Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[Dan]

Hi Steve,

>- Original Message - 
?From: "Steve Underwood" <[EMAIL PROTECTED]>
>...
>. However, I think this has nothing to do with the
> original poster's intent.

You're right.
Let's go back to my original problem.
There is any chance to make RxFax work with any type of fax machine?
It works for me just with a single (hardware) one.
The TxFax works with all of them.

..then... we can think further, to support fax data (not fax audio) over
slow links..:-)

Thank you and best regards,
Dan
P.S. I think that RxFax and TxFax applications, together with the fax suport
in * (tone detection and routing) has a huge potential... keep up the good
work.



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Re: [Asterisk-Users] MeetMe: Zap channels don't ever disconnect. . .

[Joel Maslak]

On Mon, 15 Dec 2003, Brian Capouch wrote:

> Anyone know of a way of doing this when the scumbag ILEC won't give you
>   supervision?

Probably not much.  Try turning on callprogress and/or busydetect - it
MIGHT help.  But the only way to do this right is with supervision on an
analog trunk or with digital circuits.

-- 
Joel
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[Asterisk-Users] MeetMe: Zap channels don't ever disconnect. . .

[Brian Capouch]

I was playing around with conferencing tonight.  I was able to place a 
bunch of SIP phones and a couple of my Zap FXS phones into a conference. 
 So I thought, "Let's see what it's like when people come in from outside."

So I called a friend and had him call in on one of my Zap channels, 
WHICH IS CONNECTED TO MY POTS LINE THAT DOESN'T DO DISCONNECT SUPERVISION.

When he hung up the phone stayed in the conference room until I manually 
destroyed the channel.

I had him call back and come into the conference again; this time we 
tried a "#" tone at the moment he left, but that didn't do any good either.

I waited perhaps a half-hour to see if *any* kind of timer would release 
the line, but no dice.

Anyone know of a way of doing this when the scumbag ILEC won't give you 
 supervision?

Thx.

B.

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[Asterisk-Users] Re: FAX, IAX and *....Maybe I'm dreaming...:-) (Carl Youngblood)

[ProvoCityPower]

Carl Wrote:
 
>Hi 
Jeff,>I live in Provo and I think I understand the application you're 
>referring to.  Some folks in my neighborhood have been getting to 
be the >beta testers for these cool new fiber links that the city is 
supposed to >be laying out.  If I only lived a few blocks over, I 
would be able to >get one too.  Darn.  Anyway, I've been 
following this thread, and I'm >wondering if an alternative might be to 
provide some sort of fax jack on >the hardware you provide the customer 
that your network could notice and >then treat differently from regular 
voice data?>An even better alternative would be if asterisk could 
recognize a fax >machine on the end of the line and use a different 
protocol or codec >that would work with faxes. It sounds like some of the 
contributors to >this thread were saying this is possible.  But I'm 
not sure--I'm pretty >new to asterisk and VoIP in general, so I could be 
wrong.>Carl
Good thoughts Carl. The device we are currently 
using has 2 analog (POTS) ports that carry traffic that is converted to IP 
as well as 8 10/100 ports and a 100MB fiber uplink. It may be possible to treat 
one of the analog ports differently and use the proper codec for that traffic. 
We have other alternatives for fax traffic, but not associated with *. I am 
really hoping * can play a role in our project. The citywide expansion of the 
project looks more imminent and promising than ever.
Again, I apologize for subverting the original 
intent of this thread.
 
Thanks,
Jeff

RE: [Asterisk-Users] Cisco 7960 lockups - any experiences?

[asterisk]

I noticed that on earlier versions of the firmware I was able to crash these
phones with a flood ping (ping -f ). I send this notice to Cisco
but never followed up on it. How are you powering this phone, POE or wall
cube?

-Bill

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nuno Cruz
Sent: Sunday, December 14, 2003 8:09 PM
To: John Todd
Subject: Re: [Asterisk-Users] Cisco 7960 lockups - any experiences?

Hello John,

I have a similar story with one of mine..

If i use a cell phone near it, it reboots :) The other ones are fine.. I
think it's a hardware problem, but it only reveals itself with the cell
phone radiation :>

Sunday, December 14, 2003, 10:59:14 PM, you wrote:


JT> This is almost certainly not an Asterisk-specific posting, but due
JT> to my inability to find a VoIP-focused Cisco list, I'll post here in
JT> the hopes of finding a more diverse user community.

JT> I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk,
JT> and have been experiencing situations where the phone locks up.
JT> "Locks up" means that the bottom part of the screen ("Your current
options"
JT> and the redial/newcall/cfwdall keys) disappears, and all keys on the
JT> keypad are non-functional except for the *-6-settings reboot keys.

JT> Previous software (4.4) exhibited the same symptoms.  I replaced the
JT> 7960 with a brand new 7960G, thinking it was bad hardware.  Same
JT> symptoms.  I replaced the power supply, thinking that perhaps could
JT> be the problem  Same symptoms.  I upgraded to 6.0.  Same symptoms.
JT> I replaced the ethernet cables.  Same symptoms.  I invoked an
JT> ancient curse-removal spell involving chicken bones and eye of newt
JT> over the phone.  Same symptoms.  A phone elsewhere in the same
JT> office (same switch, etc) works with no problems, but does not have
JT> a PC hooked to the 'PC' ethernet port.  In fact, the first 7960 I
JT> tried, with the swapped out power supply, works well elsewhere.

JT> I have used 79xx boxes in many circumstances, and all have performed
JT> admirably.  However, I rarely have configured them so that a PC is
JT> connected to the other side of the device.  The PC is running
JT> Windows XP, and sees mild SSH/email/web traffic.  I cannot say if
JT> there is a correlation between traffic volume and failure intervals;
JT> I have no data.

JT> The only thing that remains the same is the PC on one side, and the
JT> switch port on the other.  Previously, the PC attached to that
JT> switch port had worked without any problems for quite some time
JT> before I put the 7960 in place.  Even if the problem is the PC or
JT> the switchport, a failure or malfunction on one of those two
JT> components should not cause the phone to malfunction.  If it is a
JT> problem with the 7960 pushing traffic across the tiny built-in
JT> switch, then we have a real problem on our hands that Cisco had better
fix, pronto.

JT> I would appreciate reports of others as to similar issues or
JT> resolutions which they found effective.

JT> JT


--
Best regards,
 Nunomailto:[EMAIL PROTECTED]

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RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[John Breeden]

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of James Sharp
> Sent: Sunday, December 14, 2003 6:01 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
>
>
> > It's just my lowly opinion but I too must agree when it comes to the
> > consumer/soho (1 to 3 line) markets.
> >
> > CAUTION!!, DANGER!! Marketing Hat On!!
> >
> > Vonage, the most "visible" marketer of a voip consumer product must also
> > agree. Vontage offers an ip "fax line". using cisco's ata. Vontage must
> > see
> > some good reason for doing so. (I assume it's h.323). A&T, MCI and Time
> > Warner will be competing directly against Vonage when they
> introduce their
> > consumer voip products. I'd bet they too will be offering an ip
> fax line.
> >
> > Odds are you will be competing against them too. If I was vonage I'd be
> > telling the world how important a ip fax line was :-)
>
> Personally, I dont think that the world in general really cares about an
> "ip fax" line.  All they want is a system that works all the time/every
> time and doesn't require elaborate and convoluted setup.  They're not
> ooohing and ahhhing about "oooh, this uses VoIP".  They just know that
> they can stop spending $30/mo on an analog phone line and they get their
> long distance either flat rate or for an absurdly low per minute rate.
>
> Don't sell it as VoIP.  Sell it as a total replacement for the
> analog line.
>

broadband phone

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[Asterisk-Users] can X100P detect phone pick up like an answering machine

[Jim Flagg]

If Asterisk is configured as a simple answering machine replacement
with the X100P connected to PSTN line. No FXS ports in the 
Asterisk machine.  Standard phones are connect in parallel with
the X100P like you would a regular answering machine.

Can Asterisk detect that a phone has been picked up and cancel
the outgoing message and/or voice recording?  What about if the
phones are connected to the pass-through port of the X100P?

I know some PC software with voice modems can do this, just
wondering if X100P/Asterisk can do it?

Thanks





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RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[James Sharp]

> It's just my lowly opinion but I too must agree when it comes to the
> consumer/soho (1 to 3 line) markets.
>
> CAUTION!!, DANGER!! Marketing Hat On!!
>
> Vonage, the most "visible" marketer of a voip consumer product must also
> agree. Vontage offers an ip "fax line". using cisco's ata. Vontage must
> see
> some good reason for doing so. (I assume it's h.323). A&T, MCI and Time
> Warner will be competing directly against Vonage when they introduce their
> consumer voip products. I'd bet they too will be offering an ip fax line.
>
> Odds are you will be competing against them too. If I was vonage I'd be
> telling the world how important a ip fax line was :-)

Personally, I dont think that the world in general really cares about an
"ip fax" line.  All they want is a system that works all the time/every
time and doesn't require elaborate and convoluted setup.  They're not
ooohing and ahhhing about "oooh, this uses VoIP".  They just know that
they can stop spending $30/mo on an analog phone line and they get their
long distance either flat rate or for an absurdly low per minute rate.

Don't sell it as VoIP.  Sell it as a total replacement for the analog line.


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[Asterisk-Users] Error loading modem driver

[arohde]

When I attempt to start asterisk with my modem setup listed it will not start
attached are the error messages i get and also the modem.conf that i am currently 
using. Any assistance would be greatly appreciated.
running CVS ver 12/7/03, modified only to allow the RxFax and TxFax to compile and run 
with it (from http://www.opencall.org)

just e-mail me privately if you need more info

Thanks in advance
Rohde
  == Parsing '/etc/asterisk/asterisk.conf': Found
Asterisk CVS-12/07/03-20:38:37, Copyright (C) 1999-2001 Linux Support Services, Inc.
Written by Mark Spencer <[EMAIL PROTECTED]>
=
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
  == Manager registered action Ping
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Redirect
  == Manager registered action Originate
  == Manager registered action MailboxStatus
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxCount
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 1 -> 2
Asterisk PBX Core Initializing
Registering builtin applications:
 [AbsoluteTimeout]
  == Registered application 'AbsoluteTimeout'
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
  == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [DigitTimeout]
  == Registered application 'DigitTimeout'
 [Goto]
  == Registered application 'Goto'
 [GotoIf]
  == Registered application 'GotoIf'
 [GotoIfTime]
  == Registered application 'GotoIfTime'
 [Hangup]
  == Registered application 'Hangup'
 [NoOp]
  == Registered application 'NoOp'
 [Prefix]
  == Registered application 'Prefix'
 [ResetCDR]
  == Registered application 'ResetCDR'
 [ResponseTimeout]
  == Registered application 'ResponseTimeout'
 [Ringing]
  == Registered application 'Ringing'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayDigits]
  == Registered application 'SayDigits'
 [SetAccount]
  == Registered application 'SetAccount'
 [SetGlobalVar]
  == Registered application 'SetGlobalVar'
 [SetLanguage]
  == Registered application 'SetLanguage'
 [SetVar]
  == Registered application 'SetVar'
 [StripMSD]
  == Registered application 'StripMSD'
 [Suffix]
  == Registered application 'Suffix'
 [Wait]
  == Registered application 'Wait'
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [chan_modem.so] => (Generic Voice Modem Driver)
  == Parsing '/etc/asterisk/modem.conf': Found
  == Loading modem driver chan_modem_hcf.so => (hcfpcimodem-1.00lnxt03112100free)
WARNING[-1085185920]: File chan_modem.c, Line 390 (modem_setup): No driver for modem 
type 'hcfpcimodem-1.00lnxt03112100free'
WARNING[-1085185920]: File chan_modem.c, Line 735 (mkif): Unable to configure modem 
'/dev/modem'
ERROR[-1085185920]: File chan_modem.c, Line 871 (load_module): Unable to register 
channel '/dev/modem'
WARNING[-1085185920]: File loader.c, Line 312 (ast_load_resource): chan_modem.so: 
load_module failed, returning -1
WARNING[-1085185920]: File loader.c, Line 358 (load_modules): Loading module 
chan_modem.so failed!


modem.conf
Description: Binary data

RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[John Breeden]

It's 
just my lowly opinion but I too must agree when it comes to the consumer/soho (1 
to 3 line) markets. 
 
CAUTION!!, DANGER!! Marketing Hat 
On!!
 
Vonage, the most "visible" marketer of a voip consumer 
product must also agree. Vontage offers an ip "fax line". 
using cisco's ata. Vontage must see some good reason for doing so. (I 
assume it's h.323). A&T, MCI and Time Warner will be competing directly 
against Vonage when they introduce their consumer voip products. I'd bet 
they too will be offering an ip fax line.
 
Odds 
are you will be competing against them too. If I was vonage I'd be telling 
the world how important a ip fax line was :-)
 
Second, the residential/soho market almost demands 
replacement of analog with voip. It's almost impossible to justify the roi 
unless you do.
 
Marketing Hat Off
 
John 
Breeden
Hawaii
 

  I'm fairly 
  new here and don't mean to be contentious. We all have different perspectives 
  as to what VOIP should be. My goal is to replace analog lines, not supplement 
  them. I'm talking residential installations. I don't think I can ask these 
  folks to leave their fax on an anolog line? I think that if we start deciding 
  things for the Customer, then VOIP will be seen as an elitist toy for 
  digitally inclined, instead of an acceptable alternative for the 
  masses.
  No offense to 
  the anti-fax coalition.
  Jeff
  

Re: [Asterisk-Users] Cisco Gateway Integration

[Steven Thomas]

yes.  Cisco 2612 Router with 2
x FXO's and 2 x FXS's.  Works well using H323, and gnugk.


Steve.









"Bruce Hedreen" <[EMAIL PROTECTED]>
Sent by: [EMAIL PROTECTED]
15/12/2003 09:57 AM
Please respond to asterisk-users
        
        To:
       <[EMAIL PROTECTED]>
        cc:
       
        Subject:
       [Asterisk-Users] Cisco Gateway Integration

       

Has anyone succesfully integrated * with
a cisco voice gateway ?
 

Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[Carl Youngblood]

Hi Jeff,
I live in Provo and I think I understand the application you're 
referring to.  Some folks in my neighborhood have been getting to be the 
beta testers for these cool new fiber links that the city is supposed to 
be laying out.  If I only lived a few blocks over, I would be able to 
get one too.  Darn.  Anyway, I've been following this thread, and I'm 
wondering if an alternative might be to provide some sort of fax jack on 
the hardware you provide the customer that your network could notice and 
then treat differently from regular voice data?

An even better alternative would be if asterisk could recognize a fax 
machine on the end of the line and use a different protocol or codec 
that would work with faxes. It sounds like some of the contributors to 
this thread were saying this is possible.  But I'm not sure--I'm pretty 
new to asterisk and VoIP in general, so I could be wrong.

Carl

ProvoCityPower wrote:

>Did DVD players have to accommodate VHS tapes? Did VHS players have to
>accept beta?
>Why does VoIP have to deal with an accent protocol that can't handle
>lossy audio, nor irregular delays?
>Also why should we be soo wasteful when fax machines need a 80K codec to
>get the data across IP, and the faster machines I see say 15 secs per
>page. So why should we send 1.2meg when 150k is fine?
>Also who says Fax should ever be required on IP? My office has been
>using VoIP for all voice traffic for over a year now, but always left
>the fax machine on a analog line. The analog line was cheap enough to
>not be a concern.
>--
>Steven Critchfield <[EMAIL PROTECTED] >
 
I'm fairly new here and don't mean to be contentious. We all have 
different perspectives as to what VOIP should be. My goal is to 
replace analog lines, not supplement them. I'm talking residential 
installations. I don't think I can ask these folks to leave their fax 
on an anolog line? I think that if we start deciding things for the 
Customer, then VOIP will be seen as an elitist toy for digitally 
inclined, instead of an acceptable alternative for the masses.
No offense to the anti-fax coalition.
Jeff

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RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[ProvoCityPower]

>Did DVD 
players have to accommodate VHS tapes? Did VHS players have to>accept 
beta? >Why does VoIP have to deal with an accent protocol that can't 
handle>lossy audio, nor irregular delays?>Also why should we 
be soo wasteful when fax machines need a 80K codec to>get the data across 
IP, and the faster machines I see say 15 secs per>page. So why should we 
send 1.2meg when 150k is fine? >Also who says Fax should ever be 
required on IP? My office has been>using VoIP for all voice traffic for 
over a year now, but always left>the fax machine on a analog line. The 
analog line was cheap enough to>not be a concern. >-- 
>Steven Critchfield <[EMAIL PROTECTED]>
 
I'm fairly new 
here and don't mean to be contentious. We all have different perspectives as to 
what VOIP should be. My goal is to replace analog lines, not supplement them. 
I'm talking residential installations. I don't think I can ask these folks to 
leave their fax on an anolog line? I think that if we start deciding things for 
the Customer, then VOIP will be seen as an elitist toy for digitally inclined, 
instead of an acceptable alternative for the masses.
No offense to 
the anti-fax coalition.
Jeff

RE: [Asterisk-Users] Asterisk Crash System Command

[Kevin]

Thanks for the feedback on this. I just downloaded the latest CVS and
I'm not sure why it's doing this.  

-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED] 
Sent: Sunday, December 14, 2003 8:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk Crash System Command

Using latest cvs?  What distro? because it doesn't happen to mine.

bkw

On Sun, 14 Dec 2003, Kevin wrote:

> When I run the asterisk System command my asterisk crashes.  When I
> monitor the console this is the error I get.  Any suggestions?
>
> exten => 1,1,System(ls)
>
> this is the error I get:
>
> [EMAIL PROTECTED] asterisk]# Ouch ... error while writing audio data: :
> Broken pipe
> Ouch ... error while writing audio data: : Broken pipe
>
>
>
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Re: [Asterisk-Users] Asterisk Crash System Command

[Brian West]

Using latest cvs?  What distro? because it doesn't happen to mine.

bkw

On Sun, 14 Dec 2003, Kevin wrote:

> When I run the asterisk System command my asterisk crashes.  When I
> monitor the console this is the error I get.  Any suggestions?
>
> exten => 1,1,System(ls)
>
> this is the error I get:
>
> [EMAIL PROTECTED] asterisk]# Ouch ... error while writing audio data: :
> Broken pipe
> Ouch ... error while writing audio data: : Broken pipe
>
>
>
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>
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[Asterisk-Users] Asterisk Crash System Command

[Kevin]

When I run the asterisk System command my asterisk crashes.  When I
monitor the console this is the error I get.  Any suggestions?

exten => 1,1,System(ls)

this is the error I get:

[EMAIL PROTECTED] asterisk]# Ouch ... error while writing audio data: :
Broken pipe
Ouch ... error while writing audio data: : Broken pipe



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Re: [Asterisk-Users] Cisco 7960 lockups - any experiences?

[Nuno Cruz]

Hello John,

I have a similar story with one of mine..

If i use a cell phone near it, it reboots :)
The other ones are fine.. I think it's a hardware problem, but it only
reveals itself with the cell phone radiation :>

Sunday, December 14, 2003, 10:59:14 PM, you wrote:


JT> This is almost certainly not an Asterisk-specific posting, but due to
JT> my inability to find a VoIP-focused Cisco list, I'll post here in the
JT> hopes of finding a more diverse user community.

JT> I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and
JT> have been experiencing situations where the phone locks up.  "Locks
JT> up" means that the bottom part of the screen ("Your current options"
JT> and the redial/newcall/cfwdall keys) disappears, and all keys on the
JT> keypad are non-functional except for the *-6-settings reboot keys.

JT> Previous software (4.4) exhibited the same symptoms.  I replaced the
JT> 7960 with a brand new 7960G, thinking it was bad hardware.  Same 
JT> symptoms.  I replaced the power supply, thinking that perhaps could
JT> be the problem  Same symptoms.  I upgraded to 6.0.  Same symptoms.  I
JT> replaced the ethernet cables.  Same symptoms.  I invoked an ancient
JT> curse-removal spell involving chicken bones and eye of newt over the
JT> phone.  Same symptoms.  A phone elsewhere in the same office (same
JT> switch, etc) works with no problems, but does not have a PC hooked to
JT> the 'PC' ethernet port.  In fact, the first 7960 I tried, with the
JT> swapped out power supply, works well elsewhere.

JT> I have used 79xx boxes in many circumstances, and all have performed
JT> admirably.  However, I rarely have configured them so that a PC is
JT> connected to the other side of the device.  The PC is running Windows
JT> XP, and sees mild SSH/email/web traffic.  I cannot say if there is a
JT> correlation between traffic volume and failure intervals; I have no
JT> data.

JT> The only thing that remains the same is the PC on one side, and the
JT> switch port on the other.  Previously, the PC attached to that switch
JT> port had worked without any problems for quite some time before I put
JT> the 7960 in place.  Even if the problem is the PC or the switchport,
JT> a failure or malfunction on one of those two components should not
JT> cause the phone to malfunction.  If it is a problem with the 7960 
JT> pushing traffic across the tiny built-in switch, then we have a real
JT> problem on our hands that Cisco had better fix, pronto.

JT> I would appreciate reports of others as to similar issues or 
JT> resolutions which they found effective.

JT> JT


-- 
Best regards,
 Nunomailto:[EMAIL PROTECTED]

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RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[John Breeden]

How is Vonage doing it?

http://www.vonage.com/features_fax.php

John Breeden
Hawaii

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Steve
> Underwood
> Sent: Sunday, December 14, 2003 2:24 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
> 
> 
> ProvoCityPower wrote:
> 
> > The question asked here, "why on earth you want to push fax data over 
> > a VoIP link at
> > all. Fax compression isn't very efficient." may speak volumes about 
> > the future role of VOIP. My plans are to role out a VOIP connection to 
> > thousands of Customers. Many have legacy fax equipment. Am I to assume 
> > that they will toss out their fax equipment and join the PC based 
> > faxing crowd? I don't think I can control this. If I am going to offer 
> > an aternative to the legacy wire providers then I have to offer a 
> > comparable service. One that for example allows a customer to use a 
> > legacy fax machine in the same way.
> >  
> > If this thought sidetracks the intent of this thread, you have my 
> > apologies, but I do think that legacy fax functionality is essential.
> 
> Legacy FAX will be very important for years. The last people to abandon 
> it are the most senior managers. Therefore, apart from anything else, 
> their buyin to a change to VoIP depends on keeping their olde worlde FAX 
> facilities alive. However, I think this has nothing to do with the 
> original poster's intent.
> 
> Sending FAXes over IP as audio is dumb. Its troublesome, error prone, 
> and consumes too much bandwidth. The right way is to use a FAX modem to 
> translate between audio and the digital data stream of the image itself. 
> Then the compact image data can be conveyed reliably. If the far end is 
> not using IP, another modem can be used to return the FAX to its 
> analogue form for delivery. If you wish to interwork traditional 
> analogue FAX machines, with newer IP capable FAX machines you *have* to 
> do this. This is what the H323 protocols do. T.38 is the relevant FAX 
> over IP protocol. I don't know if there is a similar SIP standard, but 
> there should be. FAX to e-mail and e-mail to FAX is the other inportant 
> way to blend the old with the new.
> 
> Regards,
> Steve
> 
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RE: [Asterisk-Users] Cisco 7960 lockups - any experiences?

[Paul Mahler]

So do I have this right?  You have a 7960 hooked up to your network. You
have a PC plugged into the second Ethernet port of the phone. This 7960
freezes. 

If you move the phone somewhere else in the office, without the PC, it
doesn't freeze?  Does it freeze when it's in the same location, but the PC
isn't plugged into it? That is, does this happen only when a PC is plugged
into the phone?

Paul

 
Paul Mahler 
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Sunday, December 14, 2003 2:59 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 7960 lockups - any experiences?


This is almost certainly not an Asterisk-specific posting, but due to 
my inability to find a VoIP-focused Cisco list, I'll post here in the 
hopes of finding a more diverse user community.

I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and 
have been experiencing situations where the phone locks up.  "Locks 
up" means that the bottom part of the screen ("Your current options" 
and the redial/newcall/cfwdall keys) disappears, and all keys on the 
keypad are non-functional except for the *-6-settings reboot keys.

Previous software (4.4) exhibited the same symptoms.  I replaced the 
7960 with a brand new 7960G, thinking it was bad hardware.  Same 
symptoms.  I replaced the power supply, thinking that perhaps could 
be the problem  Same symptoms.  I upgraded to 6.0.  Same symptoms.  I 
replaced the ethernet cables.  Same symptoms.  I invoked an ancient 
curse-removal spell involving chicken bones and eye of newt over the 
phone.  Same symptoms.  A phone elsewhere in the same office (same 
switch, etc) works with no problems, but does not have a PC hooked to 
the 'PC' ethernet port.  In fact, the first 7960 I tried, with the 
swapped out power supply, works well elsewhere.

I have used 79xx boxes in many circumstances, and all have performed 
admirably.  However, I rarely have configured them so that a PC is 
connected to the other side of the device.  The PC is running Windows 
XP, and sees mild SSH/email/web traffic.  I cannot say if there is a 
correlation between traffic volume and failure intervals; I have no 
data.

The only thing that remains the same is the PC on one side, and the 
switch port on the other.  Previously, the PC attached to that switch 
port had worked without any problems for quite some time before I put 
the 7960 in place.  Even if the problem is the PC or the switchport, 
a failure or malfunction on one of those two components should not 
cause the phone to malfunction.  If it is a problem with the 7960 
pushing traffic across the tiny built-in switch, then we have a real 
problem on our hands that Cisco had better fix, pronto.

I would appreciate reports of others as to similar issues or 
resolutions which they found effective.

JT

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Re: [Asterisk-Users] unable to configure my Grandstream phone

[Balaji NJL]

i hv also added the alaw

> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = bogon-calls
> ;context = default
> disallow=all
> allow=g729
> allow=gsm
> allow=ulaw
allow=alaw

now i am able to call from my MSN -> * ->GS but the
other way is not
working. i am getting lot of noise when i try to place
any call from my GS.
i am no longer getting codec not compatible error
message anymore. i am
still unable to place any calls using my GS (to my
internal MSN extensions
or to external PSTN).

thanks for ur help,
-B
>

- Original Message - 
From: "Balaji NJL" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: <[EMAIL PROTECTED]>
Sent: Sunday, December 14, 2003 3:40 PM
Subject: Re: [Asterisk-Users] unable to configure my
Grandstream phone


> Hi Paul,
>
> thanks for the quick response. i tried the following
> configuration /
> combination still no luck
>
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = bogon-calls
> ;context = default
> disallow=all
> allow=g729
> allow=gsm
> allow=ulaw
>
> when i tried g711 i am getting an error in * that
> codec not found.
>
> When i specify only g729 my MSN doesnt work.
>
> thanks,
> -B
>
> - Original Message - 
> From: "Paul Liew" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, December 14, 2003 2:14 PM
> Subject: Re: [Asterisk-Users] unable to configure my
> Grandstream phone
>
>
> >
> > - Original Message - 
> > From: Balaji NJL
> > To: [EMAIL PROTECTED]
> > Sent: Monday, December 15, 2003 8:47 AM
> > Subject: [Asterisk-Users] unable to configure my
> Grandstream phone
> >
> >
> > 
> > > Attempting native bridge of SIP/2003-b895 and
> SIP/2000-53e2
> > > WARNING[5126]: File chan_sip.c, Line 1954
> (process_sdp): No compatible
> > codecs!
> > > -- Got SIP response 481 "Call
Leg/Transaction
> Does Not Exist" back
> > from 192.168.0.58
> > >
> > > and then the call drops. When i am making a call
> using Grandstream ph,
> it
> > rings the other side when they pick up the phone
the
> call then drops. then
> i
> > get the
> > > above error message.
> > >
> > > the follwoing us sip and Grandstream conf
> > >
> > >
> > > [general]
> > > port = 5060
> > > bindaddr = 0.0.0.0
> > > context = bogon-calls
> > > ;context = default
> > > disallow=all
> > > allow=gsm
> >
> > Balaji,
> >
> > Grandstreams do not support GSM. Options available
> can be seen on the GS
> > config page. Unless you purchase G729s (for low
> bandwidth), your only
> choice
> > is G711.
> >
> > Paul
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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>
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Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[Steve Underwood]

ProvoCityPower wrote:

The question asked here, "why on earth you want to push fax data over 
a VoIP link at
all. Fax compression isn't very efficient." may speak volumes about 
the future role of VOIP. My plans are to role out a VOIP connection to 
thousands of Customers. Many have legacy fax equipment. Am I to assume 
that they will toss out their fax equipment and join the PC based 
faxing crowd? I don't think I can control this. If I am going to offer 
an aternative to the legacy wire providers then I have to offer a 
comparable service. One that for example allows a customer to use a 
legacy fax machine in the same way.
 
If this thought sidetracks the intent of this thread, you have my 
apologies, but I do think that legacy fax functionality is essential.
Legacy FAX will be very important for years. The last people to abandon 
it are the most senior managers. Therefore, apart from anything else, 
their buyin to a change to VoIP depends on keeping their olde worlde FAX 
facilities alive. However, I think this has nothing to do with the 
original poster's intent.

Sending FAXes over IP as audio is dumb. Its troublesome, error prone, 
and consumes too much bandwidth. The right way is to use a FAX modem to 
translate between audio and the digital data stream of the image itself. 
Then the compact image data can be conveyed reliably. If the far end is 
not using IP, another modem can be used to return the FAX to its 
analogue form for delivery. If you wish to interwork traditional 
analogue FAX machines, with newer IP capable FAX machines you *have* to 
do this. This is what the H323 protocols do. T.38 is the relevant FAX 
over IP protocol. I don't know if there is a similar SIP standard, but 
there should be. FAX to e-mail and e-mail to FAX is the other inportant 
way to blend the old with the new.

Regards,
Steve
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Re: [Asterisk-Users] ignorepat

[Steve Rodgers]

Sip phones generate their own dialtone. The ignore pat option is meaningless 
with regard to SIP phones. I would check the Qrandstream's dialplan and see if 
you can program it to ignore the dialtone after a '9' is pressed. I had to do 
something similar for my Sipura SPA-2000.

Steve.



On Sunday 14 December 2003 12:18, Burak Balasaygun wrote:
> Hi
>
>   I have the following configuration at home one ZAPTEL interface
> connecting to an FXO card and two SIP UAs connecting to asterisk locally. I
> have configured extensions.conf such that dialing 9 on the SIP phones
> allows me to dial an outbound number via the FXO interface . Works fine.
>
>
>   What's not working is that pressing 9 should causes either  GS BT-100
> phone to reacquire a  dialtone since I have placed ignorepat => 9 in the
> config file.
>
>   Any ideas?
>
>
>
>
> rgds
>
> burak
>
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Re: [Asterisk-Users] unable to configure my Grandstream phone

[Balaji NJL]

Hi Paul,

thanks for the quick response. i tried the following
configuration /
combination still no luck

[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
;context = default
disallow=all
allow=g729
allow=gsm
allow=ulaw

when i tried g711 i am getting an error in * that
codec not found.

When i specify only g729 my MSN doesnt work.

thanks,
-B

- Original Message - 
From: "Paul Liew" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, December 14, 2003 2:14 PM
Subject: Re: [Asterisk-Users] unable to configure my
Grandstream phone


>
> - Original Message - 
> From: Balaji NJL
> To: [EMAIL PROTECTED]
> Sent: Monday, December 15, 2003 8:47 AM
> Subject: [Asterisk-Users] unable to configure my
Grandstream phone
>
>
> 
> > Attempting native bridge of SIP/2003-b895 and
SIP/2000-53e2
> > WARNING[5126]: File chan_sip.c, Line 1954
(process_sdp): No compatible
> codecs!
> > -- Got SIP response 481 "Call Leg/Transaction
Does Not Exist" back
> from 192.168.0.58
> >
> > and then the call drops. When i am making a call
using Grandstream ph,
it
> rings the other side when they pick up the phone the
call then drops. then
i
> get the
> > above error message.
> >
> > the follwoing us sip and Grandstream conf
> >
> >
> > [general]
> > port = 5060
> > bindaddr = 0.0.0.0
> > context = bogon-calls
> > ;context = default
> > disallow=all
> > allow=gsm
>
> Balaji,
>
> Grandstreams do not support GSM. Options available
can be seen on the GS
> config page. Unless you purchase G729s (for low
bandwidth), your only
choice
> is G711.
>
> Paul
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
>
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[Asterisk-Users] Cisco 7960 lockups - any experiences?

[John Todd]

This is almost certainly not an Asterisk-specific posting, but due to 
my inability to find a VoIP-focused Cisco list, I'll post here in the 
hopes of finding a more diverse user community.

I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and 
have been experiencing situations where the phone locks up.  "Locks 
up" means that the bottom part of the screen ("Your current options" 
and the redial/newcall/cfwdall keys) disappears, and all keys on the 
keypad are non-functional except for the *-6-settings reboot keys.

Previous software (4.4) exhibited the same symptoms.  I replaced the 
7960 with a brand new 7960G, thinking it was bad hardware.  Same 
symptoms.  I replaced the power supply, thinking that perhaps could 
be the problem  Same symptoms.  I upgraded to 6.0.  Same symptoms.  I 
replaced the ethernet cables.  Same symptoms.  I invoked an ancient 
curse-removal spell involving chicken bones and eye of newt over the 
phone.  Same symptoms.  A phone elsewhere in the same office (same 
switch, etc) works with no problems, but does not have a PC hooked to 
the 'PC' ethernet port.  In fact, the first 7960 I tried, with the 
swapped out power supply, works well elsewhere.

I have used 79xx boxes in many circumstances, and all have performed 
admirably.  However, I rarely have configured them so that a PC is 
connected to the other side of the device.  The PC is running Windows 
XP, and sees mild SSH/email/web traffic.  I cannot say if there is a 
correlation between traffic volume and failure intervals; I have no 
data.

The only thing that remains the same is the PC on one side, and the 
switch port on the other.  Previously, the PC attached to that switch 
port had worked without any problems for quite some time before I put 
the 7960 in place.  Even if the problem is the PC or the switchport, 
a failure or malfunction on one of those two components should not 
cause the phone to malfunction.  If it is a problem with the 7960 
pushing traffic across the tiny built-in switch, then we have a real 
problem on our hands that Cisco had better fix, pronto.

I would appreciate reports of others as to similar issues or 
resolutions which they found effective.

JT

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[Asterisk-Users] Cisco Gateway Integration

[Bruce Hedreen]

Title: Message



Has anyone 
succesfully integrated * with a cisco voice gateway ?
 

[Asterisk-Users] Asterisk problem

[T. Chan]

 
Dear All,
I am a new user of Asterisk 
interested in setting up a VOIP network based on Asterisk. I have deployed a few 
Asterisk servers running on T400P and have started a few weeks ago to run some 
LIVE traffic on one of the servers. Most of my current traffic is via H323 to 
and from other carriers / customers and I am using the H323 driver written by 
Jeremy of Nufone.
I have had problems with 
this, as it seems that as time goes, for instance after 5 or 6 hours or 
sometimes 3 to 4 hours or less, the system would not be able to take any 
incoming calls via H323 protocol. All the calls from my H323 customers (they are 
mostly carriers sending traffic to me from other gateways such as cisco AS5300) 
would be rejected and the system just would not take the calls for some reasons. 
I could, however, call into the PRI (Zap interface) and make an outbound H323 
calls to other gateways of my carriers. That is, incoming H323 calls would not 
work after a certain period of operation time but outgoing H323 calls would 
still be possible.
I have then downloaded newer 
version of the Asterisk about 10 days ago, but then my Asterisk would start to 
crash on me, the module would just stop running by itself and I had to restart 
the Asterisk. Sometimes, it would just stop running, but 75% of the time, I 
would see this error message:
Connected to Asterisk CVS-12/06/03-03:06:28 
currently running on localhost (pid = 23720)    -- Remote 
UNIX connectionlocalhost*CLI> /usr/sbin/safe_asterisk: line 6: 23720 
Segmentation fault  asterisk ${ASTARGS} 
1>&/dev/${TTY} Asterisk ended with exit status 
139Asterisk exited on signal 11.Automatically restarting 
Asterisk.
 
Disconnected from Asterisk 
serverExecuting last minute cleanups[EMAIL PROTECTED] asterisk]# 
/usr/sbin/safe_asterisk: line 6: 23752 Segmentation 
fault  asterisk ${ASTARGS} 1>&/dev/${TTY} 
Asterisk ended with exit status 139Asterisk exited on 
signal 11.Automatically restarting Asterisk.
 
It seems that it would 
start to restart by would fail, and then I had to restart it manually. This 
Segmentation error happens only with new version of Asterisk but not with older 
version (10 days ago), does anyone here have any of such experience or know why 
this is happening? Is this because I am running T400P instead of TE410P hardware 
and that the newer version of software is not 100% compatible with the old 
hardware? The inbound H323 problem, however, exists for both the newer and older 
version of Asterisk, and I wonder if anyone has had this experience before. Is 
this pertaining to the hardware as well or is there any other 
reason/
 
Ladies and Gentlemen, any 
feedbacks and advices would be appreciated. Many 
Thanks
Tommy Chan 

[Asterisk-Users] nexthop: free service to pstn via fwd/iaxtel

[Dorian Gray]

(found on dslr voip forum)
http://www.freephoneproject.com/nexthop/
I think this is pretty cool. not too sure how stable/durable though, at 
least not until there are a lot more nodes providing service. also 
unsure of the legality vs. your average telco's consumer tos, but it's 
definitely not reselling.

jeff pulver (crom bless him) seemed to be fairly amused, stuff like this 
fits right into his vision as far as I can tell.

looking forward to the day when enough people have hardphones that 
non-free pstn access becomes fairly redundant.

cheers
++dg
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Re: [Asterisk-Users] unable to configure my Grandstream phone

[Paul Liew]

- Original Message - 
From: Balaji NJL
To: [EMAIL PROTECTED]
Sent: Monday, December 15, 2003 8:47 AM
Subject: [Asterisk-Users] unable to configure my Grandstream phone



> Attempting native bridge of SIP/2003-b895 and SIP/2000-53e2
> WARNING[5126]: File chan_sip.c, Line 1954 (process_sdp): No compatible
codecs!
> -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back
from 192.168.0.58
>
> and then the call drops. When i am making a call using Grandstream ph, it
rings the other side when they pick up the phone the call then drops. then i
get the
> above error message.
>
> the follwoing us sip and Grandstream conf
>
>
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = bogon-calls
> ;context = default
> disallow=all
> allow=gsm

Balaji,

Grandstreams do not support GSM. Options available can be seen on the GS
config page. Unless you purchase G729s (for low bandwidth), your only choice
is G711.

Paul

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[Asterisk-Users] unable to configure my Grandstream phone

[Balaji NJL]

Hi All,
 
i received my X100P and Grandstream phone last 
week. i started configuring my * and with the help of ur mailing lists i was 
able to configure it. (when ever i got struck i searched this list and found my 
answer. thanks a lot and this list is awesome). i still hv a small problem and 
hope someone could help me out.
 
This is my setup. 
 
RH 7.2 serving as my * server. i hv got couple of 
my laptops and desktops running MSN 4.7. I hv installed and configured X100P and 
Grandstream phone
 
the following configurations are 
working
 
MSN Msgr -> * -> MSN Msgr
MSN Msgr -> * -> X100P - PSTN
PSTN -> * -> MSN Msgr
PSTN -> * -> Grandstream (pl 
note)
 
the following are *not* working
 
MSN Msgr -> * -> Grandstream
Grandstream -> * -> MSN Msgr
GrandStream -> * -> PSTN
PSTN -> * -> Grandstream
 
the error i am getting in this case is 

 
- Attempting native bridge of SIP/2003-b895 and 
SIP/2000-53e2WARNING[5126]: File chan_sip.c, Line 1954 (process_sdp): No 
compatible codecs!    -- Got SIP response 481 "Call 
Leg/Transaction Does Not Exist" back from 192.168.0.58
 
and then the call drops. When i am making a call 
using Grandstream ph, it rings the other side when they pick up the phone the 
call then drops. then i get the above error message.
 
the follwoing us sip and Grandstream 
conf
 
[general]port = 5060bindaddr = 0.0.0.0context = 
bogon-calls;context = defaultdisallow=allallow=gsm
 
[2000]; Grandstream phone
 
type=friendusername=2000secret=qweqwehost=dynamiccontext=from-sipmailbox=2000dtmfmode=inband
 
[2002]
 
type=friendhost=dynamicinsecure=yesdtmfmode=inband;dtmfmode=rfc2833context=from-sipmailbox=2002;auth=plaintext
 
[2003]
 
type=friendhost=dynamicinsecure=yesdtmfmode=inbandcontext=from-sipmailbox=2003
 
;[2000]
 
;type=friend;username=2000;secret=qweqwe;auth=md5;host=dynamic;context=from-sip;dtmfmode=inband;mailbox=2000
 
[2001]
 
type=friendusername=2001secret=asdasdauth=md5host=dynamiccontext=from-sipdtmfmode=inbandmailbox=2001
 
Grandstream configuration details
>SIP Server:  192.168.0.4  (my * box)>SIP Userid:  
2000 (userid is same as extension>Authenticate ID: 
2000>Authenticate password:  qweqwe
>Send DTMF:  Via SIP info   (in order for the dtmf to be 
recognized by>voicemail)>>  


  
  
Program--1.0.4.17    Bootloader--1.0.0.11 
     HTML--1.0.0.19 
  
 

 
any idea why my Grandstream drops the calls.
 
thanks a lot and appreciate ur help.
 
-B

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RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[Steven Critchfield]

On Sun, 2003-12-14 at 12:18, [EMAIL PROTECTED] wrote:
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of ProvoCityPower
> > The question asked here, "why on earth you want to push fax data
> > over a VoIP link at all. Fax compression isn't very efficient." may
> > speak volumes about the future role of VOIP. My plans are to role
> > out a VOIP connection to thousands of Customers. Many have legacy
> > fax equipment. Am I to assume that they will toss out their fax
> > equipment and join the PC based faxing crowd? I don't think I can
> > control this. If I am going to offer an aternative to the legacy
> > wire providers then I have to offer a comparable service. One that
> > for example allows a customer to use a legacy fax machine in the
> > same way.

> I think what you're talking about here is an absolute necessity if VOIP is
> ever to compete for traditional analog service. A lot of users will not want
> to change their ways. As stupid as I think FAXing and FAX machines are there
> are still millions of people who prefer to send and recive a skewed low
> resolution fax on thermal paper than the transmit and receive them over a
> computer. Even though, I think FAXing is as ancient as sending a letter
> through the mail it needs to be supported however the end user wishes to use
> it.
> 
> With that being said, I think it simply comes down to the codecs being used.
> I seem to recall that alaw and ulaw were acceptable codecs, though I know
> nothing about this first hand. The other option is some sort of FAX proxy,
> though that seems a little too complicated if you ask me.

Did DVD players have to accommodate VHS tapes? Did VHS players have to
accept beta? 

Why does VoIP have to deal with an accent protocol that can't handle
lossy audio, nor irregular delays?

Also why should we be soo wasteful when fax machines need a 80K codec to
get the data across IP, and the faster machines I see say 15 secs per
page. So why should we send 1.2meg when 150k is fine? 

Also who says Fax should ever be required on IP? My office has been
using VoIP for all voice traffic for over a year now, but always left
the fax machine on a analog line. The analog line was cheap enough to
not be a concern. 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] GS early dial

[Miguel Cavazos]

it doesnt work here, same firmware 4.26 tryed it with 4.18 also and it
doesnt work, i press any number and it gets screw up 

i will try it with the handytone ata286 and see if it works, anyway its
the same firmware but its worth to try out

Miguel
On Sun, 2003-12-14 at 21:13, John Breeden wrote:
> Am I assuming that a GS set to early dial to * dosn't work. Or am I
> missing something? Tried inband, info and rfc288, all nojoy. I'm
> assuming that it's not/supported or GS bug, only asking because it's
> assumptions that alwas get me :-)
>  
> GS firmware 1.0.4.26
>  
> Thanx in advance
>  
> John Breeden
> Hawaii
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[Asterisk-Users] GS early dial

[John Breeden]

Am I 
assuming that a GS set to early dial to * dosn't work. Or am I missing 
something? Tried inband, info and rfc288, all nojoy. I'm assuming that it's 
not/supported or GS bug, only asking because it's assumptions that alwas get me 
:-)
 
GS 
firmware 1.0.4.26
 
Thanx 
in advance
 
John 
Breeden
Hawaii

RE: [Asterisk-Users] Asterisk and fwd

[Tom Shoval]

Please do.

 









From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Chris HARIGA
Sent: Sunday, December 14, 2003
8:41 PM
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
Asterisk and fwd



 



If
U want I can send U my settings. My FWD is working fine. Let me know...





 





Chris
HARIGA





 







-
Original Message - 





From: Shoval Tomer 





To: [EMAIL PROTECTED] 





Sent: Sunday, December 14, 2003 10:31 AM





Subject: [Asterisk-Users] Asterisk and fwd





 



Hi,
could anyone please provide a working sample of how to configure asterisk to
connect to fwd?

I've
tried the one at www.loligo.com and it
doesn't work. Not even when calling to 5.

 

Can
you advise on how to debug sip (or trace and view sip packets) from the
asterisk server to fwd so we can try to understand what exactly is not working?

 

Please
be advised that our asterisk server is behind a NAT firewall.

 

Thanks
in advance,

Tom.

 

[Asterisk-Users] Asterisk Crash System Command

[Kevin]

When I run the asterisk System command my asterisk crashes.  When I
monitor the console this is the error I get.  Any suggestions?

exten => 1,1,System(ls)

this is the error I get:

[EMAIL PROTECTED] asterisk]# Ouch ... error while writing audio data: :
Broken pipe
Ouch ... error while writing audio data: : Broken pipe



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Re: [Asterisk-Users] voice mail - sip:notify message

[SW]

Thanks, John.

My requirement here is little different. I am using * as a Voice Mail server
for Vocal ((in addition It does codec conversion and routing to PSTN/SIP and
PSTN/H323). Thing is Vocal doesn't seems to like the Notify message coming
from user "asterisk". If I can modify this I will have * seamlessly
providing full capabilities of voice mail to Vocal . I can't find a way to
get the VMWI to pass through to a SIP client of Vocal.

Cheers

SW

=
Message: 6
Date: Sun, 14 Dec 2003 13:33:09 -0500
To: [EMAIL PROTECTED]
From: John Todd <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] voice mail - sip:notify message
Reply-To: [EMAIL PROTECTED]

At 12:29 AM -0800 12/13/03, SW wrote:
>Hi folks,
>
>To provide MWI, * will send out a sip:notify message to the UA.
>
>The originator of this message is asterisk, as shown below;
>
>NOTIFY sip:[EMAIL PROTECTED]:5065 SIP/2.0
>Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21
>From: "asterisk" ;tag=as0ffb1bdc
><===
>To: 
>Contact: 
>Call-ID: [EMAIL PROTECTED]
>CSeq: 102 NOTIFY
>User-Agent: Asterisk PBX
>Event: message-summary
>Content-Type: application/simple-message-summary
>Content-Length: 38
>
>Messages-Waiting: yes
>Voicemail: 4/11
>
>Is there a way that I can change this 'originator' to a numeric value ?
>
>Cheers
>
>SW


SW -
   See the below conversation from April regarding the same topic.  I
don't think this became a patch or made it into CVS.

JT


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Re: [Asterisk-Users] IAX 1/2 registration

[Florian Overkamp]

Hi,

Citeren Mark Spencer <[EMAIL PROTECTED]>:

> Just noload chan_iax right?
> 
> Mark

Hmm, I understand there is little or no desire to really work on this type of 
issue, but noload chan_iax is slightly too rigorous for me. Consider this 
scenario:

I prefer using IAX2 with every partner I can talk to (IAXtel, uplinks) but I 
would still like users to be able to use IAX1 as a backward-compatible method. 
Therefore, noload chan_iax is not desirable.

Suggestion for those with this issue: Make the firewall (iptables) on your 
asterisk box drop or reject the IAX1 packets toward destinations where we do 
not want IAX1 to be used :-)


-- 
Best regards,
Florian Overkamp

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[Asterisk-Users] ignorepat

[Burak Balasaygun]

Hi 

  I have the following configuration at home one ZAPTEL interface connecting
to an FXO card and two SIP UAs connecting to asterisk locally. I have
configured extensions.conf such that dialing 9 on the SIP phones allows me to
dial an outbound number via the FXO interface . Works fine.


  What's not working is that pressing 9 should causes either  GS BT-100 phone
to reacquire a  dialtone since I have placed ignorepat => 9 in the config file.

  Any ideas?

  


rgds

burak

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[Asterisk-Users] outbound dialing / wait for keypress?

[tad]

hi there. i've got a question about outbound dialing.

here's my scenario:
1. i build a list of phone numbers from a database
2. when a call comes in, i begin dialing from the list
3. when an outbound call is answered, i connect the caller to that line.

so far, i'm able to do this with an agi script to dynamically build a
dialplan. i make repeated use of this perl call:
system("asterisk -r -x 'add extension 100,++$i,Dial(Zap/2/[number],15) into local;

to create a dialplan that looks like this:
s,1,Answer
s,2,agi,script.agi
s,3,Goto(local,100,1)
100,1,Dial(Zap/2/[number],15)
100,2,Dial(Zap/2/[another_number],15)
...

so far, so good - if the first number dialed is busy or times out,
it tries dialing the next number, and so on until one is answered at which
point the incoming and outgoing calls are connected.

my problem is how to handle the case where an outbound call is answered
by, say, an answering machine. ideally, i'd like Dial to work such that it
can be configured to play a recording and wait for a keypress before
considering the call answered. however, this doesn't seem to be the case.

i imagine that someone else has already encountered this problem - if so,
does anyone know what the solution is (i've already checked the various
resources, and haven't found anything). if not, does anyone have thoughts
on how best to approach this?

thanks,
tad




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Re: [Asterisk-Users] IAX 1/2 registration

[Dan]

Hi,

> From: "Mark Spencer" <[EMAIL PROTECTED]>
> Just noload chan_iax right?
>
> Mark

Yes, but... I still need IAX(1) till the problem with DIAX will be solved.
When a call is placed through Asterisk using IAX2 and DIAX, the call is
automatically droped apparently without any reason, after about 60'. It
works ok with IAX(1).
There is no other way to disable IAX(1) registration?

Best regards,
Dan


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Re: [Asterisk-Users] IAX 1/2 registration

[Mark Spencer]

Just noload chan_iax right?

Mark

On Sun, 14 Dec 2003, Dan wrote:

> Hi,
>
> How can I do to register an Asterisk server using just IAX2?
> If I have a line like the following in iax.conf
>
> register => user:[EMAIL PROTECTED]
>
> the server tries to register with both IAX and IAX2.
>
> if the line is :
> register => user:[EMAIL PROTECTED]:4569
>
> then the same, tries both on the same port (means IAX(1) on 4569 too).
>
> This can be shown using
> iax show registry
>
> It is possible to disable registration requests on IAX(1)?
>
> As IAXTEL does not support IAX1 anymore, I still have registration requests
> in IAX1..
>
> Thanks,
> Dan
>
>
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[Asterisk-Users] IAX 1/2 registration

[Dan]

Hi,

How can I do to register an Asterisk server using just IAX2?
If I have a line like the following in iax.conf

register => user:[EMAIL PROTECTED]

the server tries to register with both IAX and IAX2.

if the line is :
register => user:[EMAIL PROTECTED]:4569

then the same, tries both on the same port (means IAX(1) on 4569 too).

This can be shown using
iax show registry

It is possible to disable registration requests on IAX(1)?

As IAXTEL does not support IAX1 anymore, I still have registration requests
in IAX1..

Thanks,
Dan


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[Asterisk-Users] modem data calls through FXS / FXO digium cards failing

[john lawler]

Hi guys,

I think I posted on this issue before, but didn't get a response.  I've 
still not been able to resolve the issue.

I've got a small installation of Asterisk running one 4 port FXS Digium 
card and 1 FXO Digium card.  I'm having difficulty routing modem call 
through one of the extensions out through the FXO card.  By difficulty, 
I mean it won't work.  The calls won't even sync up.  If I disconnect 
the phone line from the FXO card and run the modem directly through that 
(completely bypassing Asterisk), the calls go through fine as usual.

Is there some sort of line quality I can specify for certain extensions 
to allow data calls to better pass through?  Any options I can specify 
on the Dial application that boost the quality or drop echo cancelling 
or are these strategies even going to do me any good?

Does anyone have a similar hardware configuration working?

Thanks.

jl
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Re: [Asterisk-Users] Two Stage Dialing for MF CAMA trunk

[Steve Rodgers]

Looking at the code in chan_zap.c, I only see options for feature group B and 
feature group D MF. The 2 stage MF signalling you are asking for isn't 
implmented in the latest asterisk source code. 

I would suggest you post a feature request detailing your 2 stage dialing 
requirement to bugs.digium.com as 911 support is something which could be 
useful to others.

Steve.





On Sunday 14 December 2003 09:53, William Flanagan wrote:
> Hi all,
>
> I am trying to setup a ZAP interface to do MF signaling for a handoff to a
> 911 tandem.  The signaling I need to perform on the T1 is this:
>
>
> 9-1-1 Tandem: Wink
> CLEC end office: KP (Keypulse) NPA ST (Start)
> 9-1-1 Tandem: Wink
> CLEC end office: KP I (Info Digit) NXX  ST
>
> As I'm not as familiar with the Zaptel configurabliity, I'm not really sure
> how to do this.  Do I dial twice or something similar in the dial plan?
>
> Thanks,
>
> William
>
> _
> Shop online for kids toys by age group, price range, and toy category at
> MSN Shopping. No waiting for a clerk to help you! http://shopping.msn.com
>
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Re: [Asterisk-Users] WiSip phone experiences?

[Miguel Cavazos]

thats an old review jeff pulver said firmware was better now, is it
worth 250USD? im thinking of one

does it autodetect ssid? works with low signal? sound quality?

Miguel
On Sun, 2003-12-14 at 18:34, John Todd wrote:
> http://www.loligo.com/asterisk/misc/WiSIP/
> 
> Works decently enough.  Still some software bugs to work out.
> 
> JT
> 
> 
> At 10:10 AM -0600 12/12/03, Michael Graves wrote:
> >
> >Anyone here have any good/bad things to say about first hand experience
> >with the new Wifi SIP phones? I am considering one for my office as an
> >alternative to FXS+Analog cordless.
> >
> >Thanks,
> >
> >  Michael Graves
> >
> >--
> >Michael Graves   [EMAIL PROTECTED]
> >Sr. Product Specialist  www.pixelpower.com
> >Pixel Power Inc.  [EMAIL PROTECTED]
> > 
> >FWD 54245
> >
> >"I've been everywhere man, I've been everywhere." - Hank Snow
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Re: [Asterisk-Users] WiSip phone experiences?

[John Todd]

http://www.loligo.com/asterisk/misc/WiSIP/

Works decently enough.  Still some software bugs to work out.

JT

At 10:10 AM -0600 12/12/03, Michael Graves wrote:
Anyone here have any good/bad things to say about first hand experience
with the new Wifi SIP phones? I am considering one for my office as an
alternative to FXS+Analog cordless.
Thanks,

 Michael Graves

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc.  [EMAIL PROTECTED]
FWD 54245

"I've been everywhere man, I've been everywhere." - Hank Snow
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Re: [Asterisk-Users] voice mail - sip:notify message

[John Todd]

At 12:29 AM -0800 12/13/03, SW wrote:
Hi folks,

To provide MWI, * will send out a sip:notify message to the UA.

The originator of this message is asterisk, as shown below;

NOTIFY sip:[EMAIL PROTECTED]:5065 SIP/2.0
Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21
From: "asterisk" ;tag=as0ffb1bdc
<===
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 38
Messages-Waiting: yes
Voicemail: 4/11
Is there a way that I can change this 'originator' to a numeric value ?

Cheers

SW


SW -
  See the below conversation from April regarding the same topic.  I 
don't think this became a patch or made it into CVS.

JT



At 7:59 PM -0500 4/27/03, Mark Spencer wrote:
Sounds like the SNOM expects to use our "Contact" to get a hold of us.  It
should be simple to add something like "voicemail=" in the general
section for setting the voicemail extension to use in the contact area.
Mark

On Thu, 24 Apr 2003, WipeOut . wrote:

 Here is the trace if anyone is interested..

 NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
 > Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK523b1b63
 From: "asterisk" ;tag=as3da6a846
 To: 
 Contact: 
 Call-ID: [EMAIL PROTECTED]
 CSeq: 102 NOTIFY
 User-Agent: Asterisk PBX
 Event: message-summary
 Content-Type: application/simple-message-summary
 Content-Length: 36
 Message-Waiting: yes
 Voicemail: 1/0
 > Hi,
 >
 > The MWI is working on the SNOM 200 but the problem is that when 
you press the MWI button it attempts to dial
 > "asterisk" 
 > where 192.168.1.200 is the IP address of my * box.
 >
 > How can I modify this so the return path is correct, which on my 
setup is extension 8500 for voicmailmain??
 >
 > > Thanks


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Re: [Asterisk-Users] Asterisk and fwd

[Olle E. Johansson]

Shoval Tomer wrote:

Hi, could anyone please provide a working sample of how to configure 
asterisk to connect to fwd?

I've tried the one at www.loligo.com  and it 
doesn't work. Not even when calling to 5.

Check the Asterisk FAQ at http://www.voip-info.org
Can you advise on how to debug sip (or trace and view sip packets) from 
the asterisk server to fwd so we can try to understand what exactly is 
not working?


Test the CLI command "sip debug".

Regards,
/O
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RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[asterisk]

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ProvoCityPower
Sent: Sunday, December 14, 2003 12:51 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)

> The question asked here, "why on earth you want to push fax data
> over a VoIP link at all. Fax compression isn't very efficient." may
> speak volumes about the future role of VOIP. My plans are to role
> out a VOIP connection to thousands of Customers. Many have legacy
> fax equipment. Am I to assume that they will toss out their fax
> equipment and join the PC based faxing crowd? I don't think I can
> control this. If I am going to offer an aternative to the legacy
> wire providers then I have to offer a comparable service. One that
> for example allows a customer to use a legacy fax machine in the
> same way.
>
> If this thought sidetracks the intent of this thread, you have my
> apologies, but I do think that legacy fax functionality is
> essential.
>
> Jeff

I think what you're talking about here is an absolute necessity if VOIP is
ever to compete for traditional analog service. A lot of users will not want
to change their ways. As stupid as I think FAXing and FAX machines are there
are still millions of people who prefer to send and recive a skewed low
resolution fax on thermal paper than the transmit and receive them over a
computer. Even though, I think FAXing is as ancient as sending a letter
through the mail it needs to be supported however the end user wishes to use
it.

With that being said, I think it simply comes down to the codecs being used.
I seem to recall that alaw and ulaw were acceptable codecs, though I know
nothing about this first hand. The other option is some sort of FAX proxy,
though that seems a little too complicated if you ask me.

-Bill


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Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[ProvoCityPower]

> - 
Original Message - > From: "Alastair Maw" <[EMAIL PROTECTED]>> To: <[EMAIL PROTECTED]>> Sent: Friday, December 12, 2003 4:58 PM> Subject: Re: 
[Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)> > 
> > On 12/12/03 13:56, Dan wrote:> > > This is because 
the fax is transmitted using the audio stream.> > > It is not 
related to the signaling protocol (SIP/IAX etc.) but to the> 
audio> > > codec used.> >> > Fax uses FSK 
modulation to transmit the data. If you compress this in a> > lossy 
way (GSM, MP3, whatever) then the integrity of the data is> > affected 
(more or less seriously depending on the codec used). Fax> > machines 
are generally quite picky, so compressing faxes is unlikely to> > 
work.> >> > I'm wondering why on earth you want to push fax 
data over a VoIP link at> > all. Fax compression isn't very 
efficient.> > Who wants that???> By fax data I mean the 
data contained in a fax (basically a picture file),> not the fax data 
audio stream.> It can be converted (GIF or JPG) then sent reliable over a 
slow IP link.> Just a special codec at both ends, able to pass the data 
to the fax app or a> fax machine connected to a TDM400/ AT or 
whatever.> > > It would be much less> > bandwidth 
intensive to decode the fax and send it over as proper data> > rather 
than audio, compressed using gzip/gif/png/something else.> > This 
is exactly what would be great to have it.
The question asked here, 
"why on earth you want to push fax data over 
a VoIP link atall. Fax compression isn't very efficient." may speak volumes 
about the future role of VOIP. My plans are to role out a VOIP connection to 
thousands of Customers. Many have legacy fax equipment. Am I to assume that they 
will toss out their fax equipment and join the PC based faxing crowd? I don't 
think I can control this. If I am going to offer an aternative to the legacy 
wire providers then I have to offer a comparable service. One that for example 
allows a customer to use a legacy fax machine in the same 
way.
 
If this thought 
sidetracks the intent of this thread, you have my apologies, but I do think that 
legacy fax functionality is essential.
 
Jeff

[Asterisk-Users] Two Stage Dialing for MF CAMA trunk

[William Flanagan]

Hi all,

I am trying to setup a ZAP interface to do MF signaling for a handoff to a 
911 tandem.  The signaling I need to perform on the T1 is this:

9-1-1 Tandem: Wink
CLEC end office: KP (Keypulse) NPA ST (Start)
9-1-1 Tandem: Wink
CLEC end office: KP I (Info Digit) NXX  ST
As I'm not as familiar with the Zaptel configurabliity, I'm not really sure 
how to do this.  Do I dial twice or something similar in the dial plan?

Thanks,

William

_
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[Asterisk-Users] CAMA MF signaling for a 911 Trunk

[William Flanagan]

Hi all,

I'm trying to get a handoff between me and a carrier going using Asterisk.  
I need to handoff using CAMA signaling.  On a Cisco, you can see the 
configuration types that I'm referring to on this site as an example:

http://www.cisco.com/en/US/products/hw/routers/ps221/prod_configuration_guide09186a008019b16e.html#35393

The example of the interaction is the following:

CLEC end office:  Seizure
9-1-1 Tandem: Wink
CLEC end office:  KP (Keypulse) NPA ST (Start)
9-1-1 Tandem: Wink
CLEC end office: KP I (Info Digit) NXX  ST
The call is originating from the VoIP side.  I have calling and called 
number in the SIP invite.  For the life of me, I can't figure out how to 
pulse out the digits in this format to make the signaling compliant with the 
interface.

Any suggestions would be appreciated...

William

_
Tired of slow downloads and busy signals?  Get a high-speed Internet 
connection! Comparison-shop your local high-speed providers here. 
https://broadband.msn.com

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Re: [Asterisk-Users] Asterisk and fwd

[Chris HARIGA]

If U want I can send U my settings. My FWD is 
working fine. Let me know...
 
Chris HARIGA
 

  - Original Message - 
  From: 
  Shoval 
  Tomer 
  To: [EMAIL PROTECTED] 
  
  Sent: Sunday, December 14, 2003 10:31 
  AM
  Subject: [Asterisk-Users] Asterisk and 
  fwd
  
  
  Hi, could anyone 
  please provide a working sample of how to configure asterisk to connect to 
  fwd?
  I've tried the one at 
  www.loligo.com and it doesn't work. Not 
  even when calling to 5.
   
  Can you advise on how 
  to debug sip (or trace and view sip packets) from the asterisk server to fwd 
  so we can try to understand what exactly is not 
  working?
   
  Please be advised 
  that our asterisk server is behind a NAT 
firewall.
   
  Thanks in 
  advance,
  Tom.
   

Re: [Asterisk-Users] iax parsing error

[Rich Adamson]

> I'm getting this error and don't know how to fix it.
> IAX2 seems to work though...
> 
> [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
>   == Manager registered action IAXpeers
>   == Parsing '/etc/asterisk/iax.conf': Found
> WARNING[1074439936]: File chan_iax2.c, Line 5465 (set_config): Ignoring port for now
>   == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
>   == Using TOS bits 4

I think this might be related to "port=5036" in iax.conf, knowing that
iax2 uses port 4569. If its there, comment it out and reload.



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[Asterisk-Users] Asterisk and fwd

[Shoval Tomer]

Hi,
could anyone please provide a working sample of how to configure asterisk to
connect to fwd?

I've
tried the one at www.loligo.com and it
doesn't work. Not even when calling to 5.

 

Can
you advise on how to debug sip (or trace and view sip packets) from the
asterisk server to fwd so we can try to understand what exactly is not working?

 

Please
be advised that our asterisk server is behind a NAT firewall.

 

Thanks
in advance,

Tom.

 

[Asterisk-Users] iax parsing error

[Hector Q.-datafull]

Hi,
I'm getting this error and don't know how to fix it.
IAX2 seems to work though...

[chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
  == Manager registered action IAXpeers
  == Parsing '/etc/asterisk/iax.conf': Found
WARNING[1074439936]: File chan_iax2.c, Line 5465 (set_config): Ignoring port for now
  == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
  == Using TOS bits 4

Thanks Hector.
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RE: [Asterisk-Users] voip-info.org DNS seems broken

[Patrick Cantwell]

All,

Please excuse my last post.  I meant to send to Olle directly (off-list),
and was operating on not enough sleep.
Of course, I picked the best message of all to send back to the list.

Thanks,
Pat


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Patrick
Cantwell
Sent: Sunday, December 14, 2003 9:26 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] voip-info.org DNS seems broken


Ollie,
It's been my experience that you can never have too many mirrors and/or at
least secondary DNS.

If you'd like me to do either for you, I'd be more than glad.

Thanks!
Pat

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E.
Johansson
Sent: Sunday, December 14, 2003 6:37 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] voip-info.org DNS seems broken


Thank you for all the offers. I think the list would prefer if you mailed me
off-list :-)

A gentle remark, no attack. You are really most kind to offer your
assistance.

And if any other users have any problems and can track them down, I would
appreciate
you mailing me so we can make the Wiki available to everyone all the time.

/O

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RE: [Asterisk-Users] voip-info.org DNS seems broken

[Patrick Cantwell]

Ollie,
It's been my experience that you can never have too many mirrors and/or at
least secondary DNS.

If you'd like me to do either for you, I'd be more than glad.

Thanks!
Pat

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E.
Johansson
Sent: Sunday, December 14, 2003 6:37 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] voip-info.org DNS seems broken


Thank you for all the offers. I think the list would prefer if you mailed me
off-list :-)

A gentle remark, no attack. You are really most kind to offer your
assistance.

And if any other users have any problems and can track them down, I would
appreciate
you mailing me so we can make the Wiki available to everyone all the time.

/O

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Re: [Asterisk-Users] voip-info.org DNS seems broken

[Olle E. Johansson]

Thank you for all the offers. I think the list would prefer if you mailed me off-list :-)

A gentle remark, no attack. You are really most kind to offer your assistance.

And if any other users have any problems and can track them down, I would appreciate
you mailing me so we can make the Wiki available to everyone all the time.
/O

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Re: [Asterisk-Users] voip-info.org DNS seems broken

[Linus Surguy]

> > For the last few days I can not resolve voip-info.org from many DNS
> > servers. It does resolve with some DNS servers but I suspect it may be
> > related more to caching.
> >
> I've alerted James of the problems. I haven't seen them myself, so its
hard
> for me to track.
>
> The wiki has become a too valuable resource for this community to
> continue to have these kind of problems. I assure you that James have
spent
> a lot of time trying to solve the DNS and performance problems. Any help
> tracking the problems down is appreciated!

If it helps, I'm willing for our company to offer secondary DNS.

Linus


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Re: [Asterisk-Users] voip-info.org DNS seems broken

[Brancaleoni Matteo]

If you want a mirror of you site, I can
give some space (how much is needed) , plus
mysql,php, blah blah blah
For free, of course.

Matteo.

Il dom, 2003-12-14 alle 10:00, Olle E. Johansson ha scritto:
> Bill Reid wrote:
> 
> > For the last few days I can not resolve voip-info.org from many DNS 
> > servers. It does resolve with some DNS servers but I suspect it may be 
> > related more to caching.
> > 
> I've alerted James of the problems. I haven't seen them myself, so its hard
> for me to track.
> 
> The wiki has become a too valuable resource for this community to
> continue to have these kind of problems. I assure you that James have spent
> a lot of time trying to solve the DNS and performance problems. Any help
> tracking the problems down is appreciated!
> 
> We will check the caching timings in the zones. There's also an inconsistency
> between the zone and the DNS delegations from the .org TLD that will be fixed.
> 
> /O
> 
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-- 
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl

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Re: [Asterisk-Users] * Party in Paris

[Stephen Wingfield]

Just to repeat - [EMAIL PROTECTED]

Please reply if you wish.

- Original Message -
From: "Stephen Wingfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, December 13, 2003 12:26 PM
Subject: Re: [Asterisk-Users] * Party in Paris


> SATURDAY 20th
>
> I have had far fewer emails than the noise created earlier about Mark's
> arrival in Paris. Everyone who has contacted me I have replied to once.
> Again please - if you want to come please email me at [EMAIL PROTECTED] I
will
> set a venue and time tomorrow evening to email to all concerned.
>
> Steve
>
> - Original Message -
> From: "marrandy" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, December 11, 2003 7:58 PM
> Subject: Re: [Asterisk-Users] * Party in Paris
>
>
> > On Thursday 11 December 2003 12:53 pm, Bob Knight wrote:
> > > Is the party at the Paris Hilton?
> > >
> > > sorry, couldn't help it...
> > >
> > > --
> > > Bob Knight
> >
> >
> > Bob...I'm really surprised at you !!!
> >
> > I thought you would have said,  'Is the party in Paris Hilton'
> >
> > lol  ;-)
> >
> > --
> > Q: What's hard going in and soft and sticky coming out?
> > A: Chewing gum.
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
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>
>

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Re: [Asterisk-Users] VoiceMail Password problems

[rnc Info Lists]

> Hi!
>
>> I don't get why people always say dtmfmode=info mine works fine with
>> rfc2833.
>> bkw
>
> Dunno. I tried rfc2833 first, and had exactly the same problem as
> described below with voicemail (but only there). Info then worked just
> fine (as obviously also confirmed by this user here).
>
> Is there any other setup/setting that has influence on DTMF detection?
> Like NAT (yes for me) or anything else? However, more likely it's simply
> a GS firmware thing (4.17 on mine) - or production (hardware) issue with
> GS.
>
>> > > "Incorrect Password '4433211' for user '2000' (context =)"
>> >
>> > This is a FAQ: use dtmfmode=info in your sip.conf for your Grandstream
>> > Note: Don't forget to "reload" after modifying sip.conf.
>

I am running GS Firmware 1.0.3.78 with Send DTMF = Via SIP INFO

sip.conf for that phone is:
[2001]
type=friend
username=2001
secret=test2
host=dynamic
context=local-extensions

Am able to access VM with no problem and use the phone via *->IAXtel to
access other VM systems at USA toll-free numbers.

Robert
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[Asterisk-Users] Unable to call from SNOM 200 to IP 7905G

[tony banks]

Hello 

I have configured IP 7905G  and SNOM 200  for Asterisk.  Now problem is that I can 
call from IP 7905G to SNOM 200 but not the other way round. Instead I get "FORBIDDEN" 
Message on SNOM 200 LCD when ever I try to call IP7905 phone and asterisk generate 
following messages..

Please note 810, (129.82.44.222) are the phone number and IP of for my 7905G phone and 
910, (129.82.44.226) are the phone number and IP for SNOM 200 in my configuration. 
Asterisk server IP is 129.82.44.215

Could you please help me in finding out the problem. 

/**/
   Messages generated by Asterisk Server
/**/

-- Executing Dial("SIP/910-f4ab", "sip/810|30") in new stack
-- Called 810
-- Got SIP response 404 "Not Found" back from 129.82.44.222
-- SIP/810-fafc is circuit-busy
  == Everyone is busy at this time
WARNING[1209214400]: File pbx.c, Line 1160 (pbx_extension_helper): No application ' 
voicemail2' for extension (default, 810, 102)
  == Spawn extension (default, 810, 102) exited non-zero on 'SIP/910-f4ab'
NOTICE[1125329600]: File chan_sip.c, Line 5298 (handle_request): Registration from 
'' failed for '129.82.44.222'
NOTICE[1125329600]: File chan_sip.c, Line 5298 (handle_request): Registration from 
'' failed for '129.82.44.222'
NOTICE[1125329600]: File chan_sip.c, Line 5298 (handle_request): Registration from 
'' failed for '129.82.44.222'
NOTICE[1125329600]: File chan_sip.c, Line 5298 (handle_request): Registration from 
'' failed for '129.82.44.222'
NOTICE[1125329600]: File chan_sip.c, Line 5298 (handle_request): Registration from 
'' failed for '129.82.44.222'
NOTICE[1125329600]: File chan_sip.c, Line 5298 (handle_request): Registration from 
'' failed for '129.82.44.222'


Regards 
Tony

Re: [Asterisk-Users] VoiceMail Password problems

[Eric Wieling]

It seems to be a Grandstream specific thing.  Prolly specific to certian
GS firmware revs

On Sat, 2003-12-13 at 15:18, Brian West wrote:
> I don't get why people always say dtmfmode=info mine works fine with
> rfc2833.
> 
> bkw
> 
> On Sat, 13 Dec 2003, Philipp von Klitzing wrote:
> 
> > > "Incorrect Password '4433211' for user '2000' (context =)"
> >
> > This is a FAQ: use dtmfmode=info in your sip.conf for your Grandstream
> > Note: Don't forget to "reload" after modifying sip.conf.
> >
> > Cheers, Philipp
> >
> >
> > ___
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> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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-- 
Go to http://www.digium.com/index.php?menu=documentation and look at
the "Unofficial Links" section.  This section has links to a wide
variety of 3rd party Asterisk related pages.  My page is the
"Asterisk Resource Pages".

BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

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Re: [Asterisk-Users] voip-info.org DNS seems broken

[Olle E. Johansson]

Bill Reid wrote:

For the last few days I can not resolve voip-info.org from many DNS 
servers. It does resolve with some DNS servers but I suspect it may be 
related more to caching.

I've alerted James of the problems. I haven't seen them myself, so its hard
for me to track.
The wiki has become a too valuable resource for this community to
continue to have these kind of problems. I assure you that James have spent
a lot of time trying to solve the DNS and performance problems. Any help
tracking the problems down is appreciated!
We will check the caching timings in the zones. There's also an inconsistency
between the zone and the DNS delegations from the .org TLD that will be fixed.
/O

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