Re: [Asterisk-Users] ringback

[Peter Svensson]

On Thu, 20 Jan 2005, Steve Clark wrote:

> Andrew Kohlsmith wrote:
> > On January 20, 2005 02:15 pm, Steve Clark wrote:
> >>I am dialing from one zap channel to a second zap channel. Is there a way
> >>to keep the channel I am dialing to from generating a ringback tone.
> > 
> > exten => 1,Dial(Zap/1)
> > 
> > should not generate ringback...  
> > 
> > exten => 1,Dial(Zap/1,,r)
> > 
> > should generate ringback.
> > 
> That what I thought also but it doesn't work. I think what it really does is 
> generates a ringback if the remote end is not generating it.
> 
> In any event when I tried it both ways I still got a ringback tone.

The "r" option instructs asterisk to generate ringback locally. Without it 
audio will be passed from the called party to the calling party. You can 
use the music on hold option to the dial application to break the reverse 
audio path. If you want silence, you can set up a MoH-class with a silent 
sound clip.

Peter

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[Asterisk-Users] Stanaphone incoming calls problem.

[guru]

Hi,
I have an stanaphone number, and when I first start asterisk everything
works fine, but after a few hours of asterisk running incoming calls fail,
and when calling my stanaphone remotely, it takes me directly to the
stanaphone voicemail, restarting asterisk fixes the problem for a few
hours.

I believe the problem is due to the fact that I have a dynamic IP, and the
IP might be changing very often. Is there a way to force asterisk to re
register to stanaphone at some time intervals ? or some better way to fix
this problem.

Has anyone had this problem ?

Any help will be greatly apreciated.

Thanks,

Sergio Riveros

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Re: [Asterisk-Users] Cisco 7940G

[Glenn Powers]

[EMAIL PROTECTED] wrote:
Is it somehow possible to enable web-based configuration on them?  I made the
changes in the config file but it still doesn't register with Asterisk.
 

AFAIK, Cisco 79xx phones don't have web-based configuration. They have a 
telnet interface, though
it's enabled/disabled based on the config files the phone gets from the 
TFTP server.

cheers,
glenn
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Re: [Asterisk-Users] Becoming a VOIP provider

[Julien Goodwin]

On Wed, Jan 19, 2005 at 02:53:56PM -0700, Keith Burns arranged a set of bits 
into the following:
> Be careful of LI requirements in Australia.

Which ones, there are pretty much none!
What we have in .au (for quite a lot of things in reality) is a
PERCEPTION that these things are legislated (like free speech and
lifeline services) but they arn't. And I can confirm that under
australian communications law if you're doing VoIP lifeline is not a
requirement (and if required I can supply contact details of one of the
people who recently got that statement claified in the law).

> You MAY be able to put the onus for this on your upstream (PRI/IMT)
> provider, but if you have many, this could be messy.
> 
> Best bet would be to have a solution yourself... when I was looking into
> this the good news was that the enforcement agencies (which at last
> count was around 47, any of whom could hit you for their own real-time
> feed of the conversation) were considering taking the VoIP feed (RTP)
> and the logs of the signaling. (Things may have changed, your mileage
> may vary, yada, yada, yada).
> 
> Also, after a little kiddy died of an asthma attack in rural Victoria
> because Telstra (the lazy @[EMAIL PROTECTED]  - I digress) hadn't fixed their 
> phone,
> lifeline services (E911 in the US) are more and more important to have
> nailed.. you don't want that on your conscience (your service not
> working causing harm to someone) nor would your business appreciate the
> lawsuits.
> 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Ed Robbins
> > Sent: Wednesday, January 19, 2005 2:02 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Becoming a VOIP provider
> > 
> > Ty Carter wrote:
> > 
> > >Ed:
> > >
> > >I think you must have some bad information here.VoIP is an
> Information
> > >service and not subject to CALEA regulations.
> > >
> > >
> > >
> > 
> > Whether it's a subject to those regulations or not I still know first
> > hand it's a big issue with broadband voip providers.  I work for a
> > company that develops VoiP for the broadband market and it's something
> > we had to develop for our customers.  I don't know all the details of
> it
> > and what is going on behind the scenes in terms of regulations but my
> > thinking is that voip providers have to tie into the PSTN somewhere
> and
> > the FCC can most likely tap into(no pun intended), meaning require you
> > meet the guidlines put forth in CALEA, from that legal point of view.
> I
> > had never thought about this before but I should talk to my buddy who
> > got a CLEC a few years ago, I'm wondering if there is something in
> there
> > that spells it out.
> > 
> > Ed
> > 
> > >According to the calea website:
> > >
> > >In a Notice of Proposed Rulemaking FCC 02-42 released on February 15,
> 2002,
> > >the FCC initiated a proceeding to establish rules and regulations
> regarding
> > >the classification of "wireline broadband Internet access" under the
> > >Telecommunications Act. Digital Subscriber Line (DSL) service is an
> example
> > >of wireline broadband Internet access. In this document, the FCC
> > >"tentatively" decided that wireline broadband Internet access is an
> > >"information service."
> > >
> > >In a Declaratory Ruling and Notice of Proposed Rulemaking FCC 02-77
> released
> > >on March 15, 2002, the FCC made a "declaratory ruling" that cable
> modem
> > >service (Internet access through cable TV lines) is an "information
> service"
> > >under the Telecommunication Act and initiated a proceeding to
> establish
> > >rules and regulations based on that finding.
> > >
> > >Therefore, the FCC's pending wireline broadband Internet access
> proceeding
> > >is CC Docket Nos. 02-33, 95-20, and 98-10 and the cable modem
> broadband
> > >Internet access proceeding is CS Docket No. 02-52 (collectively the
> "FCC
> > >Broadband Proceedings").
> > >
> > >It should be noted that the FCC is not primarily focusing on CALEA in
> these
> > >proceedings, rather its emphasis is on the economic and policy
> concerns
> > >involved in regulation of these services under the Communications
> Act.
> > >However, since CALEA exempts "information service" from the
> surveillance
> > >capability requirements of Section 103, these FCC decisions have the
> > >potential to exclude broadband DSL and cable modem service from CALEA
> > >compliance.
> > >
> > >The FBI filed the following comments in the Broadband
> > >
> > >
> > >
> > >>-Original Message-
> > >>From: [EMAIL PROTECTED]
> > >>[mailto:[EMAIL PROTECTED] On Behalf Of
> > >>Ed Robbins
> > >>Sent: Wednesday, January 19, 2005 3:19 PM
> > >>To: Asterisk Users Mailing List - Non-Commercial Discussion
> > >>Subject: Re: [Asterisk-Users] Becoming a VOIP provider
> > >>
> > >>Manjit Riat wrote:
> > >>
> > >>
> > >>
> > >>>That was a really nice description... Can you do 1-14 and I'll

[Asterisk-Users] Polycom IP 300/500 Conferencing Behavior

[Greg Boehnlein]

Hello,
I've got a mixture of SPIP 300 and 500 phones in production for 
various clients. I've got the XML settings configured for local 
conferencing, but I'm not seeing the expected behavior from the phone when 
I attempt to conference two calls together. According to the manual, while 
talking to the first party, you simply hit Conference, dial the second 
party and then Conference to join them. This is supposed to put the first 
party on Hold until you bridge them together with the second press of the 
Conference button.
That is all fine and well, but it doesn't quite work the way that 
the manual describes. Instead of joining the two calls together when the 
Conference key is pressed for the second time, the first party is taken 
off hold and hears dead silence. The only way to correctly join the 
parties is to hit the Hold and then Resume soft key, at which point all 
three parties can talk to each other.

As an illustration

Conf -> Dial -> Conf doesn't work.

However,

Conf -> Dial -> Conf -> Hold -> Resume DOES work.

I'm running 1.3.4 firmware on all the phones, and I can't for the 
life of me figure out what is causing this problem. It is very likely some 
misconfiguration in the XML files, but I can't find it. Anyone have any 
suggestions?

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST

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Re: [Asterisk-Users] Accessing Voice mail

[Brian Dingman]

Kurt,
Here is a real basic setup of how the a extension can be used in
context with the rest of the dialplan. The a extension must call
VoiceMailMain NOT Voicemail or you will get your voicemail again and
not the voicemail system.

[fromPSTN]
exten => s,1,Answer
exten => s,2,Dial(${RINGPHONENUMBERS},15,r)
exten => s,3,Voicemail,u${VMBOX}
exten => s,4,Hangup

exten => a,1,VoicemailMain
exten => a,2,Hangup


On Thu, 20 Jan 2005 15:30:14 -0500, kurt x <[EMAIL PROTECTED]> wrote:
> Brain,
> 
> I did what you suggested but instead of going to VoiceMailMain it
> starts the begining of
> my recorded message each time I press the "*" key.
> 
> [vmail]
> exten => a,1,Voicemail(u${ext})
> exten => a,2,Hangup
> 
> Kurt
>
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RE: [Asterisk-Users] VoIP-to-TDM processing on-card?

[clive]

As far as I am aware, Atacomm is in the process of building a DSP 
based card which would work with asterisk

I just checked, its going to be called a "ipVolution TDM120"

On 20 Jan 2005 at 14:42, Michael Baird wrote:

> On Thu, 2005-01-20 at 14:05 -0500, Olson, Dana wrote:
> > I did look there. If you read my follow up, I screwed up the original 
> > question. What I want is a card with multiple T1 ports that do the 
> > processing on the card, and not on the system CPU.
> > 
> 
> I'm not aware of any cards with DSP's on board for Asterisk (nice
> thought), the Digium cards I have rely on the PC's CPU to handle the
> calls.
> 
> > Is there a mailing list for Asterisk where people treat each other in a 
> > civil manner?
> > __
> > Dana Olson
> 
> It's only one guy who seems to attack each poster for not posting in a
> manner of which he approves (there is one/two of these fellows on every
> mailing list), don't let him ruin your day, this list is quite helpful
> and many guys will give you a good answer without the extra attitude.
> 
> Regards
> Michael Baird
> 
> 
> 
> 
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[Asterisk-Users] choppy audio.

[Chris Polk]

Hi:
any thoughts on this?
I am using broadvoice, and am creating an information line for public 
resources.
However, the audio is very choppy. I even notice when I call myself from an 
external source, my voicemail is also choppy.
However, enternally it sounds wonderful.
Any help?

Chris
- Original Message - 
From: "Howard Lowndes" <[EMAIL PROTECTED]>
To: 
Sent: Thursday, January 20, 2005 10:02 PM
Subject: [Asterisk-Users] Zap randomly hanging up


I have a zap line on a X101P which will occasionally just hang up the
call for no apparent reason.  Is there any good way of trying to
diagnose what might be causing this?  Monitoring the asterisk output in
verbose mode does not provide any indications.
--
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."
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Re: [Asterisk-Users] Segmentation Fault after Digitnetwork X100Pinstall

[Paul Fielding]

Shouldn't you contact your vendor for support and not a different
vendors support channel?
Um, I didn't think this was a Digium support channel.  I thought this was an
Asterisk Users channel.  Seems to me the question should be fair game.
(Sorry I don't have an answer to your question, though, Dave).
regards,
Paul
- Original Message - 
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Thursday, January 20, 2005 9:41 PM
Subject: Re: [Asterisk-Users] Segmentation Fault after Digitnetwork 
X100Pinstall


On Fri, 2005-01-21 at 16:57 +1300, Dave Green wrote:
I've just installed a Digitnetworks X100P clone in my * server and run
the install script for the voicepet-single-x100p tarball. The install
appeared to run OK with modprobe wcfxo successful and the ztcfg
reporting "Channel 01: FXS Kewlstart (Default) (Slaves: 01)".
When I try to start * though I get a segmentation fault after loading
res_features.so. I discovered that the Digitnetwork install script seems
to modify all of the .conf files, leaving .conf.old copies. I tried
moving the .conf.old files back to .conf but am still get the seg fault.
Shouldn't you contact your vendor for support and not a different
vendors support channel?
Another company with retarded disclaimers sending to a KNOWN publicly
archived mailing list. Fix the problem or be ridiculed regularly for the
stupidity.

CAUTION:
This message and any attachments contain privileged and confidential
information.  If you are not the intended recipient of this message, you
are hereby notified that any use, dissemination, distribution or
reproduction of this message is prohibited. If you have received this
message in error please notify the sender immediately via email and then
destroy this message and any attachments.
Any views expressed in this message are those of the individual sender
and may not necessarily reflect the views of Winstone Pulp International
Ltd.
--
Steven Critchfield <[EMAIL PROTECTED]>
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RE: [Asterisk-Users] Some more hardware and E1 questions

[Peter Childs]

 Personally I would for the time being steer clear of anything with a
 Intel E7520 Chipset (newish) such as the HP DL380 G4 etc... if you
 are using a TE410P card (ie 3.3v).

 Just my 2 cents.   But you can always give it a go :)

 Cheers,
  Peter

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Daniel
Nyström
Sent: Thursday, 20 January 2005 11:55 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Some more hardware and E1 questions


Hi again folks! ;)

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

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

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

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

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

Thanks!

BR
Daniel Nyström
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RE: [Asterisk-Users] Cisco 7940G

[Brian West]

Looks like you didn't read the docs also... you need OS79XX.TXT  too

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Thursday, January 20, 2005 11:56 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Cisco 7940G
> 
> Is it somehow possible to enable web-based configuration on them?  I made
> the
> changes in the config file but it still doesn't register with Asterisk.
> 
> Quoting Gonzalo Gasca Meza <[EMAIL PROTECTED]>:
> 
> >
> > I got similar issues, Im running P0S3-07-2-00 load
> > In your tftpboot folder in your TFTP server make sure you have these
> files:
> >
> > CTLSEP000D651CF3FB.tlv
> > SEP000D651CF3FB.cnf.xml
> >
> > SIP000D651CF3FB.cnf
> >
> > [EMAIL PROTECTED] tftpboot]# cat CTLSEP000D651CF3FB.tlv
> > P0S3-07-2-00
> > [EMAIL PROTECTED] tftpboot]# cat SEP000D651CF3FB.cnf.xml
> > 
> >
> >
> >
> >
> >
> >2000
> >
> >110.10.200.2
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >P0S3-07-2-
> 00
> >
> >
> >
> > 
> >
> > [EMAIL PROTECTED] tftpboot]# cat SIP000D651CF3FB.cnf
> >
> > # SIP Configuration Generic File (start)
> > image_version: P0S3-07-2-00
> > .
> >
> > .
> >
> >
> >
> > .
> >
> > [output cut]
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > -
> > Do you Yahoo!?
> >  Yahoo! Mail - Find what you need with new enhanced search. Learn more.
> 
> 
> 
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[Asterisk-Users] Zap randomly hanging up

[Howard Lowndes]

I have a zap line on a X101P which will occasionally just hang up the
call for no apparent reason.  Is there any good way of trying to
diagnose what might be causing this?  Monitoring the asterisk output in
verbose mode does not provide any indications.


-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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RE: [Asterisk-Users] Cisco 7940G

[Brian West]

Because its got an SCCP load on it.  It can't speak SIP you'll have to get
the sip firmware and update.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Thursday, January 20, 2005 11:56 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Cisco 7940G
> 
> Is it somehow possible to enable web-based configuration on them?  I made
> the
> changes in the config file but it still doesn't register with Asterisk.
> 
> Quoting Gonzalo Gasca Meza <[EMAIL PROTECTED]>:
> 
> >
> > I got similar issues, Im running P0S3-07-2-00 load
> > In your tftpboot folder in your TFTP server make sure you have these
> files:
> >
> > CTLSEP000D651CF3FB.tlv
> > SEP000D651CF3FB.cnf.xml
> >
> > SIP000D651CF3FB.cnf
> >
> > [EMAIL PROTECTED] tftpboot]# cat CTLSEP000D651CF3FB.tlv
> > P0S3-07-2-00
> > [EMAIL PROTECTED] tftpboot]# cat SEP000D651CF3FB.cnf.xml
> > 
> >
> >
> >
> >
> >
> >2000
> >
> >110.10.200.2
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >P0S3-07-2-
> 00
> >
> >
> >
> > 
> >
> > [EMAIL PROTECTED] tftpboot]# cat SIP000D651CF3FB.cnf
> >
> > # SIP Configuration Generic File (start)
> > image_version: P0S3-07-2-00
> > .
> >
> > .
> >
> >
> >
> > .
> >
> > [output cut]
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > -
> > Do you Yahoo!?
> >  Yahoo! Mail - Find what you need with new enhanced search. Learn more.
> 
> 
> 
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Re: [Asterisk-Users] Cisco 7940G

[sgup015]

Is it somehow possible to enable web-based configuration on them?  I made the
changes in the config file but it still doesn't register with Asterisk.

Quoting Gonzalo Gasca Meza <[EMAIL PROTECTED]>:

>
> I got similar issues, Im running P0S3-07-2-00 load
> In your tftpboot folder in your TFTP server make sure you have these files:
>
> CTLSEP000D651CF3FB.tlv
> SEP000D651CF3FB.cnf.xml
>
> SIP000D651CF3FB.cnf
>
> [EMAIL PROTECTED] tftpboot]# cat CTLSEP000D651CF3FB.tlv
> P0S3-07-2-00
> [EMAIL PROTECTED] tftpboot]# cat SEP000D651CF3FB.cnf.xml
> 
>
>
>
>
>
>2000
>
>110.10.200.2
>
>
>
>
>
>
>
>
>
>P0S3-07-2-00
>
>
>
> 
>
> [EMAIL PROTECTED] tftpboot]# cat SIP000D651CF3FB.cnf
>
> # SIP Configuration Generic File (start)
> image_version: P0S3-07-2-00
> .
>
> .
>
>
>
> .
>
> [output cut]
>
>
>
>
>
>
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>
> -
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>  Yahoo! Mail - Find what you need with new enhanced search. Learn more.



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[Asterisk-Users] Can anyone recoment T1/PRI provider in South Ontario?

[Robert Augustyn]

Hi,
I am looking for a good provider of T1/PRI in Windsor,
Ontario.

Any sugestions would be greatly appreciated.
robert
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Re: [Asterisk-Users] sip OPTIONS

[Andres]

Erik Versaevel wrote:
Hello all,
Can asterisk itself respond to a SIP OPTIONS packet? i wan't to check 
if asterisk is still alive by using sipsak (because of nagois & mon)
Sure it does.  It answers with "404 Not Found".  We monitor all our SER 
and Asterisk servers with a Nagios plugin based on sipsak that was 
witten by somebody on the SER list.  We just modified it a bit to fit 
Asterisk.  If you are interested in the plugin I can post it tomorrow.

Andres
Network Admin
http://www.telesip.net

Kind regards,
E. Versaevel
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Re: [Asterisk-Users] Segmentation Fault after Digitnetwork X100P install

[Steven Critchfield]

On Fri, 2005-01-21 at 16:57 +1300, Dave Green wrote:
> I've just installed a Digitnetworks X100P clone in my * server and run 
> the install script for the voicepet-single-x100p tarball. The install 
> appeared to run OK with modprobe wcfxo successful and the ztcfg 
> reporting "Channel 01: FXS Kewlstart (Default) (Slaves: 01)".
> 
> When I try to start * though I get a segmentation fault after loading 
> res_features.so. I discovered that the Digitnetwork install script seems 
> to modify all of the .conf files, leaving .conf.old copies. I tried 
> moving the .conf.old files back to .conf but am still get the seg fault.

Shouldn't you contact your vendor for support and not a different
vendors support channel?


Another company with retarded disclaimers sending to a KNOWN publicly
archived mailing list. Fix the problem or be ridiculed regularly for the
stupidity.
> 
> CAUTION:
> This message and any attachments contain privileged and confidential
> information.  If you are not the intended recipient of this message, you
> are hereby notified that any use, dissemination, distribution or
> reproduction of this message is prohibited. If you have received this
> message in error please notify the sender immediately via email and then
> destroy this message and any attachments.
> 
> Any views expressed in this message are those of the individual sender
> and may not necessarily reflect the views of Winstone Pulp International
> Ltd.

-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Ring an incoming call in multiple extensions

[C F]

On Thu, 20 Jan 2005 21:44:39 -0600, Steven Critchfield
<[EMAIL PROTECTED]> wrote:
> On Thu, 2005-01-20 at 21:32 -0600, [EMAIL PROTECTED] wrote:
> > Hi asterisk users!
> >
> >
> >
> > Hereÿs my issue, Iÿve deleted the ´s¡ extension cause I donÿt want any
> > action to be taken on incoming calls as my pbx is for home use, but I
> > would like to ring all my VoIP extensions at the same time the PSTN
> > line rings and to be able to pick up the call in any extension,
> > honestly I donÿt know if this is possible, some ideas ???
> 
> Quit fighting the system. You have to have an extension to do anything.
> You have to dial from an extension to ring anything.


> Once you dial and the other end answers, you can't continue extension.conf 
> processing.
Why not what ever happened to the g option in Dial?
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Re: [Asterisk-Users] Segmentation Fault after Digitnetwork X100P install

[Dave Green]

Henry Devito wrote:
Just download 1.0.3, don't use the voicepet scripts.  they are 
versions 0.7 and 0.9 of  *.  If it is a true clone from digit networks 
it will work fine with 1.0.3.  I just set up 3 different machines with 
the cards from digit networks.

Henry
Yes, thanks for that. After posting I wondered if the script installed 
the whole * package, so I recompiled/installed the cvs I'd downloaded 
and after a bit more fiddling have managed ot get it working.

Dave

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Re: [Asterisk-Users] Troubles with Broadvoice (register) {Scanned}

[Luki]

> 2) Does * check /etc/hosts or do dns lookups every time registration is
> done or an an outbound call is placed?

I'm pretty sure the OS is checking /etc/hosts and * looks up the
address every time it registers -- at least I didn't have to reload it
when I changed the IP in /etc/hosts.

I'm not sure about the performance level checker, but why not allow to
specify alternate servers? If registration at the preferred server
fails, try the next best one. Asterisk already has a timeout after
which it considers a host unreachable, so this should be fairly easy
to do. End after a (pre)specified recovery time let it retry the
preferred server.

I wonder where you would specify the alternate hosts though... DNS
seems most logical to me, but does say Broadvoice provide round robin
lookups for sip.broadvoice.com? I don't think so.

--Luki
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Re: [Asterisk-Users] Segmentation Fault after Digitnetwork X100P install

[Henry Devito]

Just download 1.0.3, don't use the voicepet scripts.  they are versions 0.7 
and 0.9 of  *.  If it is a true clone from digit networks it will work fine 
with 1.0.3.  I just set up 3 different machines with the cards from digit 
networks.

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

Sent: Thursday, January 20, 2005 9:57 PM
Subject: [Asterisk-Users] Segmentation Fault after Digitnetwork X100P 
install


I've just installed a Digitnetworks X100P clone in my * server and run the 
install script for the voicepet-single-x100p tarball. The install appeared 
to run OK with modprobe wcfxo successful and the ztcfg reporting "Channel 
01: FXS Kewlstart (Default) (Slaves: 01)".

When I try to start * though I get a segmentation fault after loading 
res_features.so. I discovered that the Digitnetwork install script seems 
to modify all of the .conf files, leaving .conf.old copies. I tried moving 
the .conf.old files back to .conf but am still get the seg fault.

Has anyone else run into this kind of problem ?   The * server has been 
running fine up until now.

I can't get it to show the * version right now, but compiled from CVS 
around 11 Jan.

Dave


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Any views expressed in this message are those of the individual sender
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Re: [Asterisk-Users] Re: API Call Bridge?

[taf taffey]

I've tried all the Wiki pages and still can't seem to
get this thing working and that's why I've posted this
mail.

I would like to dial two external numbers and
conference them together using the asterisk api
manager.
I can only get this working for a single call when
originating from a sip/xxx channel.

Would appreciate some pointers other than check wiki
again :-)


 --- Brian Roy <[EMAIL PROTECTED]> wrote: 
> On Fri, 21 Jan 2005 03:24:26 + (GMT), taf taffey
> <[EMAIL PROTECTED]> wrote:
> 
> > Is there a way to dial two
> > outbound/external numbers and bridge them together
> > using the Asterisk API manager method instead??
> > 
> > Cheers,
> > Taff.
> 
> Use the wiki luke!
> 
> http://www.voip-info.org/wiki-Asterisk+manager+API
> 
> -Chuji
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Re: [Asterisk-Users] Ring an incoming call in multiple extensions

[Brian Roy]

On Thu, 20 Jan 2005 21:32:37 -0600, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> 
> 
> Hi asterisk users!
> 
>  
> 
> Here's my issue, I've deleted the "s" extension cause I don't want any
> action to be taken on incoming calls as my pbx is for home use, but I would
> like to ring all my VoIP extensions at the same time the PSTN line rings and
> to be able to pick up the call in any extension, honestly I don't know if
> this is possible, some ideas ???
> 


Use the & between your dial targets. 

>From the sample config

;exten => 6245,1,Dial(SIP/Grandstream1&SIP/Xlite1,20,rtT)

wiki page is here
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial

Have fun,

-Chuji
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[Asterisk-Users] Segmentation Fault after Digitnetwork X100P install

[Dave Green]

I've just installed a Digitnetworks X100P clone in my * server and run 
the install script for the voicepet-single-x100p tarball. The install 
appeared to run OK with modprobe wcfxo successful and the ztcfg 
reporting "Channel 01: FXS Kewlstart (Default) (Slaves: 01)".

When I try to start * though I get a segmentation fault after loading 
res_features.so. I discovered that the Digitnetwork install script seems 
to modify all of the .conf files, leaving .conf.old copies. I tried 
moving the .conf.old files back to .conf but am still get the seg fault.

Has anyone else run into this kind of problem ?   The * server has been 
running fine up until now.

I can't get it to show the * version right now, but compiled from CVS 
around 11 Jan.

Dave


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are hereby notified that any use, dissemination, distribution or
reproduction of this message is prohibited. If you have received this
message in error please notify the sender immediately via email and then
destroy this message and any attachments.
Any views expressed in this message are those of the individual sender
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Re: [Asterisk-Users] Re: API Call Bridge?

[Brian Roy]

On Fri, 21 Jan 2005 03:24:26 + (GMT), taf taffey
<[EMAIL PROTECTED]> wrote:

> Is there a way to dial two
> outbound/external numbers and bridge them together
> using the Asterisk API manager method instead??
> 
> Cheers,
> Taff.

Use the wiki luke!

http://www.voip-info.org/wiki-Asterisk+manager+API

-Chuji
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Re: [Asterisk-Users] Ring an incoming call in multiple extensions

[Henry Devito]

Very simple
 
exten => s,1,answer
exten => 
s,2,Dial(SIP/${EXTEN1}&(SIP/${EXTEN2}

  - Original Message - 
  From: 
  [EMAIL PROTECTED] 
  To: asterisk-users@lists.digium.com 
  
  Sent: Thursday, January 20, 2005 9:32 
  PM
  Subject: [Asterisk-Users] Ring an 
  incoming call in multiple extensions
  
  
  Hi asterisk 
  users!
   
  Here’s my issue, I’ve deleted the “s” extension cause 
  I don’t want any action to be taken on incoming calls as my pbx is for home 
  use, but I would like to ring all my VoIP extensions at the same time the PSTN 
  line rings and to be able to pick up the call in any extension, honestly I 
  don’t know if this is possible, some ideas ???
   
  Thanks in advance!
  
  

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Re: [Asterisk-Users] Ring an incoming call in multiple extensions

[Howard Lowndes]

On Fri, 2005-01-21 at 14:32, [EMAIL PROTECTED] wrote:
> Hi asterisk users!
> 
>  
> 
> Hereâs my issue, Iâve deleted the âsâ extension cause I donât want 
> any
> action to be taken on incoming calls as my pbx is for home use, but I
> would like to ring all my VoIP extensions at the same time the PSTN
> line rings and to be able to pick up the call in any extension,
> honestly I donât know if this is possible, some ideas ???

You still need your "s" exten, but when you do the Dial app you just do
it to multiple extensions

exten => s,n,Dial(SIP/111&SIP/122&SIP/133)

They all ring and the first one that answers gets the call.
> 
>  
> 
> Thanks in advance!
> 
> 
> 
> __
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Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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Re: [Asterisk-Users] Ring an incoming call in multiple extensions

[Steven Critchfield]

On Thu, 2005-01-20 at 21:32 -0600, [EMAIL PROTECTED] wrote:
> Hi asterisk users!
> 
>  
> 
> Here’s my issue, I’ve deleted the “s” extension cause I don’t want any
> action to be taken on incoming calls as my pbx is for home use, but I
> would like to ring all my VoIP extensions at the same time the PSTN
> line rings and to be able to pick up the call in any extension,
> honestly I don’t know if this is possible, some ideas ???

Quit fighting the system. You have to have an extension to do anything.
You have to dial from an extension to ring anything. Once you dial and
the other end answers, you can't continue extension.conf processing. So
guess what, you have to answer the phone line with asterisk and then
dial whatever phone lines/extensions/voip phones. 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] softswitch dilemma

[Robert Jackson]

Please turn off HTML in your e-mail client.

>-Original Message-
>From: Diego Ventrice [mailto:[EMAIL PROTECTED] 
>Sent: Thursday, January 20, 2005 10:41 PM
>To: asterisk-users@lists.digium.com
>Subject: [Asterisk-Users] softswitch dilemma
>
>
>Hello everybody, 
>
>
>Im new to the list and also new to asterisk, Im wondering if 
>I could set up asterisk as a softswitch, I guess for what 
>I've been reading that It could be possible but almost all 
>the info and documentation Ive found so far is about asterisk 
>as a PBX, etc.
>
Asterisk can be setup to do almost anything.  (I once had a box 
setup to make coffee three minutes before it placed a wake-up 
call.) 

As I am sure you will be told from other people we need much 
more data to even think about helping you.

 * Define what you mean by a "softswitch".  As we can see from 
your post to the -biz list there are a few different 
opinions on that.  
 * Analyzing what specific functions/features you need.  

Only then can this list help you.  

Also, please try to research your questions via the wiki and 
google.  Both are an invaluable tool, and many topics have 
already been discussed.

Good luck,

Robert Jackson
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[Asterisk-Users] Ring an incoming call in multiple extensions

[asterisk]

Hi asterisk users!

 

Here’s my issue, I’ve deleted the “s”
extension cause I don’t want any action to be taken on incoming calls as
my pbx is for home use, but I would like to ring all my VoIP extensions at the
same time the PSTN line rings and to be able to pick up the call in any
extension, honestly I don’t know if this is possible, some ideas ???

 

Thanks in advance!






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[Asterisk-Users] softswitch dilemma

[Diego Ventrice]

Hello 
everybody, Im new to the list and also new to asterisk, Im wondering 
if I could set up asterisk as a softswitch, I guess for what I've been reading 
that It could be possible but almost all the info and documentation Ive found so 
far is about asterisk as a PBX, etc.Im willing to set a small voip 
wholesale traffic bussiness and Im not quite sure asterisk is the right chocie 
for that.An asterisk-ser or an asterisk-vocal combination may be the answer 
?Thanks in advance for any 
help.Diego
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Re: [Asterisk-Users] Re: API Call Bridge?

[taf taffey]

Done!

Going back to the issue at hand I've read the reply
from Tony in a previous mail and this is related to
call files. Is there a way to dial two
outbound/external numbers and bridge them together
using the Asterisk API manager method instead??

Cheers,
Taff.
 --- Matt Riddell <[EMAIL PROTECTED]> wrote: 
> taf taffey wrote:
> > Ok thanks.
> > 
> You might want to have a go at turning off html
> emails!
> 
> :)
> 
> -- 
> Cheers,
> 
> Matt Riddell
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> News - html)
> http://www.sineapps.com/rssfeed.php (Daily Asterisk
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[Asterisk-Users] OT: Headset for the Cisco 7960

[Nabeel Jafferali]

Hello.

Short of buying a (no doubt) expensive one designed specifically for the
Cisco 7960, what are my options for using headset with this phone? Is
there some kind of adapter to buy so I can use standard
Plantronics/Jabra headsets? Is there by any chance a Bluetooth adapter -
or should I just buy one of the adapters for the standard headset
connector and then buy the Bluetooth adapter with those connectors?

Any help would be appreciated.

--
Nabeel Jafferali
Tel: +1 (416) 628-9342  Toronto
 +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeeljafferali.net
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RE: [Asterisk-Users] Headset with X-Lite

[Bruno Hertz]

On Thu, 2005-01-20 at 16:59 -0800, Manjit Riat wrote:
> Oh sorry... just got carried away with all the help I got here.

No problem. Don't know about your headset, but usually it has
two connectors, which you plug into the mic and speaker jacks
of the sound card. XLite itself doesn't really care whether you
have a headset or not, you could also connect speakers and a
microphone. In each case, XLite should access the right channels.

You might want to check that your volume settings are in order.
If things really don't work, there is an option menu setting in
XLite where you can check if it properly recognized the sound
card, just don't remember where it was.

Regards, Bruno.


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[Asterisk-Users] Hopping through iax servers

[david kwok]

I am wondering whether hopping through a number of iax servers would 
decrease the call quality.

My iax server connect to PSTN termination provider in Australia. It 
works fine if calls are made to local areas. However, if calling 
internet state there are constant packet loss. This is also the case 
when calling overseas. I presume if calling internet state. The call is 
routed through the registration iax server and route to the interstate 
iax termination. If I am calling overseas, it may route through local 
iax and its international trunk iax and overseas iax since they need it 
for record keeping purposes.

David Kwok
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Re: [Asterisk-Users] Re: Asterisk & QSIG

[Henry Devito]

Good evening.  QSIG is the standard that  ISDN PRI is written on Q.931.  Q 
SIGNALLING refers to the information provided on the D-channel.  This 
information creates ability to have transparent dialing plans and such.   * 
supports most of this functionailty already.  I have this working on a 
Toshiba, Nortel Option 11, Norstar, and Legend PBX now.  The only limits I 
have found is in the legacy PBX's especially the Legend.


- Original Message - 
From: "Sergio Veltri" <[EMAIL PROTECTED]>
To: 
Sent: Thursday, January 20, 2005 8:24 PM
Subject: [Asterisk-Users] Re: Asterisk & QSIG


Dear Marco,
I am also interested in QSIG and Asterisk. I need to have Asterisk
register to a huge Avaya pbx for call center connectivity.
Please let me know if you find out anything.
Best regards,
Sergio Veltri
www.pointhorizon.com
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[Asterisk-Users] Re: Asterisk & QSIG

[Sergio Veltri]

Dear Marco,

I am also interested in QSIG and Asterisk. I need to have Asterisk
register to a huge Avaya pbx for call center connectivity.

Please let me know if you find out anything.

Best regards,

Sergio Veltri
www.pointhorizon.com
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Re: [Asterisk-Users] # Transfers.

[Jim Kou]

No, it's doesn't work.
Asterisk List on 2005/1/21 01:48 wrote:
I have no idea if atxfer works with app_queue/chan_agent.  Can anyone try it?
Best regards,
--JJL44
On Thu, 20 Jan 2005 17:38:25 -, Ben Merrills <[EMAIL PROTECTED]> wrote:
 

Does this work with app_queue/chan_agent?
Cheers,
Ben
--
Jim Kou
IT Engineer
Malico Inc.  Site: http://www.malico.com.tw
No.5, Ming-Lung Road, Yang-Mei, Tao-Yuang, Taiwan 32643
Tel: +886-3-472-8155#2218Fax: +886-3-472-5979
__  ______  ___  _  _  _  ___   
(  \/  )  /__\  (  )  (_  _)/ __)(  _  )  (_  _)( \( )/ __)  
)(  /(__)\  )(__  _)(_( (__  )(_)(_)(_  )  (( (__   
(_/\/\_)(__)(__)()()\___)(_)  ()(_)\_)\___)()

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Re: [Asterisk-Users] Re: Best Grandstream firmware to use?

[Wolfgang S. Rupprecht]

[EMAIL PROTECTED] (Aldo Bergamini) writes:
> [EMAIL PROTECTED] is believed to have said: 
>>http://fm.grandstream.com/gs/
> Thanks!

And thanks from here too!  I've been annoyed at the non-working
message button from 1.0.5.16 for about a month.  It is nice to have
that working again.

In case anyone else is trying to use the version 1.0.5.16 HTTP method
of upgrading the firmware, don't bother.  It doesn't work.  I couldn't
go from *.16 to *.20 until I went back to using TFTP.  The files get
loaded by the grandstream, but it never seems to burn them into
eeprom.  As soon as I set up the tfttp server again and changed the
GS's config, it loaded and burned just fine.

-wolfgang
-- 
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
 Hate software patents?  Sign here: http://thankpoland.info/
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Re: [Asterisk-Users] controlling recording

[Steven Critchfield]

On Thu, 2005-01-20 at 20:25 -0500, John Hammen wrote:
> Thanks for your quick replies - I had no idea this list was so responsive!
> 
> One more issue for you - we are trying to build a custom voicemail
> application and so far everything works great by using AGI to play
> prompts, get input and record files. However one of our customers has
> asked for some pretty advanced functionality, specifically the ability
> to:
> 
> * stop in the middle of recording (so far so good, escape character)
> * fast forward/rewind the recording as it is so far (!)
> * overdub (i.e. rerecord) starting from a desired point in the
> existing sound file (!!!)
> 
> So, it's pretty clear that AGI's "RECORD FILE" function as it stands
> is not going to cut it, so I'm assuming that do anything like this
> means writing a new custom app_something.c, does that sound right?
> Anybody have any tips or comments on how ridiculously difficult this
> might be?


Ahh so you have a client that wants to do some form of dictation. Yes
AGI as it stands can handle it. Look at the docs a bit closer. How do I
know it works, My company already has a dictation system written and n
production use using AGI.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Webmin Module for Asterisk {Scanned}

[Matt Riddell]

Henry Devito wrote:
 
www.thirdlane.com   has already written a 
close dsource webmin module.  I have no idea how much it costs or how 
well it works.
Oh my 22Kb for 3 lines of email!
Just a reminder:
The 10 line plaintext email you replied to was 3Kb
3Kb x 10,000 users = 30Mb
22Kb x 10,000 users = 220Mb
Poor digium servers!
:)
--
Cheers,
Matt Riddell
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[Asterisk-Users] SNOM 190 and dtmf

[Michael Di Martino]

I have the dtmfmode in sip.conf 
set to use rfc 2833
however, when my users have to enter pin numbers to join let say 
someone's
conference bridge the pin is received twice.
 
Any ideas on how to solve 
this?
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Re: [Asterisk-Users] Re: API Call Bridge?

[Matt Riddell]

taf taffey wrote:
Ok thanks.
You might want to have a go at turning off html emails!
:)
--
Cheers,
Matt Riddell
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[Asterisk-Users] controlling recording

[John Hammen]

Thanks for your quick replies - I had no idea this list was so responsive!

One more issue for you - we are trying to build a custom voicemail
application and so far everything works great by using AGI to play
prompts, get input and record files. However one of our customers has
asked for some pretty advanced functionality, specifically the ability
to:

* stop in the middle of recording (so far so good, escape character)
* fast forward/rewind the recording as it is so far (!)
* overdub (i.e. rerecord) starting from a desired point in the
existing sound file (!!!)

So, it's pretty clear that AGI's "RECORD FILE" function as it stands
is not going to cut it, so I'm assuming that do anything like this
means writing a new custom app_something.c, does that sound right?
Anybody have any tips or comments on how ridiculously difficult this
might be?

Thanks again,

John
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Re: [Asterisk-Users] Stumped on LD questions......

[Michael Graves]

On Thu, 20 Jan 2005 17:42:02 -0500, Mike Clark wrote:

>Ty Carter wrote:
>
>>OK.. I'm up to my eyes in LD BS!
>>
>>I can't for the life of me understand how any carrier, either VoIP or
>>traditional service provider can make heads or tails of how to hand off an *
>>based call to an LD provider.  Every provider I talk to, says I have to have
>>a traditional T1 put in to their respective networks.
>>
>>I don't want to do this.  I want a LD provider that can take a IP, SIP, IAX
>>hand-off and terminate the call.  I have talked and received pricing from
>>ATT, MCI and Alliance Telecom.  Any other suggestions  I'm looking for
>>wholesale pricing for termination as we are a CLEC. 
>>
>>Any suggestions would be greatly appreciated!
>>
>>Regards,
>>
>>Ty Carter

>Check out www.voipjet.com. They currently do IAX and I think will be 
>adding SIP  termination.  Don't know if they offer wholesale rates. 1.3 
>cents per minute at no volume commitment.

I second this. I've been very happy with both VoipJet.com and
Sixtel.net. Good rates. No minimum. IAX2 termination. And * friendly
support staff.

Michael

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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Re: [Asterisk-Users] Asterisk 1.0.4 and more ...

[Matt Riddell]

Gabriel Afana wrote:
I second this :-)
1.0.3 is working perfect for me...except for slight sound quality 
problems (jittery) using SIP.  Any new features for me there in 1.0.4?
It's basically all the bugfixes that have appeared in the STABLE version 
of CVS since 1.0.3 was released.  So, if you were downloading the V1-0 
version from CVS regularly, chances are you kinda already have 1.0.4.

If however you downloaded a 1.0.3 package, there will be numerous small 
bugfixes.

The easiest way to see that changes is to download one of the packages 
from the Digium FTP site and read the CHANGELOG.

--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Troubles with Broadvoice (register) {Scanned}

[Paul]

We could modify /etc/hosts and it is also possible to do the same with 
dns. I could do something that makes sure bvproxy.private.lan is always 
returning the IP with the best ping results. Either way we still have 2 
issues to tackle first:

1) As I noted before the host with the best current ping time was not 
working for SIP but switching to one with a higher ping time worked. So 
we actually need to do something SIP-related.

2) Does * check /etc/hosts or do dns lookups every time registration is 
done or an an outbound call is placed?

Additionally, we certainly don't want to be constantly contacting a 
group of servers to determine which one is best to use.

All this makes me think out loud of a better approach, although it is 
one that will probably involve patching the source. For any set of 
registration credentials we need to be able to specify a few things in 
sip.conf(or iax.conf) and let * handle the rest internally after that.

for each [account] context we need a few parameters
h - list of hosts to scan in initial order of preference
mint - min time between rescans to keep provider from getting pissed
maxt - max time between rescans
great - performance level not worth messing with
crappy - performance level that sucks
conditions - current performance level updated by normal call activity 
and normal periodic checks to see if we are still registered.

whenever conditions are great no further scanning is done
whenever conditions are between great and crappy periodic rescanning is 
done to try and find a better host. The period could actually be 
calculated so that rescanning is done less often as conditions improve.

Whenever conditions are crappy rescanning is  done  at  interval  mint
Allowing flexible logging options in the [account] context would be 
helpful in evaluating providers

David Shaw wrote:
Thinking out load here. 

Could we replace the hostname in the sip.conf to something like
"broadvoice". Then have a script ping all the proxys for broadvoice then
write the best IP address in the /etc/hosts?
sip.conf
register => @broadvoice
host file
XXX.XXX.XXX.XXX broadvoice
This should keep us from reloading asterisk
David
Just thinking out load
On Thu, 2005-01-20 at 11:11, Helder RogÃrio [MICROREDE] wrote:
 

It was a problem regarding the register => not being at the top of the
[general] section of sip.conf
- Original Message - 
From: "Helder RogÃrio [MICROREDE]" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, January 20, 2005 4:21 PM
Subject: Re: [Asterisk-Users] Troubles with Broadvoice (register)

   

Hi!
But the only server they gave for sip registration is sip.broadvoice.com I
have several for outbound proxy proxy.chi.broadvoice.com and etc...
Do you have any other for sip?
Best regards,
Helder
- Original Message - 
From: "Paul" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, January 20, 2005 4:15 PM
Subject: Re: [Asterisk-Users] Troubles with Broadvoice (register)

 

Sometimes I have problems and changing to another of their servers makes
it start working again. There probably is a way to make * deal with this
properly. I am using the broadvoice account for test purposes at this
time so I just edit sip.conf and restart * when this happens. What I
have observed is that the server I can't register with will still have
good ping times when this happens.
Helder RogÃrio [MICROREDE] wrote:
   

Hi!
Are you also getting in trouble while trying to register in Broadvoice?
Cumprimentos / Best regards,
Helder RogÃrio
__
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http://www.microrede.pt
***
 There are only two types of people in the world, those who have lost
 

data
 

and those who will. Â
-- Richard Nixon
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Re: [Asterisk-Users] NMI issues...

[Michael Loftis]

I dont' think it's the digium hardware fault.  Why?  I've got probably 8-10 
machines like this and they all get occasional NMIs with the 'uhhuh' 
message.  It's defiently kernel related...hell these boards don't work 
under anything older than 2.4.27.

--On Tuesday, December 21, 2004 00:42 -0600 "I put the Who? in Mishehu" 
<[EMAIL PROTECTED]> wrote:

I have read thru what other users have tried in this list when they have
experienced seemingly similar issues to what I have, without success.  I
suspect there might be an issue regarding both the X100P and TDM04b cards
being used in an Intel SE7525GP2 motherboard, as I had to even wait for a
BIOS update from Intel in order to utilize my 3ware 9500 SATA raid
controller on it (3ware Kb -->
http://www.3ware.com/kb/article.aspx?id=12435 ).  The following is a list
of the configuration and the issues I have encountered.  Any help would be
greatly appreciated.
System A:
Intel SE7525GP2 motherboard.  Has BIOS revision 06, BMC 2.40 and FRUSDR
1.40 2x2.8GHz Xeon processors w/800Mhz FSB, actively cooled retail box set
1GB of ECC registered DDR333 RAM, 2 DIMMs
3ware 9500 SATA Raid, controlling a 3-disk raid5 array
550 Watt 24-pin ATX power supply
System B:
Asus A7N8X-E Deluxe
AMD Athlon XP 3200 (barton core)
512 MB RAM
ATI 9600SE OEM card
200GB SATA drive
Process is as follows:
1.  On system A until further notice: The Intel SE7525GP2 conflicted with
the firmware on the 3ware 9500 SATA RAID controller until BIOS 06 was
released.
2.  After having been flashed, Slackware 10.0 was then installed on it.
It currently runs stock kernel 2.6.9, udev 042, and the most recent
hotplug from 2004-09-23.
3.  Installed a digium-supplied X101P into the machine into a 32bit PCI
slot. 4.  Downloaded zaptel 1.0.3, compiled against kernel 2.6.9.  Upon
modprobe wcfxo, NMI's are immediately reported (syslog to follow).  The
instructions from README.udev were followed and parameters inserted into
the rules file for udev.  Problem occurs regardless of whether or not
hotplug inserts the modules or the are manually inserted by me.  Repeated
modprobe -r wcfxo; modprobe wcfxo produces no different result as well. 5.
 Attempted all combinations of moving the card to other compatible slots,
CVS zaptel from 2004-12-17, and the TDM04b card.  Snippet of syslog as
follows:
Dec 19 15:33:17 nisui kernel: Freshmaker version: 71
Dec 19 15:33:17 nisui kernel: Freshmaker passed register test
Dec 19 15:33:18 nisui kernel: Module 0: Installed -- AUTO FXO (FCC mode)
Dec 19 15:33:18 nisui kernel: Uhhuh. NMI received for unknown reason 31 on
CPU 0.
Dec 19 15:33:18 nisui kernel: Dazed and confused, but trying to continue
Dec 19 15:33:18 nisui kernel: Do you have a strange power saving mode
enabled?
Dec 19 15:33:18 nisui kernel: Module 1: Installed -- AUTO FXO (FCC mode)
Dec 19 15:33:18 nisui kernel: Module 2: Installed -- AUTO FXO (FCC mode)
Dec 19 15:33:19 nisui kernel: Module 3: Installed -- AUTO FXO (FCC mode)
Dec 19 15:33:19 nisui kernel: Found a Wildcard TDM: Wildcard TDM400P REV
E/F (4 modules)
Dec 19 15:33:19 nisui kernel: Uhhuh. NMI received for unknown reason 31 on
CPU 0.
Dec 19 15:33:19 nisui kernel: Dazed and confused, but trying to continue
Dec 19 15:33:19 nisui kernel: Do you have a strange power saving mode
enabled?
Dec 19 15:33:19 nisui kernel: Uhhuh. NMI received for unknown reason 21 on
CPU 0.
The NMI's will continue to report reasons 21 & 31 until the module is
removed from the kernel.
6.  Attempted several kernel reconfigurations, including ones that
stripped out USB support and all SMP support.  Even on this machine with a
uniprocessor kernel, NMI was still generated.
7.  Took the X100P and installed it in System B.  System B only reports:
Notice: Configuration file is /etc/zaptel.conf
line 143: Unable to open master device '/dev/zap/ctl'
Whenever I check, /dev/zap/ctl does exist.  No NMI is generated on this
machine, though I do not know if it is capible of doing so.  Otherwise, it
appears to operate normally.
8.  Tried asterisklivecd that is maintained by somebody in Italy.
Actually got a freshmaker error on that trial, though it appeared to be
somewhat outdated.
Just to anticipate questions, whenever the TDM04b was tested, the power
connector was in fact connected.  An Intel Etherexpress Pro 100 ethernet
card was tested on Machine A to verify that there was no problem with the
slots.  The eepro100 initializes and operates perfectly.  Attached to this
email is a copy of the kernel's configuration, and a list of currently
loaded modules is as follows:
Module  Size  Used by
md5 4992  1
ipv6  260352  16
hw_random   6548  0
pciehp 97540  0
shpchp101124  0
pci_hotplug13060  2 pciehp,shpchp
e1000  86788  0
evdev  10368  0
And output from /proc/interrupts immediately after loading the wctdm
module:   CPU0   CPU1   CPU2   CPU3
 0:3791068 

RE: [Asterisk-Users] Headset with X-Lite

[Manjit Riat]

Oh sorry... just got carried away with all the help I got here.

-Original Message-
From: Bruno Hertz [mailto:[EMAIL PROTECTED] 
Sent: Thursday, January 20, 2005 3:02 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Headset with X-Lite

On Thu, 2005-01-20 at 14:51 -0800, Manjit Riat wrote:
> Just got a headset for testing asterisk and am using X-Lite. I plugged
> in the headset into the headset jack and is there any way to configure
> X-lite to use the headset instead of the speakers? Or will I have to
> plug the headset in the speaker jack ?

Manjit

a delicate question, but are you sure that this is an asterisk issue?
Because, and I'm confident you won't mind me being frank, this rather
sounds like being at most an XLite question, if not only an issue about
how to properly connect your headset. Anyway, here's the link to the
XLite support forum: http://support.xten.net/
I wish you good luck there.

Regards, Bruno.





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Re: [Asterisk-Users] Re: zaptel on 2.6.10 kernel - debian.

[Nick Bachmann]

[EMAIL PROTECTED] wrote:
Hi Guys, Gals.
Ok, so I have latest CVS sources on a debian box, 2.6.10-1-386 kernel
kernel headers isntalled in the right plauce and all that stuff .. but 
whatever I try .. same results, I only need to get ztdummy working for 
a conference .. but I always end up stuffed :(
heres the compile:
Call off the hounds, I think the problem is fixed in M3391...
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[Asterisk-Users] Re: zaptel on 2.6.10 kernel - debian.

[list]

Hi Guys, Gals. 

Ok, so I have latest CVS sources on a debian box, 2.6.10-1-386 kernel 

kernel headers isntalled in the right plauce and all that stuff .. but 
whatever I try .. same results, I only need to get ztdummy working for a 
conference .. but I always end up stuffed :( 

heres the compile: 

[EMAIL PROTECTED]:~/zaptel$ make linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"makefw.c   -o makefw
./makefw tormenta2.rbt tor2fw > tor2fw.h
Loaded 69900 bytes from file
./makefw pciradio.rbt radfw > radfw.h
Loaded 42096 bytes from file
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o zonedata.lo 
zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -DBUILDING_TONEZONE -o tonezone.lo 
tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o -lm -L. libtonezone.a
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o torisatool.o torisatool.c
cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o ztmonitor.o ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"zttest.c   -o zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o fxotune.o fxotune.c
cc -o fxotune fxotune.o
make -C /lib/modules/`uname -r`/build SUBDIRS=/home/robin/zaptel modules
make[1]: Entering directory `/usr/src/kernel-headers-2.6.10-1-386'
CC [M]  /home/robin/zaptel/ztdummy.o
Building modules, stage 2.
MODPOST
*** Warning: "cleanup_module" [/home/robin/zaptel/ztdummy.ko] undefined!
*** Warning: "init_module" [/home/robin/zaptel/ztdummy.ko] undefined!
*** Warning: "zt_transmit" [/home/robin/zaptel/ztdummy.ko] undefined!
*** Warning: "zt_receive" [/home/robin/zaptel/ztdummy.ko] undefined!
*** Warning: "zt_unregister" [/home/robin/zaptel/ztdummy.ko] undefined!
*** Warning: "zt_register" [/home/robin/zaptel/ztdummy.ko] undefined!
CC  /home/robin/zaptel/ztdummy.mod.o
LD [M]  /home/robin/zaptel/ztdummy.ko
make[1]: Leaving directory `/usr/src/kernel-headers-2.6.10-1-386' 

and the modprobe: 

debian:/home/robin/zaptel# modprobe ztdummy
WARNING: /etc/modprobe.d/zaptel line 17: ignoring bad line starting with 
'post-install'
WARNING: Error inserting zaptel (/lib/modules/2.6.10-1-386/misc/zaptel.ko): 
Invalid module format
FATAL: Error inserting ztdummy (/lib/modules/2.6.10-1-386/misc/ztdummy.ko): 
Invalid module format 

clues?
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Re: [Asterisk-Users] Asterisk 1.0.4 and more ...

[Gabriel Afana]

I second this :-)
1.0.3 is working perfect for me...except for slight sound quality problems 
(jittery) using SIP.  Any new features for me there in 1.0.4?

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

Sent: Thursday, January 20, 2005 3:58 PM
Subject: Re: [Asterisk-Users] Asterisk 1.0.4 and more ...


Hi, Do you have a suggestion on how to determine fixes, features and 
changes between 1.0.3 and 1.0.4 ?

- Original Message - 
From: "Russell Bryant" <[EMAIL PROTECTED]>
To: ; 
<[EMAIL PROTECTED]>
Sent: Thursday, January 20, 2005 6:42 PM
Subject: [Asterisk-Users] Asterisk 1.0.4 and more ...


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Greetings!
Version 1.0.4 of Asterisk, Asterisk-addons, zaptel, and libpri has been
released.  The releases are available on the Digium ftp site.
I have also started a web site called "The Asterisk Development Proxy"
where I would like to start centralizing Asterisk development resources.
~ I am hoping that this will become valuable for those getting into
Asterisk development.
http://dev.asteriskdocs.org/
As a part of this effort, I have started a development team called "The
Asterisk Maintenance Crew" to encourage amateur programmers to get
involved in development.  The number one job of this team is to help
with the maintenance of the stable branch.  We will also help to
identify and complete some easier development tasks that can be
completed for CVS head.  There is plenty of work to be done, and
programmers of any skill level can help.  If you have any desire to get
involved in Asterisk development, but aren't sure how to get started,
please consider getting involved.
I would like to send a special thanks to Andrew Thompson for providing
the web hosting, and to Leif Madsen (blitzrage) for his early
contributions to the web site and the maintenance crew.  Josh Colp
(file) has volunteered to help with the ChangeLog, which is going to be
quite a task for Asterisk 1.2, so someone please send him a muffin.
That's all for now.  Feel free to contact me with any questions or 
comments.

Russell Bryant
"drumkilla"
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.0 (GNU/Linux)
iD8DBQFB8EH9rwroOS5t/FoRAj8bAJ9JmjWFN8LeZUqSl8kC2VZdG4AiYgCdE88v
dG13ONgdu/HqBLy6gEu3THs=
=t7j1
-END PGP SIGNATURE-
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Re: [Asterisk-Users] Asterisk 1.0.4 and more ...

[Jon Bebeau]

Hi, Do you have a suggestion on how to determine fixes, features and changes 
between 1.0.3 and 1.0.4 ?

- Original Message - 
From: "Russell Bryant" <[EMAIL PROTECTED]>
To: ; <[EMAIL PROTECTED]>
Sent: Thursday, January 20, 2005 6:42 PM
Subject: [Asterisk-Users] Asterisk 1.0.4 and more ...


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Greetings!
Version 1.0.4 of Asterisk, Asterisk-addons, zaptel, and libpri has been
released.  The releases are available on the Digium ftp site.
I have also started a web site called "The Asterisk Development Proxy"
where I would like to start centralizing Asterisk development resources.
~ I am hoping that this will become valuable for those getting into
Asterisk development.
http://dev.asteriskdocs.org/
As a part of this effort, I have started a development team called "The
Asterisk Maintenance Crew" to encourage amateur programmers to get
involved in development.  The number one job of this team is to help
with the maintenance of the stable branch.  We will also help to
identify and complete some easier development tasks that can be
completed for CVS head.  There is plenty of work to be done, and
programmers of any skill level can help.  If you have any desire to get
involved in Asterisk development, but aren't sure how to get started,
please consider getting involved.
I would like to send a special thanks to Andrew Thompson for providing
the web hosting, and to Leif Madsen (blitzrage) for his early
contributions to the web site and the maintenance crew.  Josh Colp
(file) has volunteered to help with the ChangeLog, which is going to be
quite a task for Asterisk 1.2, so someone please send him a muffin.
That's all for now.  Feel free to contact me with any questions or 
comments.

Russell Bryant
"drumkilla"
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.0 (GNU/Linux)
iD8DBQFB8EH9rwroOS5t/FoRAj8bAJ9JmjWFN8LeZUqSl8kC2VZdG4AiYgCdE88v
dG13ONgdu/HqBLy6gEu3THs=
=t7j1
-END PGP SIGNATURE-
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RE: [Asterisk-Users] H323 and ASTCC

[Kanuri, Seshu (Company IT)]

You can do that in your extensions.conf for the context 
you are using for astcc originated calls
 
Seshu Kanuri


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Krystian 
FiliksSent: Thursday, January 20, 2005 5:43 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] H323 and 
ASTCC


I got My ASTCC kind of working, but 
the problem I have is that it tries to send all the calls over 
SIP.
How can I configure it with 
H323?
 
Thanks
KF




NOTICE: If received in error, please destroy and notify sender.  Sender does not waive confidentiality or privilege, and use is prohibited.

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[Asterisk-Users] Asterisk 1.0.4 and more ...

[Russell Bryant]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Greetings!
Version 1.0.4 of Asterisk, Asterisk-addons, zaptel, and libpri has been
released.  The releases are available on the Digium ftp site.
I have also started a web site called "The Asterisk Development Proxy"
where I would like to start centralizing Asterisk development resources.
~ I am hoping that this will become valuable for those getting into
Asterisk development.
http://dev.asteriskdocs.org/
As a part of this effort, I have started a development team called "The
Asterisk Maintenance Crew" to encourage amateur programmers to get
involved in development.  The number one job of this team is to help
with the maintenance of the stable branch.  We will also help to
identify and complete some easier development tasks that can be
completed for CVS head.  There is plenty of work to be done, and
programmers of any skill level can help.  If you have any desire to get
involved in Asterisk development, but aren't sure how to get started,
please consider getting involved.
I would like to send a special thanks to Andrew Thompson for providing
the web hosting, and to Leif Madsen (blitzrage) for his early
contributions to the web site and the maintenance crew.  Josh Colp
(file) has volunteered to help with the ChangeLog, which is going to be
quite a task for Asterisk 1.2, so someone please send him a muffin.
That's all for now.  Feel free to contact me with any questions or comments.
Russell Bryant
"drumkilla"
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.0 (GNU/Linux)
iD8DBQFB8EH9rwroOS5t/FoRAj8bAJ9JmjWFN8LeZUqSl8kC2VZdG4AiYgCdE88v
dG13ONgdu/HqBLy6gEu3THs=
=t7j1
-END PGP SIGNATURE-
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Re: [Asterisk-Users] iax encryption

[Steve Kann]

John Hammen wrote:
Hi All,
I was wondering if there is any way to encrypt IAX traffic? I am aware
of the ability to use md5 or RSA for authentication, but I'm talking
about the packets themselves, after authorization has already
occured...
Forgive me if this is documented somewhere,  but I all I could find
online was a presentation with the statement "there has been talk of
adding encryption to the IAX protocol". Does anyone know the current
status of this i.e. is there any active developement going on?
 

There's some code to do this in CVS, but it is incomplete.
If you want something that works right now, just tunnel your stuff 
through a VPN, like openVPN or something..

-SteveK
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Re: [Asterisk-Users] iax encryption

[Brian Capouch]

John Hammen wrote:
Hi All,
I was wondering if there is any way to encrypt IAX traffic? I am aware
of the ability to use md5 or RSA for authentication, but I'm talking
about the packets themselves, after authorization has already
occured...
Forgive me if this is documented somewhere,  but I all I could find
online was a presentation with the statement "there has been talk of
adding encryption to the IAX protocol". Does anyone know the current
status of this i.e. is there any active developement going on?
There's native IAX encryption in CVS-HEAD right now.
I haven't had a chance to dig in to the source code to infer how to set 
things up.

Perhaps someone more clueful could post a little HOWTO?
It's obviously a very, very important feature.
B.
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[Asterisk-Users] Asterisk & QSIG

[Marco Vescovi]

Hi all,
reading around and surfing the net I've found some informations about QSIG
PRI protocol, that seems a good choice to integrate 2 PBX systems with PRI
interfaces. The question is: which is the state of Asterisk support for that
protocol ? I was wondering if I could link a traditional PBX system to
Asterisk with a QSIG PRI interface ...
 
Thanks a lot.
 
marco

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No virus found in this outgoing message.
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[Asterisk-Users] iax encryption

[John Hammen]

Hi All,

I was wondering if there is any way to encrypt IAX traffic? I am aware
of the ability to use md5 or RSA for authentication, but I'm talking
about the packets themselves, after authorization has already
occured...

Forgive me if this is documented somewhere,  but I all I could find
online was a presentation with the statement "there has been talk of
adding encryption to the IAX protocol". Does anyone know the current
status of this i.e. is there any active developement going on?

Thanks in advance for any and all pointers...
-John
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Re: [Asterisk-Users] PIX!!!!!

[Chris Tuska]

Chris,
 
I have a PIX with  6.3(4) and I have my Asterisk 
behind a NAT works great.  What I did have to do is shutdown all NAT config 
in Asterisk.  What was happening is Asterisk was putting my public IP on 
the packets going out of my linux box and the PIX was trying to rewrite those 
packets.  
 
Here is what I removed in the sip.conf 
file
 
;externip = XXX.XXX.XXX.XXX ; 
Address that we're going to put in SIP 
;nat=yes
 
Chris Tuska
 
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Re: [Asterisk-Users] Re: API Call Bridge?

[taf taffey]

Ok thanks.Tony Mountifield <[EMAIL PROTECTED]> wrote:
In article <[EMAIL PROTECTED]>,taf taffey <[EMAIL PROTECTED]>wrote:> -=-=-=-=-=-> -=-=-=-=-=-> > Hi All,> Does anyone know of a way to dial two different outbound numbers and bridge> them together using the Asterisk API?I answered exactly that question on this list within the last two days.Should be easy to find as it's so recent.CheersTony-- Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: [EMAIL PROTECTED] - http://tony.mountifield.org___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options
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		 ALL-NEW 
Yahoo! Messenger 
- all new features - even more fun! 
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Re: [Asterisk-Users] Headset with X-Lite

[Bruno Hertz]

On Thu, 2005-01-20 at 14:51 -0800, Manjit Riat wrote:
> Just got a headset for testing asterisk and am using X-Lite. I plugged
> in the headset into the headset jack and is there any way to configure
> X-lite to use the headset instead of the speakers? Or will I have to
> plug the headset in the speaker jack ?

Manjit

a delicate question, but are you sure that this is an asterisk issue?
Because, and I'm confident you won't mind me being frank, this rather
sounds like being at most an XLite question, if not only an issue about
how to properly connect your headset. Anyway, here's the link to the
XLite support forum: http://support.xten.net/
I wish you good luck there.

Regards, Bruno.


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[Asterisk-Users] Headset with X-Lite

[Manjit Riat]

Just got a headset for testing asterisk and am using X-Lite. I plugged in the headset into the headset jack and is
there any way to configure X-lite to use the headset
instead of the speakers? Or will I have to plug the headset in the speaker jack
?






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RE: [Asterisk-Users] Passing PIN Numbers

[Michael Di Martino]

Title: Passing PIN Numbers



I have the dtmfmode in sip.conf 
set to use rfc 2833
however, when my users have to enter pin numbers to join let say 
someone's
conference bridge the pin is received twice.
 
Any ideas on how to solve this?


From: Rene Kluwen [mailto:[EMAIL PROTECTED] 
Sent: Monday, January 17, 2005 1:41 PMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
Passing PIN Numbers

This is a long shot, I am not sure if it will solve 
your problem:
 
Did you try to change dtmfmode in 
sip.conf?
 
Rene Kluwen
Chimit
 

  - Original Message - 
  From: 
  Michael Di Martino 
  To: asterisk-users@lists.digium.com 
  
  Sent: Friday, January 14, 2005 3:53 
  PM
  Subject: [Asterisk-Users] Passing PIN 
  Numbers
  
  To All If anyone can shed any light on this 
  it would be greatly appreciated. My phones are unable to enter pins numbers correctly when 
  required by the party they are calling. 
  For example I was given an 
  outside number to attend conference bridge. After the call was connected it 
  required me to enter a 4 digit PIN. Now here is the problem whenever I enter a 
  pin it is received twice. For example if the PIN is 1234 they receive it as 
  12341234.
  Any ideas what could be 
  wrong? 
  BTW we are using SNOM 190 ip phones 
  (sip) 
  Regards, Michael DiMartino Director of MIS The telx Group, Inc. 17 State St, 33rd Floor New York, NY 10004 T: 212.480.3300 X2022 C: 646.207.6603   
  
  

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[Asterisk-Users] Asterisk@Home and iax.cc / sixTel

[Lee]

Hi,

How is iax.cc / sixTel to be configured as a termination provider in
[EMAIL PROTECTED]

The iax.cc / sixTel instructions tell you to do this:

iax.conf:

[sixTel]
type= friend
host= iax2.sixtel.net
context = inbound
secret  = mypassword
allow   = all

extensions.conf:

exten => _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten => _011.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})


How is this to be configured in [EMAIL PROTECTED] I'm leary of changing
.conf files directly because [EMAIL PROTECTED] supposedly does that for
you.

-- 
Lee
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[Asterisk-Users] H323 and ASTCC

[Krystian Filiks]

I got My ASTCC kind of working, but the problem I have is
that it tries to send all the calls over SIP.

How can I configure it with H323?

 

Thanks

KF






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Re: [Asterisk-Users] Stumped on LD questions......

[Mike Clark]

Ty Carter wrote:
OK.. I'm up to my eyes in LD BS!
I can't for the life of me understand how any carrier, either VoIP or
traditional service provider can make heads or tails of how to hand off an *
based call to an LD provider.  Every provider I talk to, says I have to have
a traditional T1 put in to their respective networks.
I don't want to do this.  I want a LD provider that can take a IP, SIP, IAX
hand-off and terminate the call.  I have talked and received pricing from
ATT, MCI and Alliance Telecom.  Any other suggestions  I'm looking for
wholesale pricing for termination as we are a CLEC. 

Any suggestions would be greatly appreciated!
Regards,
Ty Carter



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Check out www.voipjet.com. They currently do IAX and I think will be 
adding SIP  termination.  Don't know if they offer wholesale rates. 1.3 
cents per minute at no volume commitment.
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RE: [Asterisk-Users] Stumped on LD questions......

[Robert Jackson]

> -Original Message-
> From: Ty Carter [mailto:[EMAIL PROTECTED] 
> Sent: Thursday, January 20, 2005 5:23 PM
> To: 'Asterisk Users Mailing List - Non-Commercial 
> Discussion'; [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Stumped on LD questions..
> 
> 
> OK.. I'm up to my eyes in LD BS!
> 
> I can't for the life of me understand how any carrier, either 
> VoIP or traditional service provider can make heads or tails 
> of how to hand off an * based call to an LD provider.  Every 
> provider I talk to, says I have to have a traditional T1 put 
> in to their respective networks.
> 
> I don't want to do this.  I want a LD provider that can take 
> a IP, SIP, IAX hand-off and terminate the call.  I have 
> talked and received pricing from ATT, MCI and Alliance 
> Telecom.  Any other suggestions  I'm looking for 
> wholesale pricing for termination as we are a CLEC. 
> 

Please do not cross post.  A lot of people belong to both lists 
and as a result now have to look at the same e-mail twice.  

I responded to -biz originally.

Robert Jackson
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Re: [Asterisk-Users] VoIP-to-TDM processing on-card?

[Steven Critchfield]

On Thu, 2005-01-20 at 17:15 -0500, Dana Olson wrote:
> On Thu, 20 Jan 2005 15:26:15 -0600, Steven Critchfield
> <[EMAIL PROTECTED]> wrote:
> > While it will probably be handled when you move out of outlook, please
> > wrap your lines at a reasonable length.
> 
> Please tell me that Gmail is fine... If it isn't, I'll have to find
> something else.

So far so good. Others around here use gmail. I just have no use for
their user agreement. But then again, I have my own domain to fall back
to if I so choose. 

> > It wasn't so much that your question was bad, but it didn't show that
> > you had the proper understanding of the question. If you had stated
> > which sites/URLs you had searched through to come to the conclusion that
> > there might be a reason the list would be more authoritative that those
> > URLs, it would have shown effort and therefore reason to be respected.
> > You will find that even in supposedly rough groups, effort is respected.
> > Few like freeloaders. Your question seemed like it was only about you
> > having others do your work. The extra couple of lines you would have
> > typed to show a bit of your previous effort would have sufficed to
> > eliminate that appearance.
> 
> I understand, but the hardware list and the wiki I figured were known
> by all. Anyway, I'll be sure to do that in the future.

Well... many show up here with out doing any work at all. Those links
should be known by all.

> > When it comes down to it, reasonable hardware should handle decent
> > amounts of codec translations. If you are trying to stuff more than the
> > suggested amount of TE hardware in a box and do codec translation, then
> > you need to rethink the cost of failure. If you are dependant on 12 T1s
> > (value pulled from thin air, not necessarily related), You should see
> > about spitting it out over 3 boxes so at most you only lose 4 T1s at a
> > time. For most companies that rely on the phones, losing 1/3rd of the
> > production is pretty expensive, losing  all of production is not
> > tolerable. Any card that does the codec translation for you will
> > probably make you more likely to consolidate too many interfaces into
> > one machine.
> 
> Yeah, that's why I was hoping there would be cards that do the work.
> If it comes down to it, I'll just go that route and build many boxes
> with less TE cards.

You know you shouldn't exceed 2 cards in a single machine. And with
machines being soo cheap right now, you really should consider not going
more than 1 card per machine. 

My company just bought another Dell to hold a T100P card for just over
$700. 2.4ghz P4, half gig ram, 160gig sata drive, rack mount 1u. Granted
that machine will be devoted to a single customer of ours, I'm sure it
could handle more than a single T1 interface and do conversions just
fine. 

> Anyway, I checked all around in the options of Gmail and I don't see
> anywhere to turn on or off HTML email, and I don't see anywhere that
> mentions linewrapping. If there is a problem, feel free to contact me
> off-list and let me know. I'll try my best to fix whatever is wrong.

gmail seems to have done the trick
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] Stumped on LD questions......

[Robert Jackson]

> -Original Message-
> From: Ty Carter [mailto:[EMAIL PROTECTED] 
> Sent: Thursday, January 20, 2005 5:23 PM
> To: 'Asterisk Users Mailing List - Non-Commercial 
> Discussion'; [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Stumped on LD questions..
> 
> 
> I don't want to do this.  I want a LD provider that can take 
> a IP, SIP, IAX hand-off and terminate the call.  I have 
> talked and received pricing from ATT, MCI and Alliance 
> Telecom.  Any other suggestions  I'm looking for 
> wholesale pricing for termination as we are a CLEC. 
> 
> Any suggestions would be greatly appreciated!
> 

We use LiveVoIP.  They are very good to work with, and have 
great pricing.  Their website is livevoip.com, but I would 
contact them via e-mail first.  

Good luck,

Robert Jackson
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Re: [Asterisk-Users] top-notch servers/OS/network, ulaw codec - soundstill choppy

[Gabriel Afana]

What options are there at all with Asterisk that I can play with to *tweak* 
it to try to optimize sound?  I've already check and played with the QoS 
(0x10, 0x18..etc)...no difference.

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

Sent: Thursday, January 20, 2005 1:18 PM
Subject: Re: [Asterisk-Users] top-notch servers/OS/network, ulaw codec - 
soundstill choppy


- Original Message - 
I see the sip user is an external ip.  I would take a look at your QoS
settings on your router.  Make sure the voice traffic is getting the
priority it deserves.  Also, check for packet loss.
I'd still be wondering if there's something else.  I, too, experience 
choppy SIP connectivity from external IPs, but as I've mentioned in 
previous postings, I have a Vonage ATA that seems to have no problems 
keeping a crystal clear connection as it leaves my place and goes to 
Vonage's servers, so I think there must be more to it than QoS.   I have 
to believe that there's some more jitter correction or other such 
buffering that could berhaps be played with, though I don't know what it 
would be *shrug*.?

regards,
Paul


On Thu, 20 Jan 2005 11:26:02 -0800, Gabriel Afana <[EMAIL PROTECTED]> 
wrote:
Hi,
   My SIP calls are sounding a little choppy.  I've did my research but
everything looks right on my end...what am I missing?
Running RedHat ES 3.0 on dual AMD Opteron servers.  My system is 
cololocated
in downtown LA and is fed via a gigabit handoff from XO, AT&T, Level 3 
and
Wiltel (I have a 100Mb didicated line).  So I dont think its the 
Servers,
its the network, Asterisk is working fine and all codecs look 
right...what
could be the cause?

**SNIP FROM SIP.CONF***
[general]
context=default ; Default context for incoming calls
port=5060   ; UDP Port to bind to (SIP standard port 
is
5060)
;bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds 
to
all)
srvlookup=yes   ; Enable DNS SRV lookups on outbound 
calls
   ; Note: Asterisk only uses the first host
   ; in SRV records

allow=ulaw  ; Allow codecs in order of preference
*
ga0*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)   Format
64.201.99.2479092479878  2fd496bf330  00103/00105   ulaw
ga0*CLI> show version
Asterisk 1.0.3 built by [EMAIL PROTECTED] on a x86_64 running Linux
P.S.  in my sip.conf file, it looks like I am only allowing the ulaw
codec...could that cause a problem if I happen to need to call somebody 
that
doesn't support ulaw?

Gabe
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Is it something someone said, was it something someone said?
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Re: [Asterisk-Users] PIX!!!!!

[Christopher]

Thanks guys, really appreciate the responses.  

Actually I've tried the suggestions in this document with absolutely no
luck at all unfortunately, and turning off fixup protocol udp sip was
the key to allowing my remote phone to ring to an internal phone (when
fixup is on I can see the remote phone, but it will not ring the
internal phones).  But no matter what the fixup featured is set to *
still shows that phone as "Unreachable" and the port number as 0.

Tenorio, Leandro wrote:

  Chris,
	I suggest the same, but in case you want to use the fixup
feature
http://www.cisco.com/en/US/products/hw/vpndevc/ps2030/products_configura
tion_example09186a00801fc74a.shtml

LTenorio

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Julio
Arruda
Sent: Thursday, January 20, 2005 7:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PIX!

Christopher wrote:
  
  
Can anyone point me in a good direction for configuring SIP through a 
PIX using 1:1 NAT.  I have read anything I could get my hands on and 
tried them all with very little success.  I can get it to work through

  
  
  
  
the cheap little cable modem routers, but not this PIX.
I -can- make a direct SIP call using the IP address of the * server 
([EMAIL PROTECTED]), but when I do that * still doesn't show it 
registering.  Even when I call through this method the phone comes up 
as "UNREACHABLE" and the port is listed as 0 instead of 5060 like all 
the internal phones.

  
  
I seem to recall some weird thing in the PIX, where you had to disable
the SIP fixup to work (and of course, to use some nat traversal trick,
like outbound proxy).
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-- 
B2 Technologies, LLC
- www.VoIPSupply.com
- www.ValueResale.com
454 Sonwil Drive
Buffalo, NY 14225
 
(716) 630-1555 x.27
(716) 250-3411 (Direct)
(716) 630-1548 fax
 
We sell all things VoIP related at: http://www.VoipSupply.com
Buy online, same day shipping on most items, volume discounts are available.



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Re: [Asterisk-Users] PIX!!!!!

[Vulture]

I have some cisco 7940's having calls initiated to them from a public address
with 1:1 nat and also are set to my asterisk machine over a IPsec tunnel.

Below are the applicable lines for my outside initiates call to phone inside:

You may need to take the no off depending on how your phones are setup in
reguards to external NAT IP. My phones have their real external IP set into
them.

no fixup protocol sip 5060
no fixup protocol sip udp 5060


You will need the access-list line for the accees-group outside - but shouldn't
need the inside one unless you are limiting outgoing traffic also. You could
break this into multiple lines for different ports/port ranges, but I am just
allowing all traffic from the host to reach the phone. You could probably also
just get away with allowing all/only udp from outside to the phone.


access-list from-outside permit ip host sip_endpoint_ip_here any
access-group from-outside in interface outside

access-list from-inside permit ip 192.168.3.0 255.255.255.0 host
216.136.148.193
access-group from-inside in interface inside


static (inside,outside) external_ip internal_ip netmask 255.255.255.255 0 0

Hope this helps

-Jon


Quoting Christopher <[EMAIL PROTECTED]>:

> Can anyone point me in a good direction for configuring SIP through a 
> PIX using 1:1 NAT.  I have read anything I could get my hands on and 
> tried them all with very little success.  I can get it to work through 
> the cheap little cable modem routers, but not this PIX. 
> 
> I -can- make a direct SIP call using the IP address of the * server 
> ([EMAIL PROTECTED]), but when I do that * still doesn't show it 
> registering.  Even when I call through this method the phone comes up as 
> "UNREACHABLE" and the port is listed as 0 instead of 5060 like all the 
> internal phones.
> 
> -- 
> B2 Technologies, LLC
> - www.VoIPSupply.com
> - www.ValueResale.com
> 454 Sonwil Drive
> Buffalo, NY 14225
>  
> (716) 630-1555 x.27
> (716) 250-3411 (Direct)
> (716) 630-1548 fax
>  
> We sell all things VoIP related at: http://www.VoipSupply.com
> Buy online, same day shipping on most items, volume discounts are available.
> 
> ___
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This message was sent using IMP, the Internet Messaging Program.
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[Asterisk-Users] Stumped on LD questions......

[Ty Carter]

OK.. I'm up to my eyes in LD BS!

I can't for the life of me understand how any carrier, either VoIP or
traditional service provider can make heads or tails of how to hand off an *
based call to an LD provider.  Every provider I talk to, says I have to have
a traditional T1 put in to their respective networks.

I don't want to do this.  I want a LD provider that can take a IP, SIP, IAX
hand-off and terminate the call.  I have talked and received pricing from
ATT, MCI and Alliance Telecom.  Any other suggestions  I'm looking for
wholesale pricing for termination as we are a CLEC. 

Any suggestions would be greatly appreciated!

Regards,

Ty Carter







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Re: [Asterisk-Users] VoIP-to-TDM processing on-card?

[Dana Olson]

On Thu, 20 Jan 2005 15:26:15 -0600, Steven Critchfield
<[EMAIL PROTECTED]> wrote:
> While it will probably be handled when you move out of outlook, please
> wrap your lines at a reasonable length.

Please tell me that Gmail is fine... If it isn't, I'll have to find
something else.
 
> As of right now, I don't think the sangoma card supports any codec
> conversions and asterisk support at the same time. I say this as it
> supports the zaptel api and therefore should work similarly as the TE
> cards from Digium as far as asterisk is concerned.
> 
> Also I don't see that the Wanpipe hardware supports codec translations
> either.
> 
> I may be wrong on that though.

The AudioCode device seems to be the closest thing, but that's a
gateway. Still looking into it though, as well as what the Sangoma
offerings can do.

> It wasn't so much that your question was bad, but it didn't show that
> you had the proper understanding of the question. If you had stated
> which sites/URLs you had searched through to come to the conclusion that
> there might be a reason the list would be more authoritative that those
> URLs, it would have shown effort and therefore reason to be respected.
> You will find that even in supposedly rough groups, effort is respected.
> Few like freeloaders. Your question seemed like it was only about you
> having others do your work. The extra couple of lines you would have
> typed to show a bit of your previous effort would have sufficed to
> eliminate that appearance.

I understand, but the hardware list and the wiki I figured were known
by all. Anyway, I'll be sure to do that in the future.

> I'm not sure the disclaimer itself shows anything other than a stupid
> policy by your employer. At some point your employer should actually
> seek legal advice as to whether or not the disclaimer does any good at
> all. I could disagree with it and therefore not be bound by it. As the
> majority of the world isn't in any form of business agreements with your
> employer, there isn't much you could do to compel others to abide by it.

I'm no lawyer, and can't comment on that. I just find it sorta
annoying when I send email to personal contacts from work and they get
the disclaimer.

> When it comes down to it, reasonable hardware should handle decent
> amounts of codec translations. If you are trying to stuff more than the
> suggested amount of TE hardware in a box and do codec translation, then
> you need to rethink the cost of failure. If you are dependant on 12 T1s
> (value pulled from thin air, not necessarily related), You should see
> about spitting it out over 3 boxes so at most you only lose 4 T1s at a
> time. For most companies that rely on the phones, losing 1/3rd of the
> production is pretty expensive, losing  all of production is not
> tolerable. Any card that does the codec translation for you will
> probably make you more likely to consolidate too many interfaces into
> one machine.

Yeah, that's why I was hoping there would be cards that do the work.
If it comes down to it, I'll just go that route and build many boxes
with less TE cards.

Anyway, I checked all around in the options of Gmail and I don't see
anywhere to turn on or off HTML email, and I don't see anywhere that
mentions linewrapping. If there is a problem, feel free to contact me
off-list and let me know. I'll try my best to fix whatever is wrong.

--
Dana
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RE: [Asterisk-Users] PIX!!!!!

[Tenorio, Leandro]

Chris,
I suggest the same, but in case you want to use the fixup
feature
http://www.cisco.com/en/US/products/hw/vpndevc/ps2030/products_configura
tion_example09186a00801fc74a.shtml

LTenorio

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julio
Arruda
Sent: Thursday, January 20, 2005 7:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PIX!

Christopher wrote:
> Can anyone point me in a good direction for configuring SIP through a 
> PIX using 1:1 NAT.  I have read anything I could get my hands on and 
> tried them all with very little success.  I can get it to work through

> the cheap little cable modem routers, but not this PIX.
> I -can- make a direct SIP call using the IP address of the * server 
> ([EMAIL PROTECTED]), but when I do that * still doesn't show it 
> registering.  Even when I call through this method the phone comes up 
> as "UNREACHABLE" and the port is listed as 0 instead of 5060 like all 
> the internal phones.

I seem to recall some weird thing in the PIX, where you had to disable
the SIP fixup to work (and of course, to use some nat traversal trick,
like outbound proxy).
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Re: [Asterisk-Users] PIX!!!!!

[Julio Arruda]

Christopher wrote:
Can anyone point me in a good direction for configuring SIP through a 
PIX using 1:1 NAT.  I have read anything I could get my hands on and 
tried them all with very little success.  I can get it to work through 
the cheap little cable modem routers, but not this PIX.
I -can- make a direct SIP call using the IP address of the * server 
([EMAIL PROTECTED]), but when I do that * still doesn't show it 
registering.  Even when I call through this method the phone comes up as 
"UNREACHABLE" and the port is listed as 0 instead of 5060 like all the 
internal phones.
I seem to recall some weird thing in the PIX, where you had to disable 
the SIP fixup to work (and of course, to use some nat traversal trick, 
like outbound proxy).
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Re: [Asterisk-Users] VoIP-to-TDM processing on-card?

[Steve Clark]

Steven Critchfield wrote:

Right now, I don't think there is any support for other codecs on TDM
cards. There is however discussions about using GPUs for codecs though.
When it comes down to it, reasonable hardware should handle decent
amounts of codec translations. If you are trying to stuff more than the
suggested amount of TE hardware in a box and do codec translation, then
you need to rethink the cost of failure. If you are dependant on 12 T1s
(value pulled from thin air, not necessarily related), You should see
about spitting it out over 3 boxes so at most you only lose 4 T1s at a
time. For most companies that rely on the phones, losing 1/3rd of the
production is pretty expensive, losing  all of production is not
tolerable. Any card that does the codec translation for you will
probably make you more likely to consolidate too many interfaces into
one machine.
You are correct there is no tdm processing done on the sangoma card, but it does 
handle all the communications protocol on the board unlike the digium cards.

Steve
--
"They that give up essential liberty to obtain temporary safety,
deserve neither liberty nor safety."  (Ben Franklin)
"The course of history shows that as a government grows, liberty
decreases."  (Thomas Jefferson)
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[Asterisk-Users] PIX!!!!!

[Christopher]

Can anyone point me in a good direction for configuring SIP through a 
PIX using 1:1 NAT.  I have read anything I could get my hands on and 
tried them all with very little success.  I can get it to work through 
the cheap little cable modem routers, but not this PIX. 

I -can- make a direct SIP call using the IP address of the * server 
([EMAIL PROTECTED]), but when I do that * still doesn't show it 
registering.  Even when I call through this method the phone comes up as 
"UNREACHABLE" and the port is listed as 0 instead of 5060 like all the 
internal phones.

--
B2 Technologies, LLC
   - www.VoIPSupply.com
   - www.ValueResale.com
454 Sonwil Drive
Buffalo, NY 14225
(716) 630-1555 x.27
(716) 250-3411 (Direct)
(716) 630-1548 fax
We sell all things VoIP related at: http://www.VoipSupply.com
Buy online, same day shipping on most items, volume discounts are available.
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Re: [Asterisk-Users] NMI issues...

[Nestor A. Diaz L.]

Hello everybody, i have found a message on the list regarding this
problem, i experienced this too, my machine show this on the kernel log:

Jan 15 11:21:08 catwoman kernel: Uhhuh. NMI received for unknown reason 31.
Jan 15 11:21:08 catwoman kernel: Dazed and confused, but trying to continue
Jan 15 11:21:08 catwoman kernel: Do you have a strange power saving mode 
enabled?
Jan 15 11:21:08 catwoman kernel: Uhhuh. NMI received for unknown reason 21.
Jan 15 11:21:08 catwoman kernel: Dazed and confused, but trying to continue
Jan 15 11:21:08 catwoman kernel: Do you have a strange power saving mode 
enabled?

and loops forever, i see some posts on the list and the solution was to
start linux with the parameter nmi_watchdog=1:

so put in lilo.conf:

append="nmi_watchdog=1"

If i use the nmi_watchdog=0 the behavior was the normal, (a lot of output
on the console), with nmi_watchdog=1 the machine freeze, and with
nmi_watchdog=1 the asterisk is working, however there are still a lot of
nmi interrupts, but no log and it works anyway:

catwoman:~# cat /proc/interrupts
   CPU0
  0:7938515IO-APIC-edge  timer
  1:   1439IO-APIC-edge  keyboard
  2:  0  XT-PIC  cascade
  4: 77IO-APIC-edge  serial
  8:  1IO-APIC-edge  rtc
 12: 19IO-APIC-edge  PS/2 Mouse
 14:  2IO-APIC-edge  libata
 15: 559174IO-APIC-edge  libata
 18:1978930   IO-APIC-level  eth0
 21:   79324605   IO-APIC-level  wcfxo
 22:   79325021   IO-APIC-level  wcfxo
 26: 238520   IO-APIC-level  eth1
NMI:   20571235
LOC:7938676
ERR:  0
MIS:  0

The machine is a Dell Poweredge 700 with a sata disk.

Slds.

--
Nestor A. Diaz
Ingeniero de Sistemas y Comp.
Tel. +57 1 6005490 x 211
Cel. +57 315 8190760
[EMAIL PROTECTED]
http://www.tiendalinux.com
Bogota, Colombia



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Re: [Asterisk-Users] Webmin Module for Asterisk {Scanned}

[Henry Devito]

 
www.thirdlane.com  has already written a close dsource webmin module.  I have no idea how much it costs or how well it works.
 
 
---Original Message---
 

From: David Shaw
Date: 01/20/05 10:35:53
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Webmin Module for Asterisk {Scanned}
 
I will try out your pages..
 
Thanks, David
 
PS I would love to work on your Asterisk Webmin pages, but I don't know
how.
 
 
On Thu, 2005-01-20 at 06:01, [EMAIL PROTECTED] wrote:
> There is already one, you can find it here :
> ftp://ftp.asterisk.org/pub/asterisk/webmin
>
> But I never managed to make it work, maybe it should be updated
>
> Anybody wanna take the challenge ? :)
>
> BTW, I've done some web pages that show you your configuration, and
> let you edit the text files in your browser. If you want it, drop me a
> message
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Re: [Asterisk-Users] SHORELINE IP100

[Joseph Finley]

Robert Augustyn wrote:

Joseph,
How did the insatllation go?
Any problems?
How do you power this units?
Thanks.
robert
--- Joseph Finley <[EMAIL PROTECTED]> wrote:

Robert Augustyn wrote:
Hi,
Is it the same as IP500?
Does it run the same software or do I need to
flash
it?
Is so whare do I get it?
Thanks a lot.
robert

It is a Polycom IP500 running MGCP image if you're
using ShoreTel.  We 
just finished a major ShoreTel installation at my
work place.


Installation went extremely well.  Never heard a better sounding system 
and I reviewed about a dozen of them.  The ShoreTel530 phones are 
phenomenal in sound.  Also the windows client is a nice touch.  Blows 
away my Cisco 7960 at home hooked up to *.  The phones are powered w/ 
bricks.  Didn't get approval to invest in Cisco PoE switches.

Joe


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RE: [Asterisk-Users] VoIP-to-TDM processing on-card?

[Steven Critchfield]

While it will probably be handled when you move out of outlook, please
wrap your lines at a reasonable length.

On Thu, 2005-01-20 at 15:30 -0500, Olson, Dana wrote:
> Actually, I do care, and I did search Google (albeit quickly) and I
> did look on the hardware list as well as the VoIP wiki. Maybe one of
> the cards listed there does what I need, but it wasn't listed like the
> QuickNet cards are. I thought perhaps the feature list on the site
> would be consistent, but apparently not. The fact that the hardware
> list on the Asterisk site does not include the Sangoma cards shows
> that it's not a complete list of supported hardware.

As of right now, I don't think the sangoma card supports any codec
conversions and asterisk support at the same time. I say this as it
supports the zaptel api and therefore should work similarly as the TE
cards from Digium as far as asterisk is concerned. 

Also I don't see that the Wanpipe hardware supports codec translations
either.

I may be wrong on that though.

> I don't understand what was so bad about my question. I thought it was
> direct and to the point, but apparently I left a lot of guesswork? A
> few people managed to actually answer my question - how did they do
> it?

It wasn't so much that your question was bad, but it didn't show that
you had the proper understanding of the question. If you had stated
which sites/URLs you had searched through to come to the conclusion that
there might be a reason the list would be more authoritative that those
URLs, it would have shown effort and therefore reason to be respected.
You will find that even in supposedly rough groups, effort is respected.
Few like freeloaders. Your question seemed like it was only about you
having others do your work. The extra couple of lines you would have
typed to show a bit of your previous effort would have sufficed to
eliminate that appearance.

> I'm sorry about the disclaimer, it is automatically added to any email
> that goes outside of our Exchange server. I'll get a new email account
> to use, as suggested by Steven. I won't email from this account again
> after this. I agree that it's annoying, but I fail to see how this
> makes me "lazy."

I'm not sure the disclaimer itself shows anything other than a stupid
policy by your employer. At some point your employer should actually
seek legal advice as to whether or not the disclaimer does any good at
all. I could disagree with it and therefore not be bound by it. As the
majority of the world isn't in any form of business agreements with your
employer, there isn't much you could do to compel others to abide by it.

> And to Timothy, who just wrote back to my original question, the
> reason I don't want the TE cards is that the processing is done on the
> system CPU and not on the card itself. I already have one of the TE
> cards though, and will make due if it comes to that.

Right now, I don't think there is any support for other codecs on TDM
cards. There is however discussions about using GPUs for codecs though.

When it comes down to it, reasonable hardware should handle decent
amounts of codec translations. If you are trying to stuff more than the
suggested amount of TE hardware in a box and do codec translation, then
you need to rethink the cost of failure. If you are dependant on 12 T1s
(value pulled from thin air, not necessarily related), You should see
about spitting it out over 3 boxes so at most you only lose 4 T1s at a
time. For most companies that rely on the phones, losing 1/3rd of the
production is pretty expensive, losing  all of production is not
tolerable. Any card that does the codec translation for you will
probably make you more likely to consolidate too many interfaces into
one machine.

-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] VoIP-to-TDM processing on-card?

[Andrew Kohlsmith]

On January 20, 2005 04:02 pm, Steven Critchfield wrote:
> While I thank those who come to my defense, I thought my first post was
> rather tame. I made the remarks about content and then proceeded to
> answer questions.

Your response was tame, it was Eric who got the prize this time.  :-)

-A.
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Re: [Asterisk-Users] top-notch servers/OS/network, ulaw codec - sound still choppy

[Paul Fielding]

- Original Message - 
I see the sip user is an external ip.  I would take a look at your QoS
settings on your router.  Make sure the voice traffic is getting the
priority it deserves.  Also, check for packet loss.
I'd still be wondering if there's something else.  I, too, experience choppy 
SIP connectivity from external IPs, but as I've mentioned in previous 
postings, I have a Vonage ATA that seems to have no problems keeping a 
crystal clear connection as it leaves my place and goes to Vonage's servers, 
so I think there must be more to it than QoS.   I have to believe that 
there's some more jitter correction or other such buffering that could 
berhaps be played with, though I don't know what it would be 
*shrug*.?

regards,
Paul


On Thu, 20 Jan 2005 11:26:02 -0800, Gabriel Afana <[EMAIL PROTECTED]> 
wrote:
Hi,
   My SIP calls are sounding a little choppy.  I've did my research but
everything looks right on my end...what am I missing?
Running RedHat ES 3.0 on dual AMD Opteron servers.  My system is 
cololocated
in downtown LA and is fed via a gigabit handoff from XO, AT&T, Level 3 
and
Wiltel (I have a 100Mb didicated line).  So I dont think its the Servers,
its the network, Asterisk is working fine and all codecs look 
right...what
could be the cause?

**SNIP FROM SIP.CONF***
[general]
context=default ; Default context for incoming calls
port=5060   ; UDP Port to bind to (SIP standard port 
is
5060)
;bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds 
to
all)
srvlookup=yes   ; Enable DNS SRV lookups on outbound 
calls
   ; Note: Asterisk only uses the first host
   ; in SRV records

allow=ulaw  ; Allow codecs in order of preference
*
ga0*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)   Format
64.201.99.2479092479878  2fd496bf330  00103/00105   ulaw
ga0*CLI> show version
Asterisk 1.0.3 built by [EMAIL PROTECTED] on a x86_64 running Linux
P.S.  in my sip.conf file, it looks like I am only allowing the ulaw
codec...could that cause a problem if I happen to need to call somebody 
that
doesn't support ulaw?

Gabe
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[Asterisk-Users] ASTCC config Problem

[Krystian Filiks]

Hello all,

 

I’m a little stuck in getting astcc
working.

 

I went to configure, set up the DB, Created Tables, but what
now?

 

What does the Users_Configure,
Brands, Cards,Trunks,
Routes, Sip Friends etc… do?

How do I configure Users (In the Users_configure
I get Not Configured)

How do I configure Cards and assign a card to a user?

 

Is there a tutorial on this on the web?

 

I have searched the web and could not find any helping
posts.

 

Thanks in advance.

KF






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Re: [Asterisk-Users] top-notch servers/OS/network, ulaw codec - sound still choppy

[Gabriel Afana]

Great, thanks for the info.  This is a service provided from my colo, so I 
will have to give them a call and find out whats up with their router 
settings.  As for packet loss, how do I check for that?

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

Sent: Thursday, January 20, 2005 12:32 PM
Subject: Re: [Asterisk-Users] top-notch servers/OS/network,ulaw codec - 
sound still choppy


I see the sip user is an external ip.  I would take a look at your QoS
settings on your router.  Make sure the voice traffic is getting the
priority it deserves.  Also, check for packet loss.
On Thu, 20 Jan 2005 11:26:02 -0800, Gabriel Afana <[EMAIL PROTECTED]> 
wrote:
Hi,
   My SIP calls are sounding a little choppy.  I've did my research but
everything looks right on my end...what am I missing?
Running RedHat ES 3.0 on dual AMD Opteron servers.  My system is 
cololocated
in downtown LA and is fed via a gigabit handoff from XO, AT&T, Level 3 
and
Wiltel (I have a 100Mb didicated line).  So I dont think its the Servers,
its the network, Asterisk is working fine and all codecs look 
right...what
could be the cause?

**SNIP FROM SIP.CONF***
[general]
context=default ; Default context for incoming calls
port=5060   ; UDP Port to bind to (SIP standard port 
is
5060)
;bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds 
to
all)
srvlookup=yes   ; Enable DNS SRV lookups on outbound 
calls
   ; Note: Asterisk only uses the first host
   ; in SRV records

allow=ulaw  ; Allow codecs in order of preference
*
ga0*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)   Format
64.201.99.2479092479878  2fd496bf330  00103/00105   ulaw
ga0*CLI> show version
Asterisk 1.0.3 built by [EMAIL PROTECTED] on a x86_64 running Linux
P.S.  in my sip.conf file, it looks like I am only allowing the ulaw
codec...could that cause a problem if I happen to need to call somebody 
that
doesn't support ulaw?

Gabe
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Re: [Asterisk-Users] VoIP-to-TDM processing on-card?

[Steven Critchfield]

On Thu, 2005-01-20 at 15:40 -0500, Brian Capouch wrote:
> Olson, Dana wrote:
> 
> > Anyhow, thanks to those who could see past the petty details, and sorry for 
> > pissing off those who can't.
> >
> 
> For what it's worth, back in the day I used to heat up red hot when I 
> saw Critch's tart responses to newbie questions.
> 
> Now that I'm doing the daily battle to read every mail on all the 
> Asterisk lists, I've changed my tune somewhat.
> 
> You should accept the fact that his approach got your attention, and 
> that on the main he was right: you admit you didn't look very hard, you 
> did in fact send a mail with a noxious dislaimer, and worst, you sent it 
> in HTML.
> 
> Take your lumps and move on, and recognize that Steve's contribution to 
> this list, while undeniably unpleasant when dished out, in the main is 
> performing a valuable service for all of those who strive to learn more 
> about Asterisk.

While I thank those who come to my defense, I thought my first post was
rather tame. I made the remarks about content and then proceeded to
answer questions.

If I am wrong and I wasn't tame, then maybe there is a disconnect
between myself and others about what should be offensive. It does seem
that Dana is taking the suggestions to heart and is changing for the
better.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] Re: Best Grandstream firmware to use?

[Aldo Bergamini]

[EMAIL PROTECTED] is believed to have said: 

>
>http://fm.grandstream.com/gs/
>
>
>Diego Aguirre
>FWD# 459696

Thanks!

Aldo

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Re: [Asterisk-Users] G.729? Worth it?

[Steve Kann]

Andrew Kohlsmith wrote:

  On January 19, 2005 12:23 pm, Paul Fielding wrote:
  
  
I think you might want to clarify that Best audio quality is in relation to
other highly compressed codecs.  Certainly my (albeit limited) experience
is that g711 is much more clear than g729.   Compared against gsm, for
example, however, the audio quality is quite good

  
  
It's strange...  I'm keen on iLBC because of the excellent PLC capabilties 
that will come around with the new IAX2 jitter buffer, but I just can't get 
it to sound good.
  

I can't help you with your ILBC stuff, but the jitterbuffer and PLC
patches I've been testing do PLC pretty well for GSM as well -- it uses
the same Generic PLC algorithm as G.711, (and ADPCM -- even implemented
it in lpc10..).

-SteveK




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[Asterisk-Users] AstTapi - Crashes w/ Windows 2000 - Urgent Help needed - May need to hire a developer

[Paul Rodan]

We’re encountering a problem with AstTAPI crashing on
Windows 2000 Workstations. The program we’re using is called Amicus
Attorney, it uses a standard TAPI interface to be able to dial our clients, but
on the 2 Windows 2000 workstations we’ve tried it on it has crashed, no
errors or anything. When we select the Asterisk TAPI driver, the whole windows
just closes/crashes w/ no apparent reason. 

 

Now I’ve tested AstTAPI on my laptop, but it’s
running Windows XP Pro w/ SP2 and I used Outlook. 

 

Is there a known issue with AstTAPI and Windows 2000? I know
Amicus Attorney works with other TAPI drivers/interfaces, because the old
system used this little external device hooked into the serial port (looked
like an external modem) that allowed Amicus Attorney to be able to place calls
out of our old phones/PBX. 

 

If there is a known issue, or a developer is willing to help
us fix the code, we’re willing to pay. But time is of the essence. Please
let me know! Thanks.

 






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[Asterisk-Users] Asterisk + Radius

[asterisk]

I want to configure Asterisk with FreeRaidus... Someone have done before?

Or please hint me where can i get documentation about it.

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Re: [Asterisk-Users] VoIP-to-TDM processing on-card?

[Brian Capouch]

Olson, Dana wrote:
Anyhow, thanks to those who could see past the petty details, and sorry for 
pissing off those who can't.
For what it's worth, back in the day I used to heat up red hot when I 
saw Critch's tart responses to newbie questions.

Now that I'm doing the daily battle to read every mail on all the 
Asterisk lists, I've changed my tune somewhat.

You should accept the fact that his approach got your attention, and 
that on the main he was right: you admit you didn't look very hard, you 
did in fact send a mail with a noxious dislaimer, and worst, you sent it 
in HTML.

Take your lumps and move on, and recognize that Steve's contribution to 
this list, while undeniably unpleasant when dished out, in the main is 
performing a valuable service for all of those who strive to learn more 
about Asterisk.

B.
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Re: [Asterisk-Users] RE: VoIP-to-TDM processing on-card?

[Patrick Conroy]

> Are there any cards that work with * that do the VoIP-to-TDM processing on
> the cards, with multiple T1 interfaces? 
>   
> The QuickNet Internet LineJack seems to meet the description I believe, but
> it only has a single FXS or FXO. Are there any cards that have multiple T1
> ports? 
>   

AudioCodes makes cards that will do what (it sounds like) you are
looking for.  I haven't used them, but from what I understand they
have up to 16 T1 ports and convert the calls to SIP or H323 on an
on-board Ethernet port.  They just use the PCI slot for power.  Since
they convert the calls to SIP, I would assume that asterisk wouldn't
need a special channel driver for the cards.  Hope this helps.

Patrick
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Re: [Asterisk-Users] VoIP-to-TDM processing on-card?

[Eric Wieling]

Olson, Dana wrote:
Actually, I do care, and I did search Google (albeit quickly) and I did look on 
the hardware list as well as the VoIP wiki. Maybe one of the cards listed there 
does what I need, but it wasn't listed like the QuickNet cards are. I thought 
perhaps the feature list on the site would be consistent, but apparently not. 
The fact that the hardware list on the Asterisk site does not include the 
Sangoma cards shows that it's not a complete list of supported hardware.
I don't understand what was so bad about my question. I thought it was direct 
and to the point, but apparently I left a lot of guesswork? A few people 
managed to actually answer my question - how did they do it?
I usually turn off HTML, and I apologize for that. I am writing from my work 
account, and Outlook seems to reset it, depending on what computer I sit at. I 
agree that it's annoying.
I'm sorry about the disclaimer, it is automatically added to any email that goes outside 
of our Exchange server. I'll get a new email account to use, as suggested by Steven. I 
won't email from this account again after this. I agree that it's annoying, but I fail to 
see how this makes me "lazy."
And to Timothy, who just wrote back to my original question, the reason I don't 
want the TE cards is that the processing is done on the system CPU and not on 
the card itself. I already have one of the TE cards though, and will make due 
if it comes to that.
Anyhow, thanks to those who could see past the petty details, and sorry for pissing off those who can't.
Sounds like what you really should be asking is "Does anyone know of 
cards that work with Asterisk that handle codec related stuff with an 
on-board DSP?"
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Re: [Asterisk-Users] top-notch servers/OS/network, ulaw codec - sound still choppy

[Jon Radon]

I see the sip user is an external ip.  I would take a look at your QoS
settings on your router.  Make sure the voice traffic is getting the
priority it deserves.  Also, check for packet loss.


On Thu, 20 Jan 2005 11:26:02 -0800, Gabriel Afana <[EMAIL PROTECTED]> wrote:
> Hi,
>My SIP calls are sounding a little choppy.  I've did my research but
> everything looks right on my end...what am I missing?
> 
> Running RedHat ES 3.0 on dual AMD Opteron servers.  My system is cololocated
> in downtown LA and is fed via a gigabit handoff from XO, AT&T, Level 3 and
> Wiltel (I have a 100Mb didicated line).  So I dont think its the Servers,
> its the network, Asterisk is working fine and all codecs look right...what
> could be the cause?
> 
> **SNIP FROM SIP.CONF***
> [general]
> context=default ; Default context for incoming calls
> port=5060   ; UDP Port to bind to (SIP standard port is
> 5060)
> ;bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
> all)
> srvlookup=yes   ; Enable DNS SRV lookups on outbound calls
>; Note: Asterisk only uses the first host
>; in SRV records
> 
> allow=ulaw  ; Allow codecs in order of preference
> *
> 
> ga0*CLI> sip show channels
> Peer User/ANRCall ID  Seq (Tx/Rx)   Format
> 64.201.99.2479092479878  2fd496bf330  00103/00105   ulaw
> 
> ga0*CLI> show version
> Asterisk 1.0.3 built by [EMAIL PROTECTED] on a x86_64 running Linux
> 
> P.S.  in my sip.conf file, it looks like I am only allowing the ulaw
> codec...could that cause a problem if I happen to need to call somebody that
> doesn't support ulaw?
> 
> Gabe
> 
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RE: [Asterisk-Users] VoIP-to-TDM processing on-card?

[Olson, Dana]

Actually, I do care, and I did search Google (albeit quickly) and I did look on 
the hardware list as well as the VoIP wiki. Maybe one of the cards listed there 
does what I need, but it wasn't listed like the QuickNet cards are. I thought 
perhaps the feature list on the site would be consistent, but apparently not. 
The fact that the hardware list on the Asterisk site does not include the 
Sangoma cards shows that it's not a complete list of supported hardware.

I don't understand what was so bad about my question. I thought it was direct 
and to the point, but apparently I left a lot of guesswork? A few people 
managed to actually answer my question - how did they do it?

I usually turn off HTML, and I apologize for that. I am writing from my work 
account, and Outlook seems to reset it, depending on what computer I sit at. I 
agree that it's annoying.

I'm sorry about the disclaimer, it is automatically added to any email that 
goes outside of our Exchange server. I'll get a new email account to use, as 
suggested by Steven. I won't email from this account again after this. I agree 
that it's annoying, but I fail to see how this makes me "lazy."

And to Timothy, who just wrote back to my original question, the reason I don't 
want the TE cards is that the processing is done on the system CPU and not on 
the card itself. I already have one of the TE cards though, and will make due 
if it comes to that.

Anyhow, thanks to those who could see past the petty details, and sorry for 
pissing off those who can't.

--
Dana



---Original Message---
I think of that person (no, not me) as akin to RMS.  Very smart, 
rather annoying at times, and usually RIGHT.

Steve fails to realize that most people don't care how readable their 
messages are, don't care to learn how to ask good questions, don't 
care about HTML posting, or anything else that used to make mailing 
lists and usenet work.  They post and expect magical fairies to read 
their mind and send back an answer.  Then they get upset when nobody 
wants to scroll thru 4 pages of footers and stuff that doesn't apply 
to their message.  These people are lazy and should NOT be rewarded.

Disclaimer: The information transmitted in this message is intended only for 
the person or entity to which it is addressed and may contain confidential 
and/or privileged material.  Any review, retransmission, dissemination, or 
other use of or taking of any action in reliance upon this information by 
persons or entities other than the intended recipient is prohibited.  If you 
received this message in error, please contact the sender and delete the 
material from any system.

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Re: [Asterisk-Users] Accessing Voice mail

[kurt x]

Brain,

I did what you suggested but instead of going to VoiceMailMain it
starts the begining of
my recorded message each time I press the "*" key.

[vmail]
exten => a,1,Voicemail(u${ext})
exten => a,2,Hangup

Kurt 



On Wed, 19 Jan 2005 11:48:18 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote:
> If you put the following in your Dialplan, pressing * should break you
> out of voicemail and call VoiceMailMain
> 
> exten => a,1,VoicemailMain,EXTEN
> exten => a,2,Hangup
> 
> 
> On Wed, 19 Jan 2005 11:33:23 -0500, kurt x <[EMAIL PROTECTED]> wrote:
> > I want to know if there is way to break out of the voicemail message.
> > for example:
> >
> > On my Noterl PBX when you dial you number from any where
> > you get your recorded voice mail message, but during the message I
> > press 81 and break out of that message.  It then
> > prompts me for my PIN thus allowing me to access my message
> > without using the auto attendant.
> >
> > Is this possible with Comedian?
> >
> > The below page did help.
> >
> > http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain
> >
> > Kurt
> > ___
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> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] VICIDIAL and meetme conference help

[Jake Franklin]

Hello,
I've installed VICIDIAL per the instructions on the astGUIclient 
website.  It appears everything is working correctly.  All the 
conference rooms have been set up, the database is running, and all the 
astGUIclient/VICIDIAL scripts are running.

I'm using the VICIDIAL client on windows 2000, and it also appears to be 
working correctly.  I can log in with no problems with the user I 
created, under a campaign I created.  Using my SIP softphone (SJphone), 
I can join the conference the VICIDIAL client tells me to (8600051 in 
this case).

However, after joining the conference specified by VICIDIAL, I keep 
getting the message:

Your phone does not appear to be in the VICIDIAL conference
Please Verify that you are in session 8600051
Of course, I'm already in conference 8600051.  So, somehow, VICIDIAL 
dosen't see me in the conference room.  I'm using the same username for 
my softphone in both sip.conf and the VICIDIAL configs.

Any thoughts or help would be greatly appreciated, and thanks in advance!
Jake Franklin
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Re: [Asterisk-Users] Realtime Engine

[Matthew Boehm]

> Then in my extensions table I will have data like the following
> INSERT INTO `extensions_table` VALUES (1, 'inbound', '_5172078354',
> 5,'DIAL', 'SIP/5172078354');
> INSERT INTO `extensions_table` VALUES (2, 'inbound', '_5172078355', 5,
> 'DIAL', 'SIP/5172078355');
> INSERT INTO `extensions_table` VALUES (3, 'inbound', '_5172078356', 5,
> 'DIAL', 'SIP/5172078360');
>

Why are you pattern matching the entire number? Take out those
underscores if you are matching an exact number.

-Matthew

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