation 
and configuration issues, but they worked with us to resolve them 
quickly.  The setup was not perfect, but if we had been using BellSouth 
of one of the "big guys" we would not have had any of those issues 
resolved by now.

When we added two additional channels for our PRI a couple of days after 
it was turned up someone didn't map the channels in the local Adtran 
unit and so those channels did not work.  Took 24 hours to resolve. 
Since the office is not staffed we did NOT push for a fast fix.  They 
took about 24 hours from the time we placed the order for additional 
channels until they turned them up, then an additional 24 hrs to fix the 
maping problem.  When I've dealt with RBOCS they spent at least two days 
just blaiming our equipment before they even tried to fix the problem, 
and that was with us ecsalating the issue with them.

I've been very happy with i-55.net.  Granted, they REALLY want to 
contract for our other current offices so they are treating us well, but 
I don't thing that is the major reason for our good support and service. 
 They were priced less that most of the other companies I've worked with.

John Todd wrote:
At 8:15 PM -0500 12/4/03, Jim Flagg wrote:

- Original Message -
From: "Walker Haddock" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, December 04, 2003 7:54 PM
Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 
question


 We have an installation with 9 inbound voice channels (one is the 
fax) and 768K data.  It is a Hybrid PRI.  It terminates into a
T100P.  It is working great!  The >cost was better than the POTS plus 
data.

Can I ask what Telephone/Internet service provider you are getting 
this from?
Does anybody else have a setup like this?


Very interesting.  I've had now two fights with providers (Verizon and 
SBC) who would not offer such a service, claiming that it was 
"impossible" to hybridize a PRI.  I think that's a great offering, and 
of course, it is possible, and especially appealing for Asterisk users.

I, too, would be interested in hearing from what vendor you are getting 
such a service.

JT
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Re: [Asterisk-Users] Asterisk freezing HELP

[TC]

>I have had several instances over the last month of Asterisk freezing,
> Does anyone have any suggestions? or ideas as to what may be causing it?
Sounds like some type deadlock
Take a look here for my tought and how to debuig dead locks
http://lists.digium.com/pipermail/asterisk-dev/2003-November/002384.html

Then let us know when you haved posted the info on bugs.digium.com


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[Asterisk-Users] voip-info.org is a great Resource ..BUT

[TC]

Finally there is a place for us to contribute doc & info about * 
BUT it is realy beginning to feel the heat of its popularity
its takes me forever to edit pages on the site 

On IRC there are a number of service providers willing to donate 
CPU cycles & Bandwidth to support this resource ...

Does any one know who owns the site, and how we can help him move
to one of the generous * ISP willing to provide a faster home for it ?

It would also be nice for to resolve to asterisk-wiki.org :)

ps. if the owner wants to email off list i can provide contact info
for the generious donors ...



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[Asterisk-Users] Asterisk freezing HELP

[mattf]

Hello,

I have had several instances over the last month of Asterisk freezing,
sometimes after 12 hours, sometimes after 8 days. The common elements are
that:
- all Zap channels lock[hangups don't register and no new calls in or out]
- no new in/outbound calls can be made on Zap or SIP channels
- people who are still connected to calls can continue to talk
- in the CLI interface, you can "show channels" and view other info like
"sip show peers", but you cannot do a "stop now" only way to stop asterisk
process is a kill -9
- all attempted connections to the manager interface time out
- the processor load and RAM usage on the machine is low.

I am experiencing these freezes on two separate identical RedHat 9 machines.
On each I have 4 T1s connected and a usual concurrent call volume of 45
Zap/SIP conversations at once 14 hours a day. The processor load is never
that high and the RAM usage is less than half.

There are never any consistent errors or warnings in the logs or the CLI.

I do have several AGI perl scripts in the dialplan and the SIP phones that
connect are almost all Grandstream 102's, That's all I can think of that may
be causing any problems. I have mpg123/musiconhold deactivated.

Has this happened to anyone else out there?

Does anyone have any suggestions? or ideas as to what may be causing it?

Thanks for any help,

MATT---
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RE: [Asterisk-Users] Port density: DS3 cards?

[Andy Hester]

> >I have been mulling over what it would take to get drivers done for
> >ImageStream's products.  They have a component architecture that 
> is supposed
> >to reduce development time/cost.  The component stuff is open 
> source.  The
> >part of the driver that you have to write can be open source or 
> proprietary.
> >I am not much of a coder, but someone more knowledgeable may be 
> able to do
> >it without too much trouble.  I am an ImageStream reseller - if you need
> >hardware I'll give you good pricing. ;)
> >
> >Andy
> 
> 
> Shoot, set me up with  42 2u servers 
> with dual TE410P boards, and then 12 M13 muxes, and then 1 12-port 
> DS3-to-OC12 mux (or 3 DS3-to-OC3 muxes, and one 3 port OC3-to-OC12 
> mux) and we can even test one of those OC12 boards that ImageStream 
> sells!
> 
> Why don't you ping someone at ImageStream and see if they're willing 
> to offer a DS3 developer kit for some interval (6 months? 8 months?) 
> to a developer if they show appropriate interest and expertise. 
> Anyone want to volunteer?
> 
> Actually, I'd ask a senior developer at ImageStream to see if they 
> think it's even possible first; they'll at least be able to say if 
> it's in the realm of sanity.  You have the inside track; let us know 
> what you hear.
> 
> JT
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I'll follow up on this tommorrow and let you know what I hear

Andy

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Re: [Asterisk-Users] Channelbank Recomendation and GS102 question

[TC]

- Original Message -
From: "Jonathan Moore" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, December 04, 2003 4:12 PM
Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 question


> I just went through this cycle. I did go ebay and went with a Carrier
Access
> Access Bank I, since I still haven't decided if this will be our
production
> system and I didn't want to invest too much to find out. After fighting
with it
> all day today, I would not recommend the AB1 for this application -- both
> inbound and outbound calling, because the AB1 doesn't support call
supervision
> on the inbound fxo ports. And their tech support says that Carrier Access
does
> not sell a model with call sup.
I am sorry but that CAC rep is on Crack  ;), I have em installed perfect
gear
ADIT 600 from Carrier Access most certainly does supprt Far End Disconnect
(watch the wrap)
http://www.carrieraccess.com/dbfiles/marketing/card_specsheets_pdf/fxo_dpt_8
channel_voice_service_card_spec_sheet.pdf

please see http://voip-info.org/wiki-Asterisk+Hardware
for first person experience with various CBs



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Re: [Asterisk-Users] Port density: DS3 cards?

[David Boreham]

 
> Actually, I'd ask a senior developer at ImageStream to see if they 
> think it's even possible first; they'll at least be able to say if 

I send this thread to someone at Imagestream...


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RE: [Asterisk-Users] Port density: DS3 cards?

[John Todd]

At 8:47 PM -0600 12/4/03, Andy Hester wrote:
 > The data-only cards for DS3 seem to be in the "reasonable" price
 range, though I have _no_ idea if they could be turned into
 TDM-capable cards.  Examples that were shown to me:
 http://oem.imagestream.com/PCI_720.html
 http://www.ace-electronics.com/Hardware/T1E1J1/wanPCI-1T3.html
 A little more time with Google perhaps would discover other
 solutions.  These are, from what I gather, very inexpensive devices
 in the grand scheme of things, and I believe some already offer Linux
 drivers (though no mention of open source that I could find, I
 imagine that these companies will be all over opening up more markets
 for their cards.)
 Of course, Digium could keep it's leadership and our (collective)
 money by starting to poke around at such a driver or card.  It's
 really a chicken-egg situation: nobody will want to muck with driver
 authorship or card production until there are buyers, and there won't
 be any buyers of such "experimental" technology unless it's cheap to
 experiment with, just like the T100P cards are.  Open source is still
 scary to bell-heads, and they will resist until they actually see
 (with their own eyes) a working system that replaces their $100k
 CisNorSiemAvaytelensaco boxes with a $7k PC/card combination.  Even
 then, it's still an uphill battle, but at least it's a battle,
 whereas right now it's a complete non-starter to open one's mouth
 about open source telephony gatewaying at truly large scale
 installations.  And, to be honest, the telco guys are correct at this
 moment.
 > JT

I have been mulling over what it would take to get drivers done for
ImageStream's products.  They have a component architecture that is supposed
to reduce development time/cost.  The component stuff is open source.  The
part of the driver that you have to write can be open source or proprietary.
I am not much of a coder, but someone more knowledgeable may be able to do
it without too much trouble.  I am an ImageStream reseller - if you need
hardware I'll give you good pricing. ;)
Andy


Shoot, set me up with  42 2u servers 
with dual TE410P boards, and then 12 M13 muxes, and then 1 12-port 
DS3-to-OC12 mux (or 3 DS3-to-OC3 muxes, and one 3 port OC3-to-OC12 
mux) and we can even test one of those OC12 boards that ImageStream 
sells!

Why don't you ping someone at ImageStream and see if they're willing 
to offer a DS3 developer kit for some interval (6 months? 8 months?) 
to a developer if they show appropriate interest and expertise. 
Anyone want to volunteer?

Actually, I'd ask a senior developer at ImageStream to see if they 
think it's even possible first; they'll at least be able to say if 
it's in the realm of sanity.  You have the inside track; let us know 
what you hear.

JT
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[Asterisk-Users] Implementing a ringback test function for Zap channels

[Steve Rodgers]

I'd like to add a test extension to implement ringback so that I can test a 
phone's ringer without having to use another channel in another room. The way
I'd like to implement this is to dial a test extension, get a tone, hang up, 
then one second later, have the system call me back at that extension.

There is a way to do this which is mentioned in the Asterisk white paper,
but it uses the old qcall mechanism which is depreciated. Is there any way
to do this with extension logic, or will it require writing a specialized app, 
or AGI script?

Steve.

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Re: [Asterisk-Users] Port density: DS3 cards?

[Greg Boehnlein]

On Thu, 4 Dec 2003, Bob Knight wrote:

> Steve Dolloff wrote:
> 
> >I would be seriously wary of putting a DS3's worth of voice traffic on a
> >TNT.  I don't believe they are rated to handle that much voice.  The
> >APX1000 would be a much better platform, but I don't know if you can
> >find one used.
>
> Skip the TNT's.  They are really a joke.
> I will admit, I am a bitter X-Livingston employee.
> First Lucent bought us for our cool gear, then they bought
> Ascend for sales and marketing..
> I still can't believe they kept the TNT alive and killed PM4.

The PM3 LIVES ON DUDE! :) I'm all about Livingson, and have refused to put 
the Asscend stuff in my data center. Seriously, Jake over at 
portmasters.com is doing some good stuff with the PM3. Now that we've got 
control of ComOS, it is just a matter of time before new ComOS releases 
start coming out for the unit. Several people have already rolled their 
own and added a few niggling fixes to the 3.9.1c1 code branch.

It would be great if we could find a way to use the PM3 as an inbound 
channel bank for Asterisk though. I have like 7 of them sitting in the 
back doing nothing..

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RE: [Asterisk-Users] Port density: DS3 cards?

[Andy Hester]

>
> The data-only cards for DS3 seem to be in the "reasonable" price
> range, though I have _no_ idea if they could be turned into
> TDM-capable cards.  Examples that were shown to me:
>
> http://oem.imagestream.com/PCI_720.html
> http://www.ace-electronics.com/Hardware/T1E1J1/wanPCI-1T3.html
>
> A little more time with Google perhaps would discover other
> solutions.  These are, from what I gather, very inexpensive devices
> in the grand scheme of things, and I believe some already offer Linux
> drivers (though no mention of open source that I could find, I
> imagine that these companies will be all over opening up more markets
> for their cards.)
>
> Of course, Digium could keep it's leadership and our (collective)
> money by starting to poke around at such a driver or card.  It's
> really a chicken-egg situation: nobody will want to muck with driver
> authorship or card production until there are buyers, and there won't
> be any buyers of such "experimental" technology unless it's cheap to
> experiment with, just like the T100P cards are.  Open source is still
> scary to bell-heads, and they will resist until they actually see
> (with their own eyes) a working system that replaces their $100k
> CisNorSiemAvaytelensaco boxes with a $7k PC/card combination.  Even
> then, it's still an uphill battle, but at least it's a battle,
> whereas right now it's a complete non-starter to open one's mouth
> about open source telephony gatewaying at truly large scale
> installations.  And, to be honest, the telco guys are correct at this
> moment.
>
> JT
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

I have been mulling over what it would take to get drivers done for
ImageStream's products.  They have a component architecture that is supposed
to reduce development time/cost.  The component stuff is open source.  The
part of the driver that you have to write can be open source or proprietary.
I am not much of a coder, but someone more knowledgeable may be able to do
it without too much trouble.  I am an ImageStream reseller - if you need
hardware I'll give you good pricing. ;)

Andy

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Re: [Asterisk-Users] Carrier Access Channel Bank Setup -- No hangup

[Robert Hajime Lanning]

> Thanks for the help. Can you explain the need to define all the channels
> in zapata.conf? I am not connecting devices to all the ports on the CB
> yet, so if I place the definitions into my groups 1 and 2 then things
> seem to be a bit strange when defining my outbound pstn calling.

As for defining all channels, I do it, more for completeness.  I don't know
if is would really change anything.

As for groups, I do it like this:
group = 1
channel => 1
group = 32
channel => 2-12
group = 2
channel = 13
group = 32
channel = 14-24

-- 
END OF LINE
   -MCP
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Re: [Asterisk-Users] Port density: DS3 cards?

[John Todd]

At 8:15 AM +0800 12/5/03, Steve Underwood wrote:
John Todd wrote:

Obviously, there are no DS3 TDM cards that are currently compatible 
with Zap channels.  (or are there?)

Does anyone know of an inexpensive DS3 card that could perhaps be 
used with Asterisk if one were to try to port the Zap drivers to 
such a card?  PCI, of course, would be the bus of choice.

I think there are quite a few discouraging comments to be made on 
that question.  Firstly, most companies that produce telecom 
hardware have silly overhead, and thus the price of their cards is 
astronomical.  Secondly, most companies that produce telecom 
hardware are of the opinion that transcoding (compression) should 
be done via DSP's, which inflates the cost of the card 
significantly.  Thirdly, most telecom hardware vendors would not 
consider allowing their drivers into the public domain if such 
development were to happen. I've talked to some parties (you know 
who you are) who have expressed some interest in building this type 
of interface, but a situation where I can actually put my hands on 
equipment is far better than speculative interest by those who have 
not even decided to go forward with design, no matter how 
interesting the end product sounds on the whiteboard.

However, regardless of all these negatives, I'm interested in any 
vendors anyone can offer as a starting point.

[snip]
I think this is a worthwhile thing to investigate. Does anyone here 
have experience with higher order cards under Linux? Which ones work 
well, and have solid drivers? Although the driver would (probably) 
need to be heavily modified if it is currently a data, rather than 
telephony, oriented driver, a good existing driver should save a lot 
of work. A DS3 is well within a PCI channel's capacity (the sum of 
the two directions is less than a 100mb Ethernet, although the DS3 
is continuous), but it is quite a lot of data. A suitable card would 
need an efficient interface if this is to work. For the Zaptel 
environment that would mean that it can burst data in 1ms (8 sample 
chunks), and would need to bus master the data into memory in a form 
that doesn't require masses of manipulation by software - e.g. 
reshuffling out of sequence data. Doing that for so many channels 
might create interesting latency challenges :-) Still, if you don't 
try, you can't start to address these issues, and work out a 
solution. It shouldn't be impractical to make a DS3 to TDMoE 
solution, provided the DS3 hardware is right, and the PC  has no 
quirky throughput issues.

This subject often comes up on the IRC channel. There seem to be a 
number of people interested in higher order links, but it really 
needs some positive action somewhere to kick off a real project.

Regards,
Steve


The data-only cards for DS3 seem to be in the "reasonable" price 
range, though I have _no_ idea if they could be turned into 
TDM-capable cards.  Examples that were shown to me:

http://oem.imagestream.com/PCI_720.html
http://www.ace-electronics.com/Hardware/T1E1J1/wanPCI-1T3.html
A little more time with Google perhaps would discover other 
solutions.  These are, from what I gather, very inexpensive devices 
in the grand scheme of things, and I believe some already offer Linux 
drivers (though no mention of open source that I could find, I 
imagine that these companies will be all over opening up more markets 
for their cards.)

Of course, Digium could keep it's leadership and our (collective) 
money by starting to poke around at such a driver or card.  It's 
really a chicken-egg situation: nobody will want to muck with driver 
authorship or card production until there are buyers, and there won't 
be any buyers of such "experimental" technology unless it's cheap to 
experiment with, just like the T100P cards are.  Open source is still 
scary to bell-heads, and they will resist until they actually see 
(with their own eyes) a working system that replaces their $100k 
CisNorSiemAvaytelensaco boxes with a $7k PC/card combination.  Even 
then, it's still an uphill battle, but at least it's a battle, 
whereas right now it's a complete non-starter to open one's mouth 
about open source telephony gatewaying at truly large scale 
installations.  And, to be honest, the telco guys are correct at this 
moment.

JT
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Re: [Asterisk-Users] Carrier Access Channel Bank Setup -- No hangup

[Steven Critchfield]

On Thu, 2003-12-04 at 16:36, Jonathan Moore wrote:
> Ok, I contacted the seller about the ring issue. He has offered to replace the
> fxs card in the unit. 
> 
> 1. Is the ring generator on the fxs card or part of the chasis? 
> 2. Can anyone confirm the appropriate jumper settings for connecting analog
> phones to CB?

I think the ring generator is in the chassis. If you pull the chassis
out of the case, you will see a slender board with the T1 interface on
it. It is where the transformer and a few large transistors are.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Port density: DS3 cards?

[John Todd]

On Thu, 4 Dec 2003, John Todd wrote:

 To Steven's comments: Yes, I have considered multiple Asterisk
 devices and I am very aware of de-muxing DS3's into individual T1's
 or PRI's (which bring it's own set of problems, since there is no
 multi-PRI D-channel support in * at the moment)
Ahh.. bugger.. Should I take this to mean that Asterisk on a 4 T1/PRI card
does not support NFAS?
Correct, it does not.  There have been discussions about wanting it, 
but either a) nobody has done the code or submitted the code, or b) 
nobody has paid Digium to create the code.

JT
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Re: [Asterisk-Users] Channelbank Recomendation and GS102 question

[John Todd]

At 8:15 PM -0500 12/4/03, Jim Flagg wrote:
- Original Message -
From: "Walker Haddock" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, December 04, 2003 7:54 PM
Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 question

 We have an installation with 9 inbound voice channels (one is the 
fax) and 768K data.  It is a Hybrid PRI.  It terminates into a
T100P.  It is working great!  The >cost was better than the POTS plus data.

Can I ask what Telephone/Internet service provider you are getting this from?
Does anybody else have a setup like this?


Very interesting.  I've had now two fights with providers (Verizon 
and SBC) who would not offer such a service, claiming that it was 
"impossible" to hybridize a PRI.  I think that's a great offering, 
and of course, it is possible, and especially appealing for Asterisk 
users.

I, too, would be interested in hearing from what vendor you are 
getting such a service.

JT
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Re: [Asterisk-Users] x100p/hangup detection issues?

[Jonathan Moore]

I have the same problem and haven't found a solution. In fact seems like some 
post confirmed that this basically how * works with this hardware setup. 
Someone on IRC today said there are some vmail settings for helping to prevent 
the phatom vm messages.
-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Patrick Cantwell <[EMAIL PROTECTED]>:

> Hi..
> I've got an asterisk setup with an X100P card installed.. I'm noticing
> that upon hangup, it takes a good 3 to 5 seconds before asterisk realizes the
> line has been hung up and drops the call.. this causes my SIP phone to
> continue ringing, and occassional phantom voice mail messages to be left..
> I'm located in good old standard North America, with a regular Verizon
> residential POTS line coming in.  I've checked polarity on the line (easy
> way, reversed it, then the card refused to hang up no matter what), and am
> using kewlstart on the FXO port.  I tried changing to loopstart with no
> change in results. I've poked around some and haven't found anything terribly
> helpful yet, except for another person asking about the same thing :)  Does
> anyone have any ideas?
> Thanks,
> Pat


Visit Winfield Public Schools at http://usd465.com
-
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RE: [Asterisk-Users] Experiences with Fedora 1

[Alex Zarubin]

Title: RE: [Asterisk-Users] Experiences with Fedora 1





Hi Scott,


Could you, please, help with the info on Fedora enhancements (as compared to RH9)
for Xeon multi-threading (hyperthreading?). Nothing in Fedora FAQ about it.


Thank you.


Alex Zarubin
Webley Systems, Inc.


-Original Message-
From: Scott Stingel [mailto:[EMAIL PROTECTED]]
Sent: Thursday, December 04, 2003 9:21 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Experiences with Fedora 1



Hi all-


Over the past week or two, I've been trying out asterisk under Fedora 1
Linux (RedHat).  In my setup (single and dual Xeon motherboards), I have so
far had a very good experience in terms of performance.  In doing E1 load
testing, I've found that Fedora handles heavy load much better than RedHat9,
probably because of its better use of the multi-threading capabilities of
the Xeon.


Before I deploy Fedora to customer sites, though, I'm interested in other
people's experience with Fedora.  If you're using Fedora, please tell us:
what's been your experience?


Thanks
Scott Stingel



Scott M. Stingel 
Emerging Voice Technology Inc.
Palo Alto, California and London, England
 
URL:    www.evtmedia.com  


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Re: [Asterisk-Users] Port density: DS3 cards?

[David Boreham]

> I know of OC3 ATM cards for linux, but AFAIK few telcos
> want to do VoATM these days, do you know of an OC3 SONET
> card? I can't find one even for POS...

Hmm...I was thinking Imagestream, but now I look closely
their cards are all ATM. Still, it might be worth talking to someone
there (Jeff for example) to see what they can do. If there's $$
in this market, they'd be a good vendor to go after it.

I suspect that this problem will be addressed by some big
honking box which has an OC-3/Sonet in one side and GigE/VoIP
out the other side. Clearly one could build such a box with Linux,
if the cards existed.



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Re: [Asterisk-Users] Replicating Legacy Phone Behavior

[Jonathan Moore]

See below

-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Nick Bachmann <[EMAIL PROTECTED]>:

> Greg Boehnlein wrote:
> 
> > First and foremost, these Key System installers are big believers 
> >in VoIP and convergence technologies. While the KSU vendors may see 
> 
> This has been my experiance as well.  Everybody but PBX vendors like 
> VoIP.  The KSU people like it because it gives them more work and job 
> security, but the vendors don't like protocols they don't allow them to 
> have lock-in.  After all, I can use a mix of phones on my * PBX, but if 
> I want a digital phone on my Fujitsu system, I've got to get a Fuji 
> phone.  Phones can be like razor blades for Norelco... it's the product 
> that keeps on selling itself!
> 
> >Asterisk as competition, the installers on the ground see it as an 
> >excellent addition to help connect remote offices and workers together, 
> >but they are driven by the needs of their customers, most of whom want to 
> >KISS (Keep It Simple, Stupid). I.E. they want an Asterisk based VoIP 
> >solution to work in a similar manner to their existing PBX or Phone 
> >System
>  >
> > As a result, these are some of the questions that they threw at me 
> >that I am trying to figure out:
> >
> >1. Legacy KSU and PBX users are used to seeing blinking lights on their 
> >phone that indicate outside lines in use, call on hold, voice mail 
> >waiting, do not disturb etc.. Is it possible to have these features using 
> >SIP phones on the dekstop? I.E. if a user puts a caller on hold at one 
> >extension, can it blink a light on all extensions so that user can be 
> >picked up at another extension? This gets into issues regarding 
> >re-training people with new phones etc.. Kind of like the issue of "I 
> >don't want to press enter to make a call.. Why can't this phone just work 
> >like my old analog phone?"
> >  
> >
> Having a DSS (the blinking lights for each extension, short for "Digital
> Station Selector") is a feature that I wish Asterisk had.  A week or so
> ago there was discussion about a new Windows-based Asterisk application
> (Asterisk Call Manager for Windows?) and it was said that in a later
> version there was a plan to add a "Console mode" (the name for the
> Uberphone that the DSS attaches to).  If I weren't so swamped, I'd ofter
> to help out :-).
> 
> Figuring out which extensions were busy would be easy with the Manager
> API, but I'm not sure how you could forward incoming calls bound for
> another extension.  I guess if it were easy enough, I guess I could mke 
> a Javax/Swing app to do it.
> 
> If there's already an app that does this, I haven't see it, but I'd love to!
> 

I think this is a hard one to deal with because you are going to need support 
for this on both the phone and in *. I am really know expert but I think the 
only phones on the market that might be turned into this are some of the 
programmable ones, maybe the 7690 or the Pingtel by programming the "lights" on 
the lcd. Pingtels look to have a nice display on them and are supposed to be 
programmable with Java. The problem I see with this is that these are all high 
end fairly expensive phones. The snom 200 may have some hooks for this, but are 
limited to the 5 programmable buttons.

The more I think about it the more I think the 7960 might be doable, since I 
know my Cisco sales rep was trying to sell me on DSS.

I also remember reading some references to SIP protocol updates that might 
include some of these types of features (pageing also)

> >2. How does one go about creating call queues and advanced features such 
> >as UCD and ACD using Asterisk?
> >
> Take a look at http://www.voip-info.org/wiki-Asterisk+config+queues.conf
> and the pages it references.
> 
> >3. Is it possible to do Phone to Phone paging with SIP phones? This is a 
> >feature that I personally use a lot on my Legacy Phone System. I simply 
> >hit the extension of the persion I want to chat, and it beeps their phone 
> >and we can talk. Sort of like an Intercom system.
> >  
> 
> That would be a phone feature... I think some of the Cisco phones do 
> it... it's billed as "AutoAnswer" I think.
> 

I have been looking at this angle too. I think the trick is to find a phone 
that is a "multi line" VoiP model and allows per line configuration of auto-
answer. I was thinking of using a pattern where even extensions are for ringing 
the phone and odd numbers are for the intercom/pageing. Candidate phones that 
may be able to do this that I am researching include snoms, cisco 7960, and 
swiss something (mcgp based phone). I have mostly been looking at the low end 
of the phone market, so there may be many others at the high end I am not aware 
of.


> Nick
> 
> P.S. The Asterisk-users lists are searchable if you want to check if 
> your question has been answered before you post.  Also, the 
> voip-info.org Wiki is very informative.
> 
> __

Re: [Asterisk-Users] Port density: DS3 cards?

[Bob Knight]

Steve Dolloff wrote:

I would be seriously wary of putting a DS3's worth of voice traffic on a
TNT.  I don't believe they are rated to handle that much voice.  The
APX1000 would be a much better platform, but I don't know if you can
find one used.
Stephen 
 

Skip the TNT's.  They are really a joke.
I will admit, I am a bitter X-Livingston employee.
First Lucent bought us for our cool gear, then they bought
Ascend for sales and marketing..
I still can't believe they kept the TNT alive and killed PM4.
 

-Original Message-
From: Ernest W. Lessenger [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 4:51 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Port density: DS3 cards?
At 02:34 PM 12/4/2003, you wrote:
   

However, considering the traffic volumes that you are talking about,
 

is
 

it
   

really true to say that the traditional telco cards are
 

astronomically
 

priced, given the amount of revenue that can be generated per month
 

on a
 

DS3?
 

Eight quad-span T-1 cards from Digium: $8,970
Three reasonable-quality asterisk servers: $1,000
One T-1/DS-3 MUX: $5000
Total system cost: $14,970

That actually sounds quite reasonable to me. However, if I were doing
   

this
 

myself I would look hard at getting a MAX TNT with VoIP capability off
eBay. The price would be equivalent or less, the interface would be
   

more
 

complicated, but all the DSP would be done by the MAX.

--Ernest

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--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
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Re: [Asterisk-Users] Asterisk and Avaya IP phones

[Dan Tucny]

On Thu, 2003-12-04 at 23:02, Ed Rubright wrote:
> The company I work for has deployed an Avaya IP phone system.  They
> have deployed the Avaya 4602 and 4620 IP telephones.  They might be
> sending me one of these phones for use in my home office.
> 
> Question: Can I make this IP telephone register and work with my
> Asterisk server?  I don't know if it is a SIP phone?  I searched thru
> the Avaya site, but can't find whether itʼs a SIP phone or not. 
> Thought maybe someone on this list would know.
> 

It's not SIP, currently, well over a year ago Avaya demonstrated SIP
functionality in both the phones and in their Multivantage PBX software.
This has not been released, apparently due to lack of business demand...
I've tried making one work as is with asterisk and a number of other
h323 products, however, I've not yet had any success... Avaya seem to
have really filled these with lots of proprietary hacks... 

> Question: Would I be able to register my Asterisk server or an
> individual SIP phone (Cisco 7960 or Polycom IP600) with the Avaya
> server these 46xx IP telephones use?  I don't know what model of the
> Avaya server the company has purchased, so I have limited info here.
> 

You would not be able to get a SIP phone talking to the Avaya PBX for
the reasons mentioned above... You could possibly get an h323 trunk
between the * and the Avaya PBX, but, I have not tried this yet so this
too may not work... Also, something worth mentioning, the number of IP
trunks you can have is limited by the number the Avaya PBX is licenced
to use... If your company is not currently using trunks, there may not
be any trunks available to use...

Not a lot of good news there, but I hope it's helpful to you...

Dan


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Re: [Asterisk-Users] Channelbank Recomendation and GS102 question

[Jim Flagg]

- Original Message - 
From: "Walker Haddock" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, December 04, 2003 7:54 PM
Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 question


> We have an installation with 9 inbound voice channels (one is the fax) and 768K 
> data.  It is a Hybrid PRI.  It terminates into a
T100P.  It is working great!  The >cost was better than the POTS plus data.

Can I ask what Telephone/Internet service provider you are getting this from?
Does anybody else have a setup like this?

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[Asterisk-Users] x100p/hangup detection issues?

[Patrick Cantwell]

Hi..
    I've got an asterisk setup with 
an X100P card installed.. I'm noticing that upon hangup, it takes a good 3 to 5 
seconds before asterisk realizes the line has been hung up and drops the call.. 
this causes my SIP phone to continue ringing, and occassional phantom voice mail 
messages to be left.. I'm located in good old standard North America, with a 
regular Verizon residential POTS line coming in.  I've checked polarity on 
the line (easy way, reversed it, then the card refused to hang up no matter 
what), and am using kewlstart on the FXO port.  I tried changing to 
loopstart with no change in results. I've poked around some and haven't 
found anything terribly helpful yet, except for another person asking about the 
same thing :)  Does anyone have any ideas?
Thanks,
Pat

Re: [Asterisk-Users] Channelbank Recomendation and GS102 question

[Jonathan Moore]

I looked at the same idea, but since we are not in a major metro area we 
couldn't get a mixed data/voice circuit. If a local number isn't necessary (say 
for 800# inbound) you could also go with a VoIP provider like Vonage for PSTN 
connectivity which would also eleminate the need for a channel bank (or even 
the T1 card)
-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Walker Haddock <[EMAIL PROTECTED]>:

> On Thu, Dec 04, 2003 at 06:26:49PM -0600, Paul Oster wrote:
> > Hi All.
> > 
> > I'm working on an * configuration.  We require 8 inbound POTS lines, and 
> > CT1 or PRI seems like it will be
> 
> We have an installation with 9 inbound voice channels (one is the fax) and
> 768K data.  It is a Hybrid PRI.  It terminates into a T100P.  It is working
> great!  The cost was better than the POTS plus data.
> 
> > quite expensive at that level.  I've read that a T1 Channelbank plus 
> > the  T100P would be a (the?) way to go
> > for this situation.  What is the recommended channelbank for use in this 
> > scenario?  From searching the archives
> > I see a lot of suggestions to get "a channelbank" from ebay.  I would 
> > prefer to be able to use new products
> > so I can easily duplicate the setup for other branch offices in my
> company.
>  
> We're not using a channel bank.  I have one port on a TDM card for the fax
> machine.
> 
> > 
> > My second question relatees to the Grandstream phones.  When they are a 
> > member of a queue group, I get a loud
> > annoying ring in the handset when its in use and another call comes in 
> > on the queue.
> > 
> > Is there a way to enforce 1 call per phone in sip.conf?  Either that or 
> > a way to tell the GS102 to return busy when
> > * trys to send them a call.
> 
> You need to use the latest CVS, it includes the work done by Paul Lieu.  It
> works great.  You just configure your sip.conf according to the notes in the
> bug report:
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=408
> 
> It solved my GS BT102 call waiting ring in ear problem!
> 
> Walker
> > 
> > Thanks in advance.
> > 
> > Paul M. Oster
> > 
> > 
> > 
> > 
> > 
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> 
> -- 
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> Walker Haddock   http://www.datacrest.com
> DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
> 1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
> Birmingham, AL 35216  fax:  1-205-823-7838
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Re: [Asterisk-Users] correct way for cvs update?

[Brian West]

You don't need zapata anymore...

'make update' works in asterisk directory.

bkw

On Thu, 4 Dec 2003, William Waites wrote:

> On Thu, Dec 04, 2003 at 02:20:20PM -0600, Rich Adamson wrote:
> > What's the correct way to do cvs update now?
> >
> > 'cvs update' seems to work in the asterisk directory, but not the zapata
> > or other source directories.
>
> I use 'cvs update -PAd'
>
> AFAIK it should work in the zapata and libpri directories...
>
> -w
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Re: [Asterisk-Users] Channelbank Recomendation and GS102 question

[Jonathan Moore]

I just went through this cycle. I did go ebay and went with a Carrier Access 
Access Bank I, since I still haven't decided if this will be our production 
system and I didn't want to invest too much to find out. After fighting with it 
all day today, I would not recommend the AB1 for this application -- both 
inbound and outbound calling, because the AB1 doesn't support call supervision 
on the inbound fxo ports. And their tech support says that Carrier Access does 
not sell a model with call sup. There are various work arounds, but I think not 
matter what it would be more functional to have a channel bank that supports 
call sup. I also think there are issues with some CBs and caller ID. If you 
want caller ID make sure to ask that it is supported and with which cards.

Adtran seems to be the most recommended vendor on the list. Many refs to TA 
750s.
-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Paul Oster <[EMAIL PROTECTED]>:

> Hi All.
> 
> I'm working on an * configuration.  We require 8 inbound POTS lines, and 
> CT1 or PRI seems like it will be
> quite expensive at that level.  I've read that a T1 Channelbank plus 
> the  T100P would be a (the?) way to go
> for this situation.  What is the recommended channelbank for use in this 
> scenario?  From searching the archives
> I see a lot of suggestions to get "a channelbank" from ebay.  I would 
> prefer to be able to use new products
> so I can easily duplicate the setup for other branch offices in my company.
> 
> My second question relatees to the Grandstream phones.  When they are a 
> member of a queue group, I get a loud
> annoying ring in the handset when its in use and another call comes in 
> on the queue.
> 
> Is there a way to enforce 1 call per phone in sip.conf?  Either that or 
> a way to tell the GS102 to return busy when
> * trys to send them a call.
> 
> Thanks in advance.
> 
> Paul M. Oster
> 
> 
> 
> 
> 
> Free 20MB Web Site Hosting and Personalized E-mail Service!
> Get It Now At Doteasy.com http://www.doteasy.com/et/
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> 


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Re: [Asterisk-Users] Channelbank Recomendation and GS102 question

[Walker Haddock]

On Thu, Dec 04, 2003 at 06:26:49PM -0600, Paul Oster wrote:
> Hi All.
> 
> I'm working on an * configuration.  We require 8 inbound POTS lines, and 
> CT1 or PRI seems like it will be

We have an installation with 9 inbound voice channels (one is the fax) and 768K data.  
It is a Hybrid PRI.  It terminates into a T100P.  It is working great!  The cost was 
better than the POTS plus data.

> quite expensive at that level.  I've read that a T1 Channelbank plus 
> the  T100P would be a (the?) way to go
> for this situation.  What is the recommended channelbank for use in this 
> scenario?  From searching the archives
> I see a lot of suggestions to get "a channelbank" from ebay.  I would 
> prefer to be able to use new products
> so I can easily duplicate the setup for other branch offices in my company.
 
We're not using a channel bank.  I have one port on a TDM card for the fax machine.

> 
> My second question relatees to the Grandstream phones.  When they are a 
> member of a queue group, I get a loud
> annoying ring in the handset when its in use and another call comes in 
> on the queue.
> 
> Is there a way to enforce 1 call per phone in sip.conf?  Either that or 
> a way to tell the GS102 to return busy when
> * trys to send them a call.

You need to use the latest CVS, it includes the work done by Paul Lieu.  It works 
great.  You just configure your sip.conf according to the notes in the bug report:

http://bugs.digium.com/bug_view_page.php?bug_id=408

It solved my GS BT102 call waiting ring in ear problem!

Walker
> 
> Thanks in advance.
> 
> Paul M. Oster
> 
> 
> 
> 
> 
> Free 20MB Web Site Hosting and Personalized E-mail Service!
> Get It Now At Doteasy.com http://www.doteasy.com/et/
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DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
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RE: [Asterisk-Users] Asterisk and Avaya IP phones

[Andy Hester]

Title: Asterisk and Avaya IP phones




  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Ed 
  RubrightSent: Thursday, December 04, 2003 5:03 PMTo: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk 
  and Avaya IP phones
  The company I work for has deployed an Avaya IP 
  phone system.  They have deployed the Avaya 4602 and 4620 IP 
  telephones.  They might be sending me one of these phones for use in my 
  home office.
  Question: Can I make this IP telephone register and 
  work with my Asterisk server?  I don't know if it is a SIP phone?  I 
  searched thru the Avaya site, but can't find whether it’s a SIP phone or 
  not.  Thought maybe someone on this list would know.
  Question: Would I be able to register my Asterisk 
  server or an individual SIP phone (Cisco 7960 or Polycom IP600) with the Avaya 
  server these 46xx IP telephones use?  I don't know what model of the 
  Avaya server the company has purchased, so I have limited info 
here.
  Thanks in advance, = Ed 
  Rubright  
   
  These 
  phones are H.323 phones from what I remember of the documentation, so they are 
  compatible in a VERY general sense.  However, I'm sure that there 
  are alot of proprietary things that the phone does that would have to be 
  sniffed out in order to make it fully compatible if that is even 
  possible.  I wouldn't expect any help from Avaya either. 
  :)
   
  HTH
  Andy 

[Asterisk-Users] Channelbank Recomendation and GS102 question

[Paul Oster]

Hi All.

I'm working on an * configuration.  We require 8 inbound POTS lines, and 
CT1 or PRI seems like it will be
quite expensive at that level.  I've read that a T1 Channelbank plus 
the  T100P would be a (the?) way to go
for this situation.  What is the recommended channelbank for use in this 
scenario?  From searching the archives
I see a lot of suggestions to get "a channelbank" from ebay.  I would 
prefer to be able to use new products
so I can easily duplicate the setup for other branch offices in my company.

My second question relatees to the Grandstream phones.  When they are a 
member of a queue group, I get a loud
annoying ring in the handset when its in use and another call comes in 
on the queue.

Is there a way to enforce 1 call per phone in sip.conf?  Either that or 
a way to tell the GS102 to return busy when
* trys to send them a call.

Thanks in advance.

Paul M. Oster




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Re: [Asterisk-Users] Port density: DS3 cards?

[Greg Boehnlein]

On Thu, 4 Dec 2003, John Todd wrote:

> To Steven's comments: Yes, I have considered multiple Asterisk 
> devices and I am very aware of de-muxing DS3's into individual T1's 
> or PRI's (which bring it's own set of problems, since there is no 
> multi-PRI D-channel support in * at the moment)

Ahh.. bugger.. Should I take this to mean that Asterisk on a 4 T1/PRI card 
does not support NFAS?

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Re: [Asterisk-Users] Port density: DS3 cards?

[Steve Underwood]

John Todd wrote:

Obviously, there are no DS3 TDM cards that are currently compatible 
with Zap channels.  (or are there?)

Does anyone know of an inexpensive DS3 card that could perhaps be used 
with Asterisk if one were to try to port the Zap drivers to such a 
card?  PCI, of course, would be the bus of choice.

I think there are quite a few discouraging comments to be made on that 
question.  Firstly, most companies that produce telecom hardware have 
silly overhead, and thus the price of their cards is astronomical.  
Secondly, most companies that produce telecom hardware are of the 
opinion that transcoding (compression) should be done via DSP's, which 
inflates the cost of the card significantly.  Thirdly, most telecom 
hardware vendors would not consider allowing their drivers into the 
public domain if such development were to happen. I've talked to some 
parties (you know who you are) who have expressed some interest in 
building this type of interface, but a situation where I can actually 
put my hands on equipment is far better than speculative interest by 
those who have not even decided to go forward with design, no matter 
how interesting the end product sounds on the whiteboard.

However, regardless of all these negatives, I'm interested in any 
vendors anyone can offer as a starting point.

Please, don't pester me with comments like "Why do you need 28 PRI's?" 
or "You'll never use that much capacity."  Assume that I actually DO 
have that volume of traffic, and assume there are several dozen other 
people on this list (lurkers and active people) who have the same 
requirements, and assume there are hundreds more people out there who 
have the requirement but haven't considered Asterisk because DS3 isn't 
an option.
I think this is a worthwhile thing to investigate. Does anyone here have 
experience with higher order cards under Linux? Which ones work well, 
and have solid drivers? Although the driver would (probably) need to be 
heavily modified if it is currently a data, rather than telephony, 
oriented driver, a good existing driver should save a lot of work. A DS3 
is well within a PCI channel's capacity (the sum of the two directions 
is less than a 100mb Ethernet, although the DS3 is continuous), but it 
is quite a lot of data. A suitable card would need an efficient 
interface if this is to work. For the Zaptel environment that would mean 
that it can burst data in 1ms (8 sample chunks), and would need to bus 
master the data into memory in a form that doesn't require masses of 
manipulation by software - e.g. reshuffling out of sequence data. Doing 
that for so many channels might create interesting latency challenges 
:-) Still, if you don't try, you can't start to address these issues, 
and work out a solution. It shouldn't be impractical to make a DS3 to 
TDMoE solution, provided the DS3 hardware is right, and the PC  has no 
quirky throughput issues.

This subject often comes up on the IRC channel. There seem to be a 
number of people interested in higher order links, but it really needs 
some positive action somewhere to kick off a real project.

Regards,
Steve
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Re: [Asterisk-Users] app_queue different behaviour

[Richard Lyman]

Anton Yurchenko wrote:

Michiel Betel wrote:

Anton,

Take a look at the latest version of the patch in:

http://bugs.digium.com/bug_view_advanced_page.php?bug_id=214

It does adds an abiliti to make an announcment to a user once they are 
in queue, but no this behaviour with cheking if all operators are busy 
or not. Thank you

Good luck!
Michiel


Anton Yurchenko wrote:

Hello,

is there a way to make app queue to first try to ring the agents and 
start music on hold only when they are all talking to other callers?
So when the caller calls, and there are free operators he hears 
ringing, and * is not picking up until call is answere, or specified 
timeout.
And if the caller calls , and there are no free operators , some 
message like "please wait for next avalable operator"  and them the 
music on hold start.

thanks

just use multiple queues, with varying timeouts (yes you need to 
add 'one' of the mod's with queuetimeout patch.
example:  call comes in, passed to queue, (applied queuetimeout 
mod), set to 2 seconds (that's enough time to check 'logged in' 
agents, increase for the 'callback' agents), call is shoved back 
into dialplan, playback(please-hold-for-agent), toss in next 
queue, with say a queuetimeout set to 30 (or 60 or MORE) seconds, 
and so on

playing your 'please wait for an available operator' on 
queuetimeout,

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Re: [Asterisk-Users] correct way for cvs update?

[William Waites]

On Thu, Dec 04, 2003 at 02:20:20PM -0600, Rich Adamson wrote:
> What's the correct way to do cvs update now?
> 
> 'cvs update' seems to work in the asterisk directory, but not the zapata
> or other source directories.

I use 'cvs update -PAd' 

AFAIK it should work in the zapata and libpri directories...

-w
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Re: [Asterisk-Users] Port density: DS3 cards?

[William Waites]

On Thu, Dec 04, 2003 at 06:25:16PM -0500, William Waites wrote:
> 
> btw, jason thorpe at nasa has benchmarked gige cards on netbsd/i386
> doing well in excess of 500Mbps so it /is/ possible.
> 

Just another data point:

  We also made measurements in November 2000 from a Pentium III running
  Linux with a Gbit interface at SLAC, via an OC12 (622Mbps) link provided
  by the experimental NTON network from SLAC to Caltech to another Pentium
  III host. Over this link we achieved about 500Mbits/s with a single stream
  and a window size of about 800KBytes or more. The results are shown to the
  right. 

  http://www-iepm.slac.stanford.edu/monitoring/bulk/caltech.html

Note that they are doing the tests with TCP which needs window size tuning
at these speeds. That wouldn't be an issue for IAX2 or TDMoE...

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Re: [Asterisk-Users] Experiences with Fedora 1

[firedude]

I haven't personally switched to Fedora but I did decide to upgrade a lot 
of the packages on my * box from RH9 to Fedora.  I have not spent a lot of 
time monitoring how it has handled the load but it does seem to run quite 
smoothely. After having installed many of the packages to satisfy library 
and other requirements.  I would suggest that anyone who is doing an 
upgrade from RH9, not only upgrade the glibc package but also the 
binutils, gcc, libgcc and other pertinent packages because I experienced 
quite a bit of problem with * stopping and not seeing a particular library 
before I upgraded all of these.

Just my personal experience.

AJ




On Thu, 4 Dec 2003, Scott Stingel wrote:

> Hi all-
> 
> Over the past week or two, I've been trying out asterisk under Fedora 1
> Linux (RedHat).  In my setup (single and dual Xeon motherboards), I have so
> far had a very good experience in terms of performance.  In doing E1 load
> testing, I've found that Fedora handles heavy load much better than RedHat9,
> probably because of its better use of the multi-threading capabilities of
> the Xeon.
> 
> Before I deploy Fedora to customer sites, though, I'm interested in other
> people's experience with Fedora.  If you're using Fedora, please tell us:
> what's been your experience?
> 
> Thanks
> Scott Stingel
> 
> 
> Scott M. Stingel 
> Emerging Voice Technology Inc.
> Palo Alto, California and London, England
>  
> URL:www.evtmedia.com  
> 
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Re: [Asterisk-Users] Port density: DS3 cards?

[William Waites]

On Thu, Dec 04, 2003 at 03:31:06PM -0800, David Boreham wrote:
> There are DS3 (and OC-3) PCI cards available 
> with Linux drivers (for data). Might be worthwhile
> contacting a vendor of those things to see if there's
> a way to suck the TDM voice data 
> off a channelized DS3.

I know of OC3 ATM cards for linux, but AFAIK few telcos
want to do VoATM these days, do you know of an OC3 SONET
card? I can't find one even for POS...

-w

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Re: [Asterisk-Users] Replicating Legacy Phone Behavior

[Nick Bachmann]

Greg Boehnlein wrote:

	First and foremost, these Key System installers are big believers 
in VoIP and convergence technologies. While the KSU vendors may see 
This has been my experiance as well.  Everybody but PBX vendors like 
VoIP.  The KSU people like it because it gives them more work and job 
security, but the vendors don't like protocols they don't allow them to 
have lock-in.  After all, I can use a mix of phones on my * PBX, but if 
I want a digital phone on my Fujitsu system, I've got to get a Fuji 
phone.  Phones can be like razor blades for Norelco... it's the product 
that keeps on selling itself!

Asterisk as competition, the installers on the ground see it as an 
excellent addition to help connect remote offices and workers together, 
but they are driven by the needs of their customers, most of whom want to 
KISS (Keep It Simple, Stupid). I.E. they want an Asterisk based VoIP 
solution to work in a similar manner to their existing PBX or Phone 
System
>
	As a result, these are some of the questions that they threw at me 
that I am trying to figure out:

1. Legacy KSU and PBX users are used to seeing blinking lights on their 
phone that indicate outside lines in use, call on hold, voice mail 
waiting, do not disturb etc.. Is it possible to have these features using 
SIP phones on the dekstop? I.E. if a user puts a caller on hold at one 
extension, can it blink a light on all extensions so that user can be 
picked up at another extension? This gets into issues regarding 
re-training people with new phones etc.. Kind of like the issue of "I 
don't want to press enter to make a call.. Why can't this phone just work 
like my old analog phone?"
 

Having a DSS (the blinking lights for each extension, short for "Digital
Station Selector") is a feature that I wish Asterisk had.  A week or so
ago there was discussion about a new Windows-based Asterisk application
(Asterisk Call Manager for Windows?) and it was said that in a later
version there was a plan to add a "Console mode" (the name for the
Uberphone that the DSS attaches to).  If I weren't so swamped, I'd ofter
to help out :-).
Figuring out which extensions were busy would be easy with the Manager
API, but I'm not sure how you could forward incoming calls bound for
another extension.  I guess if it were easy enough, I guess I could mke 
a Javax/Swing app to do it.

If there's already an app that does this, I haven't see it, but I'd love to!

2. How does one go about creating call queues and advanced features such 
as UCD and ACD using Asterisk?

Take a look at http://www.voip-info.org/wiki-Asterisk+config+queues.conf
and the pages it references.
3. Is it possible to do Phone to Phone paging with SIP phones? This is a 
feature that I personally use a lot on my Legacy Phone System. I simply 
hit the extension of the persion I want to chat, and it beeps their phone 
and we can talk. Sort of like an Intercom system.
 
That would be a phone feature... I think some of the Cisco phones do 
it... it's billed as "AutoAnswer" I think.

Nick

P.S. The Asterisk-users lists are searchable if you want to check if 
your question has been answered before you post.  Also, the 
voip-info.org Wiki is very informative.

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Re: [Asterisk-Users] Carrier Access Channel Bank Setup -- No hangup

[Jonathan Moore]

Ok, I contacted the seller about the ring issue. He has offered to replace the
fxs card in the unit. 

1. Is the ring generator on the fxs card or part of the chasis? 
2. Can anyone confirm the appropriate jumper settings for connecting analog
phones to CB?

-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Steven Critchfield <[EMAIL PROTECTED]>:

> On Thu, 2003-12-04 at 13:42, Jonathan Moore wrote:
> > Quoting Steven Critchfield <[EMAIL PROTECTED]>:
> > 
> > > On Thu, 2003-12-04 at 11:49, Jonathan Moore wrote:
> > > > I just purchased a T100p from digium and a Carrier Access Access Bank
> 1
> > > channel
> > > > bank (12fxs/12fxo). I have the setup partially working thanks to some
> help
> > > from
> > > > IRC. However I still have the following issues I can't seem to resolve
> > > > 
> > > > 1. When calling into the system from the PSTN call hangup is not
> detected.
> > > *
> > > > leaves line in use until it is shutdown.
> > > 
> > > You need to get disconnect supervision on you phone lines. Nothing short
> > > is probably going to work for you.
> > > 
> > 
> > Formerly I was using a single line FXO card from Digium and it worked
> perfectly
> > with kewl start signalling. Does this still mean I need to get disconnect
> > supervision from the telco? Just changing from ls to ks in configs doesn't
> seem
> > to have any affect.
> 
> Where you using the busydetection and I forget what the other one is? If
> so, you can probably turn them on for these channels as well and get it
> to work again.
> 
> > > > 2. When calling an analog phone connected to channel bank the phone
> > > doesn't
> > > > ring. If you are in call and some else calls the extension you get the
> > > call
> > > > waiting tones and a flash works to flip to the new line.
> > > 
> > > Do any of the phone lines ring? Sounds like you have one of the AB1's
> > > that are known to get burnt out ring generators.
> > 
> > Is there a way to definitively test the ring generators on the channel
> bank?
> 
> Outside of verifying the switches and jumpers are right, and then seeing
> if the ringers work when dialed, not that I know of.
> -- 
> Steven Critchfield  <[EMAIL PROTECTED]>
> 
> ___
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> 


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Re: [Asterisk-Users] Port density: DS3 cards?

[David Boreham]

There are DS3 (and OC-3) PCI cards available 
with Linux drivers (for data). Might be worthwhile
contacting a vendor of those things to see if there's
a way to suck the TDM voice data 
off a channelized DS3.


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Re: [Asterisk-Users] Port density: DS3 cards?

[William Waites]

On Thu, Dec 04, 2003 at 04:58:03PM -0600, Steven Critchfield wrote:
> > 
> > a standard 32 bit 33MHz PCI bus has a maximum bandwidth of
> > 133MBps == 1Gbps. a DS3 is 45Mbps. even if you pass the data
> > over the bus 10 times, you're still only using up half the
> > peak bandwidth.
> 
> Thats only if you could get full theoretical speeds. I have friends who
> work on ranked supercomputers that will tell you how far short most
> chipsets fall of the theoretical.

Oh, granted. I would only do this on a reasonably high end PC
with a good chipset.

> Also remember the DS3 speed you
> mention is a one way speed. Voice being bidirectional means that it
> would pass the PCI bus in and some going out. Then if you plan on doing
> any recording, there will be another crossing of the PCI bus to either
> go out the ethernet cable to a drive subsystem that could handle the
> speed, or to a decent SCSI system locally.  

I wouldn't suggest doing that! I would do something like:

  | cluster of asterisks
OC3/DS3 <---> * <---> TDMoE <---> | for recording, vm,
  | voip gateway, etc.

and keep the config on the DS3-TDMoE box as simple as
possible.

ideally the DS3 interface is plugged into a 64bit 66MHz bus as well.

btw, jason thorpe at nasa has benchmarked gige cards on netbsd/i386
doing well in excess of 500Mbps so it /is/ possible.

-w

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RE: [Asterisk-Users] Port density: DS3 cards?

[Asterisk online forums]

Lucent TNT box price is attractive, but based on real experience it is
not very VOIP friendly. You have to consider it. It is hard to
interconnect with Cisco for example. I have no idea about Max
TNT-Asterisk interconnection.
We are using Nextone softswitch  and able to serve clients  and
interconnect via Cisco's to Max TNT only via NExtone, but direct
interconnect Cisco-MaxTNT almost impossible. However, if you are using
TNT's on both terminating/originating ends, then it is extremely great
solution.

Regards,
Alexander




  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
Lessenger
Sent: Thursday, December 04, 2003 5:51 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Port density: DS3 cards?

At 02:34 PM 12/4/2003, you wrote:
>However, considering the traffic volumes that you are talking about, is
it
>really true to say that the traditional telco cards are astronomically
>priced, given the amount of revenue that can be generated per month on
a
>DS3?

Eight quad-span T-1 cards from Digium: $8,970
Three reasonable-quality asterisk servers: $1,000
One T-1/DS-3 MUX: $5000

Total system cost: $14,970

That actually sounds quite reasonable to me. However, if I were doing
this 
myself I would look hard at getting a MAX TNT with VoIP capability off 
eBay. The price would be equivalent or less, the interface would be more

complicated, but all the DSP would be done by the MAX.

--Ernest


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RE: [Asterisk-Users] Port density: DS3 cards?

[Steve Dolloff]

I would be seriously wary of putting a DS3's worth of voice traffic on a
TNT.  I don't believe they are rated to handle that much voice.  The
APX1000 would be a much better platform, but I don't know if you can
find one used.

Stephen 

> -Original Message-
> From: Ernest W. Lessenger [mailto:[EMAIL PROTECTED]
> Sent: Thursday, December 04, 2003 4:51 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Port density: DS3 cards?
> 
> At 02:34 PM 12/4/2003, you wrote:
> >However, considering the traffic volumes that you are talking about,
is
> it
> >really true to say that the traditional telco cards are
astronomically
> >priced, given the amount of revenue that can be generated per month
on a
> >DS3?
> 
> Eight quad-span T-1 cards from Digium: $8,970
> Three reasonable-quality asterisk servers: $1,000
> One T-1/DS-3 MUX: $5000
> 
> Total system cost: $14,970
> 
> That actually sounds quite reasonable to me. However, if I were doing
this
> myself I would look hard at getting a MAX TNT with VoIP capability off
> eBay. The price would be equivalent or less, the interface would be
more
> complicated, but all the DSP would be done by the MAX.
> 
> --Ernest
> 
> 
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Re: [Asterisk-Users] Port density: DS3 cards?

[John Todd]

I am uncertain of PCI bus speed limits - too many conflicting reports 
are wedged into my head.

However, the intent here is to dump calls out via VoIP and not simply 
switch between channels elsewhere on the DS3, so overcoming that 
limitation needs to be addressed (if it exists at all, as a follow-up 
post has countered) or some other non-PCI solution created.  Ideally, 
I'd like to see TDM on DS3 in, IAX2 on ethernet out after some 
minimal call control through a context.

To Steven's comments: Yes, I have considered multiple Asterisk 
devices and I am very aware of de-muxing DS3's into individual T1's 
or PRI's (which bring it's own set of problems, since there is no 
multi-PRI D-channel support in * at the moment) but the primary 
concern is that space, power, and heat are at a premium in the 
circumstances under which I am speculating.  10u is much more than 2u.

I'll note that expensive solutions at this density already exist from 
several vendors, and are better than Asterisk right now at handling 
these types of call volumes and media translations.  However, I pose 
the question to the list to note that there _is_ an interest at these 
sizes, and that for Asterisk to step to the next level (no longer 
just a PBX) that type of support is desired.  One step at a time 
:-)

JT


I don't want to criticize your idea, but you do have to consider certain
points. Starting from (as has already been mentioned) the bandwidth of DS3
is far too much to reasonably shove down the PCI bus without data loss /
excessive overheads. Thus a sensible approach would be one where the card
performs the switching, (H100/H110 or otherwise), leaving the Asterisk unit
to maybe handle signalling and call control only. You could go one further,
and if you require 'voice' resource, to switch that onto the PCI bus as well
for processing.
The way I see this, the best implementation plan would actually be to take a
standard DS3 card with a H110/H100 bus, and then look for a third party card
which could switch timeslots on the H110/H100 bus to the PCI bus. This
composite approach would allow a zero latency switching path, but still
include the flexibility of Asterisk.
However, considering the traffic volumes that you are talking about, is it
really true to say that the traditional telco cards are astronomically
priced, given the amount of revenue that can be generated per month on a
DS3?
Linus

- Original Message -
From: "John Todd" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, December 04, 2003 8:06 PM
Subject: [Asterisk-Users] Port density: DS3 cards?

 Obviously, there are no DS3 TDM cards that are currently compatible
 with Zap channels.  (or are there?)
 Does anyone know of an inexpensive DS3 card that could perhaps be
 used with Asterisk if one were to try to port the Zap drivers to such
 a card?  PCI, of course, would be the bus of choice.
 I think there are quite a few discouraging comments to be made on
 > that question.  Firstly, most companies that produce telecom hardware
 > have silly overhead, and thus the price of their cards is
 > astronomical.  Secondly, most companies that produce telecom hardware
 are of the opinion that transcoding (compression) should be done via
 DSP's, which inflates the cost of the card significantly.  Thirdly,
 most telecom hardware vendors would not consider allowing their
 drivers into the public domain if such development were to happen.
 I've talked to some parties (you know who you are) who have expressed
 some interest in building this type of interface, but a situation
 where I can actually put my hands on equipment is far better than
 speculative interest by those who have not even decided to go forward
 with design, no matter how interesting the end product sounds on the
 whiteboard.
 However, regardless of all these negatives, I'm interested in any
 vendors anyone can offer as a starting point.
 Please, don't pester me with comments like "Why do you need 28
 > PRI's?" or "You'll never use that much capacity."  Assume that I
 > actually DO have that volume of traffic, and assume there are several
 > dozen other people on this list (lurkers and active people) who have
 > the same requirements, and assume there are hundreds more people out
 there who have the requirement but haven't considered Asterisk
 because DS3 isn't an option.
 JT
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Re: [Asterisk-Users] Port density: DS3 cards?

[Adam Hart]

> On Thu, Dec 04, 2003 at 10:34:02PM -, Linus Surguy wrote:
> > I don't want to criticize your idea, but you do have to consider certain
> > points. Starting from (as has already been mentioned) the bandwidth of
DS3
> > is far too much to reasonably shove down the PCI bus without data loss /
> > excessive overheads.
>
> ???
>
> a standard 32 bit 33MHz PCI bus has a maximum bandwidth of
> 133MBps == 1Gbps. a DS3 is 45Mbps. even if you pass the data
> over the bus 10 times, you're still only using up half the
> peak bandwidth.
>
According to his logic, my 100mbit network card shouldn't be working to
capacity :)

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[Asterisk-Users] Asterisk and Avaya IP phones

[Ed Rubright]

Title: Asterisk and Avaya IP phones






The company I work for has deployed an Avaya IP phone system.  They have deployed the Avaya 4602 and 4620 IP telephones.  They might be sending me one of these phones for use in my home office.

Question: Can I make this IP telephone register and work with my Asterisk server?  I don't know if it is a SIP phone?  I searched thru the Avaya site, but can't find whether it’s a SIP phone or not.  Thought maybe someone on this list would know.

Question: Would I be able to register my Asterisk server or an individual SIP phone (Cisco 7960 or Polycom IP600) with the Avaya server these 46xx IP telephones use?  I don't know what model of the Avaya server the company has purchased, so I have limited info here.

Thanks in advance,

=

Ed Rubright

Re: [Asterisk-Users] Port density: DS3 cards?

[Steven Critchfield]

On Thu, 2003-12-04 at 16:52, William Waites wrote:
> On Thu, Dec 04, 2003 at 10:34:02PM -, Linus Surguy wrote:
> > I don't want to criticize your idea, but you do have to consider certain
> > points. Starting from (as has already been mentioned) the bandwidth of DS3
> > is far too much to reasonably shove down the PCI bus without data loss /
> > excessive overheads.
> 
> ??? 
> 
> a standard 32 bit 33MHz PCI bus has a maximum bandwidth of
> 133MBps == 1Gbps. a DS3 is 45Mbps. even if you pass the data
> over the bus 10 times, you're still only using up half the
> peak bandwidth.

Thats only if you could get full theoretical speeds. I have friends who
work on ranked supercomputers that will tell you how far short most
chipsets fall of the theoretical. Also remember the DS3 speed you
mention is a one way speed. Voice being bidirectional means that it
would pass the PCI bus in and some going out. Then if you plan on doing
any recording, there will be another crossing of the PCI bus to either
go out the ethernet cable to a drive subsystem that could handle the
speed, or to a decent SCSI system locally.  
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Port density: DS3 cards?

[Ernest W. Lessenger]

Correcting an idiot-math error (24/4 != 8 and 1000*3 != 1000) ...

At 02:34 PM 12/4/2003, you wrote:
However, considering the traffic volumes that you are talking about, is it
really true to say that the traditional telco cards are astronomically
priced, given the amount of revenue that can be generated per month on a
DS3?
Six quad-span T-1 cards from Digium: $8,970
Three reasonable-quality asterisk servers: $3,000
One T-1/DS-3 MUX: $5000
Total system cost: $16,970

That actually sounds quite reasonable to me. However, if I were doing this 
myself I would look hard at getting a MAX TNT with VoIP capability off 
eBay. The price would be equivalent or less, the interface would be more 
complicated, but all the DSP would be done by the MAX.

--Ernest

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Re: [Asterisk-Users] Port density: DS3 cards?

[Ernest W. Lessenger]

At 02:34 PM 12/4/2003, you wrote:
However, considering the traffic volumes that you are talking about, is it
really true to say that the traditional telco cards are astronomically
priced, given the amount of revenue that can be generated per month on a
DS3?
Eight quad-span T-1 cards from Digium: $8,970
Three reasonable-quality asterisk servers: $1,000
One T-1/DS-3 MUX: $5000
Total system cost: $14,970

That actually sounds quite reasonable to me. However, if I were doing this 
myself I would look hard at getting a MAX TNT with VoIP capability off 
eBay. The price would be equivalent or less, the interface would be more 
complicated, but all the DSP would be done by the MAX.

--Ernest

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Re: [Asterisk-Users] Port density: DS3 cards?

[William Waites]

On Thu, Dec 04, 2003 at 10:34:02PM -, Linus Surguy wrote:
> I don't want to criticize your idea, but you do have to consider certain
> points. Starting from (as has already been mentioned) the bandwidth of DS3
> is far too much to reasonably shove down the PCI bus without data loss /
> excessive overheads.

??? 

a standard 32 bit 33MHz PCI bus has a maximum bandwidth of
133MBps == 1Gbps. a DS3 is 45Mbps. even if you pass the data
over the bus 10 times, you're still only using up half the
peak bandwidth.

-w
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Re: [Asterisk-Users] Port density: DS3 cards?

[William Waites]

On Thu, Dec 04, 2003 at 02:43:40PM -0600, Eric Wieling wrote:
> I believe there are boxes that will take a DS-3 from the Telco and spit 
> out T-1's to your telecom equipment.  Not sure what they are called.

you're thinking of something like the nortel access node express...
doing it this way will also spread the load over multiple asterisk
boxes which may or may not be a good thing depending on the
requirements...

fwiw, you could also take the telco circuit as a SONET OC3 which,
assuming proper engineering, would put you as part of a sonet ring
and give added redundancy. often the telcos will run an OC3 to
the basement anyways and just peel off a DS3 for you... depending
on the facilities...

to add to john's question, what about the possibility of ATM or
TDM OC3 cards for asterisk? at that point you could probably
*build* an access-node-alike out of asterisk...

-w

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[Asterisk-Users] Needed - Asterisk Consulting

[Sean P. Robertson]

A customer contacted us today concerning getting a VoIP to PSTN system with
a few IP Phones setup.  Asterisk should fit his needs. It is not a big job,
but I think that this customer is going to need onsite work.

Please contact me off list if you are an interested reseller in the
Washington, DC area.

Sean

___

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NETXUSA
p. 800-289-6389
f.  864-233-4344  "Ask me about Voice over IP."
http://www.netxusa.com/


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Re: [Asterisk-Users] Port density: DS3 cards?

[Linus Surguy]

I don't want to criticize your idea, but you do have to consider certain
points. Starting from (as has already been mentioned) the bandwidth of DS3
is far too much to reasonably shove down the PCI bus without data loss /
excessive overheads. Thus a sensible approach would be one where the card
performs the switching, (H100/H110 or otherwise), leaving the Asterisk unit
to maybe handle signalling and call control only. You could go one further,
and if you require 'voice' resource, to switch that onto the PCI bus as well
for processing.

The way I see this, the best implementation plan would actually be to take a
standard DS3 card with a H110/H100 bus, and then look for a third party card
which could switch timeslots on the H110/H100 bus to the PCI bus. This
composite approach would allow a zero latency switching path, but still
include the flexibility of Asterisk.

However, considering the traffic volumes that you are talking about, is it
really true to say that the traditional telco cards are astronomically
priced, given the amount of revenue that can be generated per month on a
DS3?

Linus

- Original Message -
From: "John Todd" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, December 04, 2003 8:06 PM
Subject: [Asterisk-Users] Port density: DS3 cards?


>
> Obviously, there are no DS3 TDM cards that are currently compatible
> with Zap channels.  (or are there?)
>
> Does anyone know of an inexpensive DS3 card that could perhaps be
> used with Asterisk if one were to try to port the Zap drivers to such
> a card?  PCI, of course, would be the bus of choice.
>
> I think there are quite a few discouraging comments to be made on
> that question.  Firstly, most companies that produce telecom hardware
> have silly overhead, and thus the price of their cards is
> astronomical.  Secondly, most companies that produce telecom hardware
> are of the opinion that transcoding (compression) should be done via
> DSP's, which inflates the cost of the card significantly.  Thirdly,
> most telecom hardware vendors would not consider allowing their
> drivers into the public domain if such development were to happen.
> I've talked to some parties (you know who you are) who have expressed
> some interest in building this type of interface, but a situation
> where I can actually put my hands on equipment is far better than
> speculative interest by those who have not even decided to go forward
> with design, no matter how interesting the end product sounds on the
> whiteboard.
>
> However, regardless of all these negatives, I'm interested in any
> vendors anyone can offer as a starting point.
>
> Please, don't pester me with comments like "Why do you need 28
> PRI's?" or "You'll never use that much capacity."  Assume that I
> actually DO have that volume of traffic, and assume there are several
> dozen other people on this list (lurkers and active people) who have
> the same requirements, and assume there are hundreds more people out
> there who have the requirement but haven't considered Asterisk
> because DS3 isn't an option.
>
> JT
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>

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Re: [Asterisk-Users] Operating environment for *

[Jonathan Moore]

Don't have answers to your main questions but there is a place "share war
stories." The Asterisk Wiki

http://www.voip-info.org/wiki-Asterisk

Not that many scenarios posted, but a few.
-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Ahmad Faiz <[EMAIL PROTECTED]>:

> Hi all,
> 
> I've got some questions to post in regard to running asterisk in a
> production-grade environment, specifically targeting high-density IVR
> applications. No VoIP involved, just straight PSTN -> * and perhaps the
> occasional outdials or agent-based predictive dialing.
> 
> 1) Which user would you run * under?
> 2) What other security-related issues do you have to resolve?
> 3) How do you handle crashes (murphy -will- visit you some day)?
> 4) What are the best redundancy techniques to use?
> 5) With respect to Digium's E1 card, what's the max # of boards you've been
> able to install in a single box and still have * work well?
> 
> Thanks in advance. Perhaps someone could start a site where users can
> contribute war stories of their * deployment -- that would make for good
> reading!
> 
> Cheers,
> Faiz
> 
> 
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> 


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RE: [Asterisk-Users] Operating environment for *

[Scott Stingel]

Answering question number 5 only:

My customer's system is an extremely busy IVR, used in a game-show call-in
environment, with short calls and high peak call rates.  The maximum number
of ports so far that my system can handle, with a single fast P4 processor,
is 4 E1 spans (one E400P).  Even at this size of 120 channels, I experience
some dropped calls and channel lock-outs (which recover every few minutes).
However, there are three pieces of good news that should improve the
situation:

(1) There apparently is a bug is how the code buffers the frames on the E1
ports - an overflow causes many re-tries and possibly lost calls (in my
experience).  Mark and Martin at Digium are aware of the problem, and are
currently looking into it.  I will let you know what they find.

(2) I've had good luck so far using a dual-Xeon processor board and Redhat's
Fedora Linux.  I have an IVR load-tester and the results are very
encouraging.  In a couple weeks I'll have results from the field.

(3) Digium's TE410P has the ability to be a bus-master.  This produces some
minor performance gains in my tests.

When all three of these things are incorporated into an environment, I think
you could probably expect that the dual-Xeon setup could handle as many as 8
or more E1 spans in a heavy load IVR environment.  

Please let me know if you need my load tester script, as I'm happy to share
it.

Good luck!
Scott

Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED]    
URL:www.evtmedia.com    



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Ahmad Faiz
> Sent: Thursday, December 04, 2003 9:29 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Operating environment for *
> 
> 
> Hi all,
> 
> I've got some questions to post in regard to running asterisk in a
> production-grade environment, specifically targeting high-density IVR
> applications. No VoIP involved, just straight PSTN -> * and 
> perhaps the
> occasional outdials or agent-based predictive dialing.
> 
> 1) Which user would you run * under?
> 2) What other security-related issues do you have to resolve?
> 3) How do you handle crashes (murphy -will- visit you some day)?
> 4) What are the best redundancy techniques to use?
> 5) With respect to Digium's E1 card, what's the max # of 
> boards you've been
> able to install in a single box and still have * work well?
> 
> Thanks in advance. Perhaps someone could start a site where users can
> contribute war stories of their * deployment -- that would 
> make for good
> reading!
> 
> Cheers,
> Faiz
> 
> 
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> 
> 

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Re: [Asterisk-Users] Operating environment for *

[Steven Critchfield]

On Thu, 2003-12-04 at 15:29, Ahmad Faiz wrote:
> Hi all,
> 
> I've got some questions to post in regard to running asterisk in a
> production-grade environment, specifically targeting high-density IVR
> applications. No VoIP involved, just straight PSTN -> * and perhaps the
> occasional outdials or agent-based predictive dialing.
> 
> 1) Which user would you run * under?

While not the best practice, root is normally used. With proper device
and directory permissions, it should be able to run as anyone you wish. 

> 2) What other security-related issues do you have to resolve?

Unload all VoIP modules if you are not using them. Other than that, do
your normal network hardening. 

> 3) How do you handle crashes (murphy -will- visit you some day)?

There is a script that I picked up from Tilghman that runs out of
/etc/init.d that will continually restart asterisk unless it exited with
a return of 0(one of the stop commands). It also sets it up so it dumps
core in a configured directory and can email you that it has done so.  

> 4) What are the best redundancy techniques to use?

This could be an entire book. It all depends on how you dealing with
your lines and how much redundancy you wish to pay for. 

Someone here recently mentioned T1 devices that could do failover
switching so you could have a hot spare waiting to grab the T1 line in
the case of the asterisk machine going down. But this would require
duplication of hardware for this to work.And this is really the biggest
cost and you still have single point of failures in the actual wire
itself. Of course your telco can provide you backup lines to your
install, but then you incur even more cost and it will be a monthly
cost.

> 5) With respect to Digium's E1 card, what's the max # of boards you've been
> able to install in a single box and still have * work well?

2 cards max regardless of port density right now. 2 quad span cards
should be just fine per machine.

> Thanks in advance. Perhaps someone could start a site where users can
> contribute war stories of their * deployment -- that would make for good
> reading!

www.voip-info.org 

Specifically 
http://www.voip-info.org/wiki-Asterisk+hardware+recommendations
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Carrier Access Channel Bank Setup -- No hangup

[Jonathan Moore]

-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Steven Critchfield <[EMAIL PROTECTED]>:

> On Thu, 2003-12-04 at 13:42, Jonathan Moore wrote:
> > Quoting Steven Critchfield <[EMAIL PROTECTED]>:
> > 
> > > On Thu, 2003-12-04 at 11:49, Jonathan Moore wrote:
> > > > I just purchased a T100p from digium and a Carrier Access Access Bank
> 1
> > > channel
> > > > bank (12fxs/12fxo). I have the setup partially working thanks to some
> help
> > > from
> > > > IRC. However I still have the following issues I can't seem to resolve
> > > > 
> > > > 1. When calling into the system from the PSTN call hangup is not
> detected.
> > > *
> > > > leaves line in use until it is shutdown.
> > > 
> > > You need to get disconnect supervision on you phone lines. Nothing short
> > > is probably going to work for you.
> > > 
> > 
> > Formerly I was using a single line FXO card from Digium and it worked
> perfectly
> > with kewl start signalling. Does this still mean I need to get disconnect
> > supervision from the telco? Just changing from ls to ks in configs doesn't
> seem
> > to have any affect.
> 
> Where you using the busydetection and I forget what the other one is? If
> so, you can probably turn them on for these channels as well and get it
> to work again.

No I wasn't using either before (unless they default) I didn't have either
keyword in the file and it seems to detect hangup better. Only former issue with
single fxo card was, if I dialed through to an extension and hung up before
answer the extension would continue ringing for 5 seconds after hangup. 

With keywords on it now seems to take 30 seconds for the call to clear if I
hangup in main directory. I will have to test ...

Takes about 8 seconds for internal phone to stop ringing. Line doesn't clear for
another 30-60 seconds. Won't this cause major havic in voice mail? Better yet
tieing up lines?

> 
> > > > 2. When calling an analog phone connected to channel bank the phone
> > > doesn't
> > > > ring. If you are in call and some else calls the extension you get the
> > > call
> > > > waiting tones and a flash works to flip to the new line.
> > > 
> > > Do any of the phone lines ring? Sounds like you have one of the AB1's
> > > that are known to get burnt out ring generators.
> > 
> > Is there a way to definitively test the ring generators on the channel
> bank?
> 
> Outside of verifying the switches and jumpers are right, and then seeing
> if the ringers work when dialed, not that I know of.

Anyone know the correct jumper settings for an access bank one. I am going off
of what is posted at

http://www.billheckel.com/asterisk/cac.html
I think this setup assumed 2 fxs cards.

> -- 
> Steven Critchfield  <[EMAIL PROTECTED]>
> 
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[Asterisk-Users] Operating environment for *

[Ahmad Faiz]

Hi all,

I've got some questions to post in regard to running asterisk in a
production-grade environment, specifically targeting high-density IVR
applications. No VoIP involved, just straight PSTN -> * and perhaps the
occasional outdials or agent-based predictive dialing.

1) Which user would you run * under?
2) What other security-related issues do you have to resolve?
3) How do you handle crashes (murphy -will- visit you some day)?
4) What are the best redundancy techniques to use?
5) With respect to Digium's E1 card, what's the max # of boards you've been
able to install in a single box and still have * work well?

Thanks in advance. Perhaps someone could start a site where users can
contribute war stories of their * deployment -- that would make for good
reading!

Cheers,
Faiz


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[Asterisk-Users] is there any way to search the mailing list archive and order results by date?

[Ken Godee]

http://www.mail-archive.com/asterisk-users%40lists.digium.com/index.html

Returns searches in chronological order.

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Re: [Asterisk-Users] Carrier Access Channel Bank Setup -- No hangup

[Steven Critchfield]

On Thu, 2003-12-04 at 13:42, Jonathan Moore wrote:
> Quoting Steven Critchfield <[EMAIL PROTECTED]>:
> 
> > On Thu, 2003-12-04 at 11:49, Jonathan Moore wrote:
> > > I just purchased a T100p from digium and a Carrier Access Access Bank 1
> > channel
> > > bank (12fxs/12fxo). I have the setup partially working thanks to some help
> > from
> > > IRC. However I still have the following issues I can't seem to resolve
> > > 
> > > 1. When calling into the system from the PSTN call hangup is not detected.
> > *
> > > leaves line in use until it is shutdown.
> > 
> > You need to get disconnect supervision on you phone lines. Nothing short
> > is probably going to work for you.
> > 
> 
> Formerly I was using a single line FXO card from Digium and it worked perfectly
> with kewl start signalling. Does this still mean I need to get disconnect
> supervision from the telco? Just changing from ls to ks in configs doesn't seem
> to have any affect.

Where you using the busydetection and I forget what the other one is? If
so, you can probably turn them on for these channels as well and get it
to work again.

> > > 2. When calling an analog phone connected to channel bank the phone
> > doesn't
> > > ring. If you are in call and some else calls the extension you get the
> > call
> > > waiting tones and a flash works to flip to the new line.
> > 
> > Do any of the phone lines ring? Sounds like you have one of the AB1's
> > that are known to get burnt out ring generators.
> 
> Is there a way to definitively test the ring generators on the channel bank?

Outside of verifying the switches and jumpers are right, and then seeing
if the ringers work when dialed, not that I know of.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] Another audio file

[cloos]

If anyone is interested, I've trimmed one of Allison's recordings down
to the single word 'welcome', for use as a generic first message when a
line is answered.

I've put it up at:

http://jhcloos.com/sounds/asterisk/welcome.gsm

and will submit it to bugs.digium.com as well.

-JimC
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Re: [Asterisk-Users] is there any way to search the mailing listarchive and order results by date?

[Steven Critchfield]

On Thu, 2003-12-04 at 14:09, [EMAIL PROTECTED] wrote: 
> > On Thu, 2003-12-04 at 09:30, [EMAIL PROTECTED] wrote:
> >> i use google, with site:digium.com to search the archives, but i've
> >> never
> >> found a way to show the newest messages first, or limit the results to
> >> messages within a date range.  anybody know a better way to search that
> >> allows this?
> >
> > Thats what a personal archive of the list is good for. My personal copy
> > goes back to the end of 2001. I know that doesn't help much.
> >
> > Of course if you know the month and year you are looking for, google
> > should be able to search for those too.


> Bummer.  I have put in different months as keywords a few times trying to
> get more recent messages.  Too bad something like marc.theaimsgroup.com
> hasnt picked up the * lists.  I use it to search the ltsp lists
> frequently.  If someone were to offer to host a publicly searchable
> archive would you be willing to submit your personal archive so that it
> would go back to 2001?

Digiums archive goes back to 1999. 

-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] Male voice over work

[Brian West]

I have been in contact with OnTrack Studios and he male voice work for
asterisk.  If you wish to contact him [EMAIL PROTECTED]

I know someone on the list was looking for a male voice.

Thanks,
Brian

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Re: [Asterisk-Users] Port density: DS3 cards?

[Steven Critchfield]

On Thu, 2003-12-04 at 14:06, John Todd wrote:
> Obviously, there are no DS3 TDM cards that are currently compatible 
> with Zap channels.  (or are there?)
> 
> Does anyone know of an inexpensive DS3 card that could perhaps be 
> used with Asterisk if one were to try to port the Zap drivers to such 
> a card?  PCI, of course, would be the bus of choice.

Your first problem will be bus speeds. A single DS3 is 44.736Mbps. each
way. So if you double this and get the 89.472Mbps, you are going to be
coming close to the real limits of the 32/33mhz PCI bus without having
done any work on the data you are shuffling. So while it could be done
here, I'd start worrying about stability. Sure you could switch up to
faster PCI buses like the 64/66mhz bus, but then you will start limiting
what systems you can use. Then again, if you are putting that many
channels through a single machine, there wasn't many choices for the
hardware to begin with.

My question I guess would come down to why bring a DS3 into a PC when
you could get a multiplexer that took your DS3 and split it down to T1s
so you could use already developed hardware. You could then build in
some redundancy and if a machine goes down and takes a few T1s down with
it, you just route around it till you fix it.

Figure you could go 7 1U super micros with a TE410P in each one will get
you your 28 T1s. Then either make one machine do the work of traffic cop
and connect calls between T1s or point them down the line to other
machines that can then terminate the call. Since the traffic cop machine
wouldn't need to actually service calls but for a short period during
routing, if it were to fail it would just drop any calls it was in the
middle of routing and wouldn't route new calls. In this case you could
have a hot spare waiting in the wings to do a on fail dial here type
route and it could service the new calls. In this case you have 9u of
space used and no machine can take down more than 4 T1s worth of calls.
Ohh, you need to add 1u of space fore the multiplexer. So 10u of space
for a DS3 using currently available technology and software with a bit
of failover support.  
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Port density: DS3 cards?

[Eric Wieling]

I believe there are boxes that will take a DS-3 from the Telco and spit 
out T-1's to your telecom equipment.  Not sure what they are called.

John Todd wrote:

Obviously, there are no DS3 TDM cards that are currently compatible with 
Zap channels.  (or are there?)

Does anyone know of an inexpensive DS3 card that could perhaps be used 
with Asterisk if one were to try to port the Zap drivers to such a 
card?  PCI, of course, would be the bus of choice.

I think there are quite a few discouraging comments to be made on that 
question.  Firstly, most companies that produce telecom hardware have 
silly overhead, and thus the price of their cards is astronomical.  
Secondly, most companies that produce telecom hardware are of the 
opinion that transcoding (compression) should be done via DSP's, which 
inflates the cost of the card significantly.  Thirdly, most telecom 
hardware vendors would not consider allowing their drivers into the 
public domain if such development were to happen. I've talked to some 
parties (you know who you are) who have expressed some interest in 
building this type of interface, but a situation where I can actually 
put my hands on equipment is far better than speculative interest by 
those who have not even decided to go forward with design, no matter how 
interesting the end product sounds on the whiteboard.

However, regardless of all these negatives, I'm interested in any 
vendors anyone can offer as a starting point.

Please, don't pester me with comments like "Why do you need 28 PRI's?" 
or "You'll never use that much capacity."  Assume that I actually DO 
have that volume of traffic, and assume there are several dozen other 
people on this list (lurkers and active people) who have the same 
requirements, and assume there are hundreds more people out there who 
have the requirement but haven't considered Asterisk because DS3 isn't 
an option.

JT
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Re: [Asterisk-Users] Carrier Access Channel Bank Setup -- No hangup

[Jonathan Moore]

Thanks for the help. Can you explain the need to define all the channels in
zapata.conf? I am not connecting devices to all the ports on the CB yet, so if I
place the definitions into my groups 1 and 2 then things seem to be a bit
strange when defining my outbound pstn calling.


Quoting Robert Hajime Lanning <[EMAIL PROTECTED]>:

> Changes are below.  Use KewlStart for the FXO channels.  (Loopstart +
> remote disconnect suppervision)  Define all T1 channels.  FXS channels
> can be loopstart without any issues.
> 
> 
> > I just purchased a T100p from digium and a Carrier Access Access Bank 1
> > channel bank (12fxs/12fxo). I have the setup partially working thanks
> > to some help from IRC. However I still have the following issues I
> > can't seem to resolve
> >
> > 1. When calling into the system from the PSTN call hangup is not
> > detected. * leaves line in use until it is shutdown.
> >
> > 2. When calling an analog phone connected to channel bank the phone
> > doesn't ring. If you are in call and some else calls the extension
> > you get the call waiting tones and a flash works to flip to the new
> > line.
> >
> > zaptel.conf
> > span=1,1,0,esf,b8zs
> > fxsls=13-24
> > fxols=1-12
> 
> use "fxsks=13-24"
> 
> > loadzone = us
> > defaultzone=us
> >
> > zapata.conf
> > [channels]
> >
> > context = local
> > language = en
> > callwaiting = yes
> > threewaycalling = yes
> > transfer = yes
> > cancelforward = yes
> > callreturn = no
> > usecallerid = yes
> > hidecallerid = no
> > echocancel = yes
> > echocancelwhenbridged = yes
> > ;immediate = no
> > txgain=1.0
> > rxgain=1.0
> > callprogress=no
> > busydetect=no
> >
> > group = 2
> >
> > ;use with FXO PCI card
> > signalling = fxo_ls
> > ;channel => 13-24
> > channel => 1
> 
> channel => 1-12
> 
> >
> > context = local
> >
> > group = 1
> > ;use with FXS USB card
> > signalling = fxs_ls
> 
> signalling = fxs_ks
> 
> > ;callerid = "John Doe" <(710) 555-6200>
> > channel => 13
> 
> channel => 13-24
> 
> -- 
> END OF LINE
>-MCP
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Re: [Asterisk-Users] Carrier Access Channel Bank Setup -- No hangup

[Jonathan Moore]

Quoting Steven Critchfield <[EMAIL PROTECTED]>:

> On Thu, 2003-12-04 at 11:49, Jonathan Moore wrote:
> > I just purchased a T100p from digium and a Carrier Access Access Bank 1
> channel
> > bank (12fxs/12fxo). I have the setup partially working thanks to some help
> from
> > IRC. However I still have the following issues I can't seem to resolve
> > 
> > 1. When calling into the system from the PSTN call hangup is not detected.
> *
> > leaves line in use until it is shutdown.
> 
> You need to get disconnect supervision on you phone lines. Nothing short
> is probably going to work for you.
> 

Formerly I was using a single line FXO card from Digium and it worked perfectly
with kewl start signalling. Does this still mean I need to get disconnect
supervision from the telco? Just changing from ls to ks in configs doesn't seem
to have any affect.


> > 2. When calling an analog phone connected to channel bank the phone
> doesn't
> > ring. If you are in call and some else calls the extension you get the
> call
> > waiting tones and a flash works to flip to the new line.
> 
> Do any of the phone lines ring? Sounds like you have one of the AB1's
> that are known to get burnt out ring generators.

Is there a way to definitively test the ring generators on the channel bank?


Thanks for the help.

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RE: [Asterisk-Users] vmail.cgi with Redhat 9.0

[jerk face]

--- "Evan P. Hall" <[EMAIL PROTECTED]> wrote:
> -Original Message-
> From: jerk face [mailto:[EMAIL PROTECTED] 
> Sent: Thursday, December 04, 2003 9:02 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] vmail.cgi with Redhat 9.0
> 
> 
> > I recently switched from Mandrake to Redhat and I
> > noticed that vmail.cgi does not work with the
> default
> > apache installation that comes with Redhat.
> > Here is what I get in my error logs:
> > 
> > [Thu Dec 04 11:59:57 2003] [notice] suEXEC
> mechanism
> > enabled (wrapper: /usr/sbin/suexec)
> > [Thu Dec 04 11:59:58 2003] [notice] Digest:
> generating
> > secret for digest authentication ...
> > [Thu Dec 04 11:59:58 2003] [notice] Digest: done
> > [Thu Dec 04 11:59:59 2003] [notice] Apache/2.0.40
> (Red
> > Hat Linux) configured -- resuming normal
> operations
> > [Thu Dec 04 12:00:08 2003] [error] [client
> > 192.168.10.12] Directory index forbidden by rule:
> > /var/www/html/
> > [Thu Dec 04 12:00:15 2003] [error] [client
> > 192.168.10.12] Premature end of script headers:
> > vmail.cgi
> > [Thu Dec 04 12:00:15 2003] [error] [client
> > 192.168.10.12] Can't do setuid
> > 
> > 
> > Does anybody know how I could fix this problem?
> > 
> > Thank you for your time.
> 
> You need to install the perl 'perl-suidperl' rpm
> package to enable
> execution of suid perl scripts.
> 
> -Evan


Problem Solved.

The steps I took to solve the problem:
Install perl-suidperl rpm
make webvmail in the asterisk directory

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[Asterisk-Users] correct way for cvs update?

[Rich Adamson]

What's the correct way to do cvs update now?

'cvs update' seems to work in the asterisk directory, but not the zapata
or other source directories.



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RE: [Asterisk-Users] vmail.cgi with Redhat 9.0

[Wade J. Weppler]

>>I've patched app_voicemail.c to
>>create everything as 777.  As this is a 
>>dedicated Asterisk box, I don't see
>>the harm in giving everyone on the 
>>system full access.

>Would you be willing to share this patch? 

It was a simple patch.  Just search for 0700 in app_voicemail.c and
change them all to 0777

-wade

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Re: [Asterisk-Users] is there any way to search the mailing listarchive and order results by date?

[listbox]

> On Thu, 2003-12-04 at 09:30, [EMAIL PROTECTED] wrote:
>> i use google, with site:digium.com to search the archives, but i've
>> never
>> found a way to show the newest messages first, or limit the results to
>> messages within a date range.  anybody know a better way to search that
>> allows this?
>
> Thats what a personal archive of the list is good for. My personal copy
> goes back to the end of 2001. I know that doesn't help much.
>
> Of course if you know the month and year you are looking for, google
> should be able to search for those too.
> --
> Steven Critchfield  <[EMAIL PROTECTED]>
>
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>
>
Bummer.  I have put in different months as keywords a few times trying to
get more recent messages.  Too bad something like marc.theaimsgroup.com
hasnt picked up the * lists.  I use it to search the ltsp lists
frequently.  If someone were to offer to host a publicly searchable
archive would you be willing to submit your personal archive so that it
would go back to 2001?

-jeff
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[Asterisk-Users] Port density: DS3 cards?

[John Todd]

Obviously, there are no DS3 TDM cards that are currently compatible 
with Zap channels.  (or are there?)

Does anyone know of an inexpensive DS3 card that could perhaps be 
used with Asterisk if one were to try to port the Zap drivers to such 
a card?  PCI, of course, would be the bus of choice.

I think there are quite a few discouraging comments to be made on 
that question.  Firstly, most companies that produce telecom hardware 
have silly overhead, and thus the price of their cards is 
astronomical.  Secondly, most companies that produce telecom hardware 
are of the opinion that transcoding (compression) should be done via 
DSP's, which inflates the cost of the card significantly.  Thirdly, 
most telecom hardware vendors would not consider allowing their 
drivers into the public domain if such development were to happen. 
I've talked to some parties (you know who you are) who have expressed 
some interest in building this type of interface, but a situation 
where I can actually put my hands on equipment is far better than 
speculative interest by those who have not even decided to go forward 
with design, no matter how interesting the end product sounds on the 
whiteboard.

However, regardless of all these negatives, I'm interested in any 
vendors anyone can offer as a starting point.

Please, don't pester me with comments like "Why do you need 28 
PRI's?" or "You'll never use that much capacity."  Assume that I 
actually DO have that volume of traffic, and assume there are several 
dozen other people on this list (lurkers and active people) who have 
the same requirements, and assume there are hundreds more people out 
there who have the requirement but haven't considered Asterisk 
because DS3 isn't an option.

JT
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Re: [Asterisk-Users] RE: voicemail file permissions

[CW_ASN - Gus]

Guys, I'm using RH9 with vmail.cgi without any modifications... I'm just do
a 'make webvmail' after 'make install'... I don't have any troubles...

Regards,

Gus

- Original Message -
From: "Carlton J. O'Riley" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, December 04, 2003 4:23 PM
Subject: [Asterisk-Users] RE: voicemail file permissions


Here is a script I use in a cron job that runs every 5 minutes to make it so
that my webserver (which runs as the apache group) can access the voicemails
through the web.  Seems to fix my problems.  Although if I get the email
there is a voicemail it might be 5 minutes before I can get to it via the
web, but you could increase the frequency at which this job runs.  Five
minutes has been fine for me.  It'd be nice to be able to set the owner and
group and permissions for voicemail files in the configuration file for
voicemail.  If I had time I'd probably do it myself.

Carlton

#!/bin/sh
/bin/chgrp -R apache /var/spool/asterisk/vm/*
/bin/chmod -R g+rw /var/spool/asterisk/vm/*

>  hi, i realised that when voicemails are recorded it is set to 700 file =
permission and which leads to a serrious problem when
>accessing the = voicemail thru the web using vmail,cgi
>
>  how can i automatically set the file permission to 755 or 777 so that = i
can make it readeable from the web? which file in * helps
> to record = the voicemail and create that voicemail in a certain dir?? if
any onw = knows, i can perhaps find that line and change as > nesseciate.
>
>  anyone tried vmail.cgi could help.
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Re: [Asterisk-Users] John Brown from Chagres!

[Steven Critchfield]

While I have not had any dealings with this company, I really enjoy that
you can have open and transparent dealings here.  I have seen this of
one or two other companies on other lists and find it refreshing to hear
about how companies are doing especially when they help the community.

On Wed, 2003-12-03 at 20:44, John Brown (CV) wrote:
> Several  things conspired to muck things up the last 3-4 weeks.
> 1. Surgery (repair of a previous hernia)
> 2. Travel to work at opening our EU warehouse
> 3. TSA dropping my laptop, thus breaking my access to our VPN
> 4. New PRI going to a Asterisk box for our PBX and having the PRI be
>mucked up. Qwest dorking the number port.
> 5. 2 employees have a stressed life (they are 20 something
>and well, life is stressful for them) deciding not to really
>do the work they where suppose to.  ergo  cats away, mice play.
>We fired the kiddies.
> 
> My other biz partner will be spending more time at Chagres now that
> he has sold his other company.  He will handle operations, I'll
> handle sales and biz-dev.
> 
> Two new employees start on Monday that will handle orders and
> customer calls.
> 
> Inventory enroute from Grandstream, which will resolve all backorders.
> We will have stock of BT-101's BT-102's.  HT-286s stock levels will
> be raised next week and we will have those as well.
> 
> Chagres is alive and going well.  We will have inventory
> of all GS product next week, most Digium product (T100P on
> 2 week delay from Digium), and maybe SIPURA. 
> 
> 
> Good news is that we now have a Euro warehouse and starting in early
> January will ship Euro orders from Rotterdam.  This will save
> our Euro customers much in shipping costs and transit time.
> 
> I want to thank everyone for putting up with the mad couple of
> weeks, but things are shaped up and I think we can move forward.
> 
> If anyone needs me urgently, my direct line is 505 998 0567
> If I don't answer please leave a short but *clear* voice mail.  
> I do check this voice mailbox several times per day.
> 
> 
> john brown
> 
> fwd: 50870
> direct: +1 505 998 0567
> office: +1 505 830 1200
> fax   : +1 505 830 1201
> 
> 
> On Tue, Dec 02, 2003 at 06:22:27PM -0600, Brian West wrote:
> > I just talked to him lastnight... He was out of the office for a week or
> > so.  He got back and had to fire a few people for not doing their jobs..
> > and that he is slowly but surely getting caught up and that QWest
> > screwed up their number porting.  They moved their numbers from QWest to
> > anohter provider and they aren't working... as of lastnight he was about
> > to smack Qwest! :P
> > 
> > Just an FYI
> > 
> > bkw
> > 
> > On Wed, 3 Dec 2003, Aaron Martin wrote:
> > 
> > > Sorry to everyone on the list, but for some reason this is the only reliable way 
> > > to get hold of John.
> > >
> > > John Brown of Chagres Technologies, please contact me!  I have been trying for 
> > > weeks now to get hold of you via email and phone after wire transfering money 
> > > into your account for the Grandstream phones we ordered, but so far I have not 
> > > had a single response, nor have the phones arrived!
> > >
> > > Please contact me ASAP
> > >
> > > Aaron Martin
> > > Comtek Computing Solutions Ltd.
> > >
> > >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> ___
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Re: [Asterisk-Users] Carrier Access Channel Bank Setup -- No hangup

[Steven Critchfield]

On Thu, 2003-12-04 at 11:49, Jonathan Moore wrote:
> I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel
> bank (12fxs/12fxo). I have the setup partially working thanks to some help from
> IRC. However I still have the following issues I can't seem to resolve
> 
> 1. When calling into the system from the PSTN call hangup is not detected. *
> leaves line in use until it is shutdown.

You need to get disconnect supervision on you phone lines. Nothing short
is probably going to work for you.

> 2. When calling an analog phone connected to channel bank the phone doesn't
> ring. If you are in call and some else calls the extension you get the call
> waiting tones and a flash works to flip to the new line.

Do any of the phone lines ring? Sounds like you have one of the AB1's
that are known to get burnt out ring generators.

> zaptel.conf
> span=1,1,0,esf,b8zs
> fxsls=13-24
> fxols=1-12

Switch these from ls to ks for kewlstart, or disconnect supervision
signaling.

> loadzone = us
> defaultzone=us
> 
> zapata.conf
> [channels]
> 
> context = local
> language = en
> callwaiting = yes
> threewaycalling = yes
> transfer = yes
> cancelforward = yes
> callreturn = no
> usecallerid = yes
> hidecallerid = no
> echocancel = yes
> echocancelwhenbridged = yes
> ;immediate = no
> txgain=1.0
> rxgain=1.0
> callprogress=no
> busydetect=no
> 
> group = 2
> 
> ;use with FXO PCI card
> signalling = fxo_ls

If you change above to ks, do the same here.

> ;channel => 13-24
> channel => 1
> 
> context = local
> 
> group = 1
> ;use with FXS USB card
> signalling = fxs_ls

again, if you change to ks, do so here also.

> ;callerid = "John Doe" <(710) 555-6200>
> channel => 13
-- 
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Re: [Asterisk-Users] Carrier Access Channel Bank Setup -- No hangup

[Robert Hajime Lanning]

Changes are below.  Use KewlStart for the FXO channels.  (Loopstart +
remote disconnect suppervision)  Define all T1 channels.  FXS channels
can be loopstart without any issues.


> I just purchased a T100p from digium and a Carrier Access Access Bank 1
> channel bank (12fxs/12fxo). I have the setup partially working thanks
> to some help from IRC. However I still have the following issues I
> can't seem to resolve
>
> 1. When calling into the system from the PSTN call hangup is not
> detected. * leaves line in use until it is shutdown.
>
> 2. When calling an analog phone connected to channel bank the phone
> doesn't ring. If you are in call and some else calls the extension
> you get the call waiting tones and a flash works to flip to the new
> line.
>
> zaptel.conf
> span=1,1,0,esf,b8zs
> fxsls=13-24
> fxols=1-12

use "fxsks=13-24"

> loadzone = us
> defaultzone=us
>
> zapata.conf
> [channels]
>
> context = local
> language = en
> callwaiting = yes
> threewaycalling = yes
> transfer = yes
> cancelforward = yes
> callreturn = no
> usecallerid = yes
> hidecallerid = no
> echocancel = yes
> echocancelwhenbridged = yes
> ;immediate = no
> txgain=1.0
> rxgain=1.0
> callprogress=no
> busydetect=no
>
> group = 2
>
> ;use with FXO PCI card
> signalling = fxo_ls
> ;channel => 13-24
> channel => 1

channel => 1-12

>
> context = local
>
> group = 1
> ;use with FXS USB card
> signalling = fxs_ls

signalling = fxs_ks

> ;callerid = "John Doe" <(710) 555-6200>
> channel => 13

channel => 13-24

-- 
END OF LINE
   -MCP
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Re: [Asterisk-Users] Web Page initiated phone to phone

[Steven Critchfield]

On Thu, 2003-12-04 at 13:21, Todd Wallace wrote:
> Is it possible to initiate 2 outbound calls from a web page and conference
> them together in a bridge on an asterisk server?

Yes, sample.call. The first part is the phone number to dial, and the
application is dial with the other phone number.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] is there any way to search the mailing list archive and order results by date?

[Steven Critchfield]

On Thu, 2003-12-04 at 09:30, [EMAIL PROTECTED] wrote:
> i use google, with site:digium.com to search the archives, but i've never
> found a way to show the newest messages first, or limit the results to
> messages within a date range.  anybody know a better way to search that
> allows this?

Thats what a personal archive of the list is good for. My personal copy
goes back to the end of 2001. I know that doesn't help much.

Of course if you know the month and year you are looking for, google
should be able to search for those too.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] vmail.cgi with Redhat 9.0

[Evan P. Hall]

-Original Message-
From: jerk face [mailto:[EMAIL PROTECTED] 
Sent: Thursday, December 04, 2003 9:02 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] vmail.cgi with Redhat 9.0


> I recently switched from Mandrake to Redhat and I
> noticed that vmail.cgi does not work with the default
> apache installation that comes with Redhat.
> Here is what I get in my error logs:
> 
> [Thu Dec 04 11:59:57 2003] [notice] suEXEC mechanism
> enabled (wrapper: /usr/sbin/suexec)
> [Thu Dec 04 11:59:58 2003] [notice] Digest: generating
> secret for digest authentication ...
> [Thu Dec 04 11:59:58 2003] [notice] Digest: done
> [Thu Dec 04 11:59:59 2003] [notice] Apache/2.0.40 (Red
> Hat Linux) configured -- resuming normal operations
> [Thu Dec 04 12:00:08 2003] [error] [client
> 192.168.10.12] Directory index forbidden by rule:
> /var/www/html/
> [Thu Dec 04 12:00:15 2003] [error] [client
> 192.168.10.12] Premature end of script headers:
> vmail.cgi
> [Thu Dec 04 12:00:15 2003] [error] [client
> 192.168.10.12] Can't do setuid
> 
> 
> Does anybody know how I could fix this problem?
> 
> Thank you for your time.

You need to install the perl 'perl-suidperl' rpm package to enable
execution of suid perl scripts.

-Evan
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[Asterisk-Users] RE: voicemail file permissions

[Carlton J. O'Riley]

Here is a script I use in a cron job that runs every 5 minutes to make it so
that my webserver (which runs as the apache group) can access the voicemails
through the web.  Seems to fix my problems.  Although if I get the email
there is a voicemail it might be 5 minutes before I can get to it via the
web, but you could increase the frequency at which this job runs.  Five
minutes has been fine for me.  It'd be nice to be able to set the owner and
group and permissions for voicemail files in the configuration file for
voicemail.  If I had time I'd probably do it myself.

Carlton

#!/bin/sh
/bin/chgrp -R apache /var/spool/asterisk/vm/*
/bin/chmod -R g+rw /var/spool/asterisk/vm/*

>  hi, i realised that when voicemails are recorded it is set to 700 file =
permission and which leads to a serrious problem when 
>accessing the = voicemail thru the web using vmail,cgi
>
>  how can i automatically set the file permission to 755 or 777 so that = i
can make it readeable from the web? which file in * helps
> to record = the voicemail and create that voicemail in a certain dir?? if
any onw = knows, i can perhaps find that line and change as > nesseciate.
>
>  anyone tried vmail.cgi could help.
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[Asterisk-Users] Web Page initiated phone to phone

[Todd Wallace]

Is it possible to initiate 2 outbound calls from a web page and conference
them together in a bridge on an asterisk server?

Todd Wallace

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RE: [Asterisk-Users] vmail.cgi with Redhat 9.0

[jerk face]

--- "Wade J. Weppler" <[EMAIL PROTECTED]> wrote:
> RedHat's PERL doesn't allow suid.  You'll have to
> turn of the "s" flag
> on vmail.cgi (chmod -s /var/www/cgi-bin/vmail.cgi)
> and fiddle with
> permissions.
> 
> -wade

Once I turn off the 's' flag, I can run the program
but I can't view the messages.  By changing the
permissions on the voicemail folder I can listen to my
message using vmail.cgi.

In the archives
(http://lists.digium.com/pipermail/asterisk-users/2003-May/011845.html)
I found:
>>I've patched app_voicemail.c to
>>create everything as 777.  As this is a 
>>dedicated Asterisk box, I don't see
>>the harm in giving everyone on the 
>>system full access.

Would you be willing to share this patch? 


> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
> Behalf Of Olle E.
> Johansson
> Sent: Thursday, December 04, 2003 1:36 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] vmail.cgi with Redhat
> 9.0
> 
> jerk face wrote:
> 
> > I recently switched from Mandrake to Redhat and I
> > noticed that vmail.cgi does not work with the
> default
> > apache installation that comes with Redhat.
> > Here is what I get in my error logs:
> > 
> > [Thu Dec 04 11:59:57 2003] [notice] suEXEC
> mechanism
> > enabled (wrapper: /usr/sbin/suexec)
> > [Thu Dec 04 11:59:58 2003] [notice] Digest:
> generating
> > secret for digest authentication ...
> > [Thu Dec 04 11:59:58 2003] [notice] Digest: done
> > [Thu Dec 04 11:59:59 2003] [notice] Apache/2.0.40
> (Red
> > Hat Linux) configured -- resuming normal
> operations
> > [Thu Dec 04 12:00:08 2003] [error] [client
> > 192.168.10.12] Directory index forbidden by rule:
> > /var/www/html/
> > [Thu Dec 04 12:00:15 2003] [error] [client
> > 192.168.10.12] Premature end of script headers:
> > vmail.cgi
> > [Thu Dec 04 12:00:15 2003] [error] [client
> > 192.168.10.12] Can't do setuid
> > 
> > 
> > Does anybody know how I could fix this problem?
> "Premature end of script headers" means that there
> was an error in the
> script.
> Check the error log of the Apache server for more
> information.
> You can also try to run the script from the UNIX
> prompt and see if you
> get any
> error messages.
> 
> Regards,
> /Olle
> 
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> 
> 
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Re: [Asterisk-Users] Asterisk behind NAT << How to do it.

[William Waites]

On Thu, Dec 04, 2003 at 07:56:56PM +0100, robert ivanc wrote:

> this patch seems to break my GS phones that are connecting to * via NAT. 
> The one before that works ok - 249 or something? They can't connect 
> anymore - get a Not Found error back.

That is very strange -- the *only* difference between those two versions
of the patch is the variable naming. Can you give me some more debugging
information? Some more information on your setup and perhaps a trace of
the SIP conversation? I don't have a GS phone to test with here.

Thanks,
-w
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Re: [Asterisk-Users] vmail.cgi with Redhat 9.0

[jerk face]

--- Lists <[EMAIL PROTECTED]> wrote:
> On Thu, 4 Dec 2003, Olle E. Johansson wrote:
> 
> > jerk face wrote:
> > 
> > > I recently switched from Mandrake to Redhat and
> I
> > > noticed that vmail.cgi does not work with the
> default
> > > apache installation that comes with Redhat.
> > > Here is what I get in my error logs:
> > > 
> > > [Thu Dec 04 11:59:57 2003] [notice] suEXEC
> mechanism
> > > enabled (wrapper: /usr/sbin/suexec)
> > > [Thu Dec 04 11:59:58 2003] [notice] Digest:
> generating
> > > secret for digest authentication ...
> > > [Thu Dec 04 11:59:58 2003] [notice] Digest: done
> > > [Thu Dec 04 11:59:59 2003] [notice]
> Apache/2.0.40 (Red
> > > Hat Linux) configured -- resuming normal
> operations
> > > [Thu Dec 04 12:00:08 2003] [error] [client
> > > 192.168.10.12] Directory index forbidden by
> rule:
> > > /var/www/html/
> > > [Thu Dec 04 12:00:15 2003] [error] [client
> > > 192.168.10.12] Premature end of script headers:
> > > vmail.cgi
> > > [Thu Dec 04 12:00:15 2003] [error] [client
> > > 192.168.10.12] Can't do setuid
> > > 
> > > 
> > > Does anybody know how I could fix this problem?
> > "Premature end of script headers" means that there
> was an error in the script.
> > Check the error log of the Apache server for more
> information.
> > You can also try to run the script from the UNIX
> prompt and see if you get any
> > error messages.
> > 
> > Regards,
> > /Olle
> 
> 
> go into /var/www/cgi-bin
> do a 
> chmod 700 vmail.cgi
> chown apache.users vmail.cgi
> 
> Michael
> 

>>> chmod 700 vmail.cgi
>>> chown apache.users vmail.cgi
That will let me run the script, but now it won't
display the messages.  My guess is because apache
doesn't have permissions to view the INBOX folder.

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Re: [Asterisk-Users] Asterisk behind NAT << How to do it.

[robert ivanc]

Arnold Ligtvoet wrote:

  Leif wrote:
  
  
Awesome!  Have you tried the newer patch / diff for 1.259 (which as of
right now is the newest chan_sip file).  If you goto bugs.digium.com and
login anonymously and jump to bug 104, then you can get the newest
patch.  Same instructions as before.

  
  
  

this patch seems to break my GS phones that are connecting to * via
NAT. The one before that works ok - 249 or something? They can't
connect anymore - get a Not Found error back.

Regards,

  Robert


  Installed the new patch, no errors here. Ran make and copied chan_sip.o. All
went fine.

  
  
I just updated it to test the new sip.conf structure which is

externip=
localnet=
localmask=

  
  
Updated my sip.conf to match these settings. The result seems to be better,
yesterday I noticed a slight delay in the setup of the audio channel, the
speaking clock would only start at the second word, this is now gone.

  
  
Still working great for me here!

BTW!   Can you login to the bug tracker and post a comment ?  Thanks!

  
  
I do have one strange issue. I have a test setup here which is very simple.
* server and one windows machine. * is connected via ISDN (chan_i4l) to my
home pbx. On my windows machine I installed Diax, SjPhone and SIPPS. The
strange thing I now have is that both skinny clients are able to receive
audio but not send any when I call an extension on my pbx (so external for
*). I first thought it was my mic, but diax is working fine.

I have already been looking at my sip.conf for the extensions but I'm not
sure if this is the problem. Anyway my sip.conf now is :
[general]
disallow=all; Disallow all codecs
allow=ulaw  ; Allow codecs in order of preference
allow=alaw
allow=ilbc
allow=gsm

; for fix 1.259
externip=212.238.144.173
localnet=192.168.0.100
localmask=255.255.255.0

[phone1]
type=friend
host=dynamic
defaultip=192.168.0.2
dtmfmode=inband
mailbox=1000 ; Mailbox for message waiting indicator
context=default
callerid="Me" <2124>
;reinvite=no
;canreinvite=no
;nat=yes
;insecure=yes

I'll wait your reply for the one-way sound 'issue' (probably me!) before
posting to the bugtracker. Hopefully someone has some clue as to why my sip
clients are not able to send sound.

Thanks,
Arnold Ligtvoet.

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RE: [Asterisk-Users] vmail.cgi with Redhat 9.0

[Wade J. Weppler]

RedHat's PERL doesn't allow suid.  You'll have to turn of the "s" flag
on vmail.cgi (chmod -s /var/www/cgi-bin/vmail.cgi) and fiddle with
permissions.

-wade

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Thursday, December 04, 2003 1:36 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] vmail.cgi with Redhat 9.0

jerk face wrote:

> I recently switched from Mandrake to Redhat and I
> noticed that vmail.cgi does not work with the default
> apache installation that comes with Redhat.
> Here is what I get in my error logs:
> 
> [Thu Dec 04 11:59:57 2003] [notice] suEXEC mechanism
> enabled (wrapper: /usr/sbin/suexec)
> [Thu Dec 04 11:59:58 2003] [notice] Digest: generating
> secret for digest authentication ...
> [Thu Dec 04 11:59:58 2003] [notice] Digest: done
> [Thu Dec 04 11:59:59 2003] [notice] Apache/2.0.40 (Red
> Hat Linux) configured -- resuming normal operations
> [Thu Dec 04 12:00:08 2003] [error] [client
> 192.168.10.12] Directory index forbidden by rule:
> /var/www/html/
> [Thu Dec 04 12:00:15 2003] [error] [client
> 192.168.10.12] Premature end of script headers:
> vmail.cgi
> [Thu Dec 04 12:00:15 2003] [error] [client
> 192.168.10.12] Can't do setuid
> 
> 
> Does anybody know how I could fix this problem?
"Premature end of script headers" means that there was an error in the
script.
Check the error log of the Apache server for more information.
You can also try to run the script from the UNIX prompt and see if you
get any
error messages.

Regards,
/Olle

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Re: [Asterisk-Users] vmail.cgi with Redhat 9.0

[jerk face]

--- "Olle E. Johansson" <[EMAIL PROTECTED]> wrote:
> jerk face wrote:
> 
> > I recently switched from Mandrake to Redhat and I
> > noticed that vmail.cgi does not work with the
> default
> > apache installation that comes with Redhat.
> > Here is what I get in my error logs:
> > 
> > [Thu Dec 04 11:59:57 2003] [notice] suEXEC
> mechanism
> > enabled (wrapper: /usr/sbin/suexec)
> > [Thu Dec 04 11:59:58 2003] [notice] Digest:
> generating
> > secret for digest authentication ...
> > [Thu Dec 04 11:59:58 2003] [notice] Digest: done
> > [Thu Dec 04 11:59:59 2003] [notice] Apache/2.0.40
> (Red
> > Hat Linux) configured -- resuming normal
> operations
> > [Thu Dec 04 12:00:08 2003] [error] [client
> > 192.168.10.12] Directory index forbidden by rule:
> > /var/www/html/
> > [Thu Dec 04 12:00:15 2003] [error] [client
> > 192.168.10.12] Premature end of script headers:
> > vmail.cgi
> > [Thu Dec 04 12:00:15 2003] [error] [client
> > 192.168.10.12] Can't do setuid
> > 
> > 
> > Does anybody know how I could fix this problem?
> "Premature end of script headers" means that there
> was an error in the script.
> Check the error log of the Apache server for more
> information.
> You can also try to run the script from the UNIX
> prompt and see if you get any
> error messages.
> 
> Regards,
> /Olle
> 

The logs I provided were from
/etc/httpd/logs/error_log.
I also ran vmail.cgi from the command line and there
were no errors.

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