Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.

[Fons van der Beek]

I use MiSDN
so


head -n 1 /proc/zaptel/*
 gives

head: cannot open `/proc/zaptel/*' for reading: No such file or directory


Grygoriy Dobrovolskyy schreef:
> paste output of
> head -n 1 /proc/zaptel/*
>
> 2008/3/9, Fons van der Beek <[EMAIL PROTECTED] 
> >:
>
> what clock?
> rxclock
> crystalclock
>
>
> I currently use
>
> "card=1,0x4"
>
>
>
>
>
>
> Grygoriy Dobrovolskyy schreef:
>
> > Well i have installed asterisk on spare system to replace old one,
> > with new tyan motherboard, surprise came when i installed digium
> > fxs/fxo and b410 p, system unstable, random bips on start, misdn
> > module Not loading, heh, old system worked on asus p5nd2 sli
> without a
> > problem with them. There were 2 reasons why i wanted change mobo 1:
> > chipset on asus card generated too much heat, 2: i had a new 3ware
> > controller. So you never know really.
> >
> > Loud scrathing sound? sometimes a card problem, try on other
> hardware.
> >
> > Pci interrupts, also maybe sync problem (you can enable b410
> clock in
> > misdn-init.conf)
> >
> >
> > Also turn off all sound/usb/etc unused devices.
> >
> >
>
> > 2008/3/8, Royce Souther <[EMAIL PROTECTED]
>   >>:
>
> >
> > [EMAIL PROTECTED] lspci -v -s 01:07.0
> > pcilib: Cannot open /sys/bus/pci/devices
> > :01:07.0 Communication controller: Tiger Jet Network Inc.
> > Tiger3XX Modem/ISDN interface
> > Subsystem: Unknown device b1d9:0003
> > Flags: bus master, medium devsel, latency 32, IRQ 10
> > I/O ports at ac00 [size=256]
> > Memory at fbfff000 (32-bit, non-prefetchable) [size=4K]
> > Capabilities: [40] Power Management version 2
> >
> > This is all I know about it. The client bought them about a
> month
> > ago and installed them himself then asked me to setup the
> Asterisk
> > program for him. The problem motherboard was a four year old
> > Gigabyte with a Promise IDE controller.
> >
> > The new motherboard that works well is an ASUS but I don't know
> > anything else about it.
> >
> > On Sat, Mar 8, 2008 at 9:09 AM, Steve Totaro
> > <[EMAIL PROTECTED]
> 
>
> >  >> wrote:
> >
> > Which revision of the Digium TDM400?
> >
> > On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther
>
> > <[EMAIL PROTECTED] 
> >> wrote:
> > > IRQ's seem to have been the problem. Thanks Steve
> Totaro for
> > that tip.
> > >
> > > The Digium cards were at the same IRQ as the IDE
> controller,
> > I moved the
> > > cards and hard drives to a different system and all is
> good now.
> > >
> > > Thanks.
> > >
> > >
> > >
> > > On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro
> > > <[EMAIL PROTECTED]
> 
>
> >  >> wrote:
> > > > Check for IRQ issues, move the card to a different slot.
> > > >
> > > > You could ask permission to record calls so maybe
> you can
> > hear the
> > > > sound yourself.
> > > >
> > > > I would then go ahead and swap out cards.  I have had
> > TDM400 with bad
> > > > modules and also bad ports on the cards themselves,
> so it
> > could a
> > > > hardware issue.
> > > >
> > > > This is what I suspect, especially if you did not
> put any
> > surge
> > > > suppression on your telco lines.  Usually, at least
> in my
> > experience,
> > > > ticks or beeps indicate IRQ, hissing or loud static
> > indicate something
> > > > with/on the board is bad.  ALWAYS use surge
> suppression on
> > your lines!
> > > >
> > > > Thanks,
> > > > Steve Totaro
> > > >
> > > >
> > > >
> > > >
> > > > On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther
>
> > <[EMAIL PROTECTED] 
> >> wrote:
> > > > > I have setup a few Asterisk systems for customers
> using
> > Digium TDM400
> > > cards
> > > > > and Aastra phones. No probl

[asterisk-users] Read function

[Daniel Suleyman]

Dear all, interesting behaivior of the Read function.

I have  SIP phone(XLITE) attached to my Asterisk.

SIP.conf
[7007]
type=friend
qualify=900
host=192.168.85.27
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=alaw
allow=ulaw

extensions.conf

1,1,Answer;
1,2,Read(CNT,,2)
1,3,SayNaumber(${CNT})

Function read do not write anything to CNT or write "".

in SayNumber it is always equel to ""; even if I previously defins CNT = 123;

And read function not exit if I pres #.(I think it is exit only on timeout)

Strange can anybody point on mistake?

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Re: [asterisk-users] Fwd: {s} - extension

[Daniel Suleyman]

ok, then I'm not understanding something.
How I can call with xlite to my Asterisk not sending extension?
when I want to call I pick up phone, hear ring (piii) and I need
to type some extension otherwise nothing hapens

2008/3/8, Tilghman Lesher <[EMAIL PROTECTED]>:
> On Saturday 08 March 2008 02:52:39 Daniel Suleyman wrote:
> > Here is the log, my extensions is in the default section
> 
> > INVITE sip:[EMAIL PROTECTED] SIP/2.0
> 
> > [ Context 'default' created by 'pbx_config' ]
> >   '7007' => 1. Goto(7007-${CNT}|1)
> > [pbx_config] '7007-2' =>   1. Set(GLOBAL(CNT)=1)
> >  [pbx_config] 2. Answer()   [pbx_config] 3.
> > Playback(hello-world|skip) [pbx_config] 4. Hangup()
> >   [pbx_config] '7008' => 1.
> > Set(GLOBAL(connid)=0)  [pbx_config] 2.
> > Set(GLOBAL(resultid)=0)[pbx_config] 3.
> > Set(GLOBAL(fetchid)=0) [pbx_config] 4. MYSQL(Connect
> > connid localhost root test test) [pbx_config]
> > 5. MYSQL(Query resultid ${connid} Select a from a)
> > [pbx_config]
> > 6. MYSQL(Fetch fetchid ${resultid} a)
> > [pbx_config] 7. MYSQL(Clear ${resultid})   [pbx_config] 8.
> > MYSQL(Disconnect ${connid})[pbx_config] 9. goto(${a}|1)
> >   [pbx_config] 's' =>1. Answer()
> >[pbx_config] '_7007-.' =>  1.
> > Set(GLOBAL(CNT)=$[${CNT}+1])   [pbx_config] 2. Answer()
> >   [pbx_config] 3. Playback(tt-weasels|skip)
> >  [pbx_config] 4. Hangup()
> > [pbx_config]
> 
>
> I don't see extension "999" anywhere in your default context.  "s" is ONLY
> used when an extension is NOT sent, not as a default when nothing matches.
>
> --
> Tilghman
>
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Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.

[Royce Souther]

[EMAIL PROTECTED] cat /proc/zaptel/*
Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"

   1 WCTDM/0/0 FXOKS (In use)
   2 WCTDM/0/1 FXSKS (In use)
   3 WCTDM/0/2 FXSKS (In use)
   4 WCTDM/0/3 FXSKS (In use)
Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2"

   5 WCTDM/1/0 FXOKS (In use)
   6 WCTDM/1/1 FXSKS (In use)
   7 WCTDM/1/2 FXSKS (In use)
   8 WCTDM/1/3 FXSKS (In use)
[EMAIL PROTECTED]


On Sat, Mar 8, 2008 at 7:08 PM, Grygoriy Dobrovolskyy <[EMAIL PROTECTED]>
wrote:

> paste output of
> head -n 1 /proc/zaptel/*
>
> 2008/3/9, Fons van der Beek <[EMAIL PROTECTED]>:
>
> > what clock?
> > rxclock
> > crystalclock
> >
> >
> > I currently use
> >
> > "card=1,0x4"
> >
> >
> >
> >
> >
> >
> > Grygoriy Dobrovolskyy schreef:
> >
> > > Well i have installed asterisk on spare system to replace old one,
> > > with new tyan motherboard, surprise came when i installed digium
> > > fxs/fxo and b410 p, system unstable, random bips on start, misdn
> > > module Not loading, heh, old system worked on asus p5nd2 sli without a
> > > problem with them. There were 2 reasons why i wanted change mobo 1:
> > > chipset on asus card generated too much heat, 2: i had a new 3ware
> > > controller. So you never know really.
> > >
> > > Loud scrathing sound? sometimes a card problem, try on other hardware.
> > >
> > > Pci interrupts, also maybe sync problem (you can enable b410 clock in
> > > misdn-init.conf)
> > >
> > >
> > > Also turn off all sound/usb/etc unused devices.
> > >
> > >
> >
> > > 2008/3/8, Royce Souther <[EMAIL PROTECTED]  > >>:
> >
> > >
> > > [EMAIL PROTECTED] lspci -v -s 01:07.0
> > > pcilib: Cannot open /sys/bus/pci/devices
> > > :01:07.0 Communication controller: Tiger Jet Network Inc.
> > > Tiger3XX Modem/ISDN interface
> > > Subsystem: Unknown device b1d9:0003
> > > Flags: bus master, medium devsel, latency 32, IRQ 10
> > > I/O ports at ac00 [size=256]
> > > Memory at fbfff000 (32-bit, non-prefetchable) [size=4K]
> > > Capabilities: [40] Power Management version 2
> > >
> > > This is all I know about it. The client bought them about a month
> > > ago and installed them himself then asked me to setup the Asterisk
> > > program for him. The problem motherboard was a four year old
> > > Gigabyte with a Promise IDE controller.
> > >
> > > The new motherboard that works well is an ASUS but I don't know
> > > anything else about it.
> > >
> > > On Sat, Mar 8, 2008 at 9:09 AM, Steve Totaro
> > > <[EMAIL PROTECTED]
> >
> > > > wrote:
> > >
> > > Which revision of the Digium TDM400?
> > >
> > > On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther
> >
> > > <[EMAIL PROTECTED] > wrote:
> > > > IRQ's seem to have been the problem. Thanks Steve Totaro for
> > > that tip.
> > > >
> > > > The Digium cards were at the same IRQ as the IDE controller,
> > > I moved the
> > > > cards and hard drives to a different system and all is good
> > now.
> > > >
> > > > Thanks.
> > > >
> > > >
> > > >
> > > > On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro
> > > > <[EMAIL PROTECTED]
> >
> > > > wrote:
> > > > > Check for IRQ issues, move the card to a different slot.
> > > > >
> > > > > You could ask permission to record calls so maybe you can
> > > hear the
> > > > > sound yourself.
> > > > >
> > > > > I would then go ahead and swap out cards.  I have had
> > > TDM400 with bad
> > > > > modules and also bad ports on the cards themselves, so it
> > > could a
> > > > > hardware issue.
> > > > >
> > > > > This is what I suspect, especially if you did not put any
> > > surge
> > > > > suppression on your telco lines.  Usually, at least in my
> > > experience,
> > > > > ticks or beeps indicate IRQ, hissing or loud static
> > > indicate something
> > > > > with/on the board is bad.  ALWAYS use surge suppression on
> > > your lines!
> > > > >
> > > > > Thanks,
> > > > > Steve Totaro
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther
> >
> > > <[EMAIL PROTECTED] > wrote:
> > > > > > I have setup a few Asterisk systems for customers using
> > > Digium TDM400
> > > > cards
> > > > > > and Aastra phones. No problems with sound quality at all
> > > except at this
> > > > one
> > > > > > site.
> > > > > >
> > > > > > Every time I try their system I don't hear any problems

Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.

[Royce Souther]

Not sure what you are asking for here? It looks like you have a config file
that sets a clock speed but I do not know what file that is.

How can I find this informaiton?

On Sat, Mar 8, 2008 at 5:48 PM, Fons van der Beek <[EMAIL PROTECTED]>
wrote:

> what clock?
> rxclock
> crystalclock
>
>
> I currently use
>
> "card=1,0x4"
>

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Re: [asterisk-users] How Do I continue after Dial Command ??

[Grygoriy Dobrovolskyy]

Two ways, use n priority or add 'g' iption in dial command.

2008/3/9, Jim Duda <[EMAIL PROTECTED]>:
>
> How do I get a context to continue to execute commands after the caller
> hangs up after a Dial command?  I'm using the "e" option to the Dial
> application.  I though the "e" option would allow the context to
> continue.  This doesn't want to work for me.
>
> I'm using asterisk-1.6.beta5
>
> I never get to "3" below.  I get a message saying the "2" ended with a
> non-zero status.
>
> [sphinx]
> exten => s,1,AGI(MisterHouse.agi,"Sphinx Connect")
> exten => s,2,Dial(CONSOLE/1,,e)
> exten => s,3,AGI(MisterHouse.agi,"Sphinx Disconnect")
> exten => s,4,Hangup
>
> Thanks,
>
> Jim
>
>
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Re: [asterisk-users] PRI suppliers in Switzerland

[Grygoriy Dobrovolskyy]

That's why it is confusing call it E1 and i will be happy

2008/3/9, Hans Witvliet <[EMAIL PROTECTED]>:
>
> On Sat, 2008-03-08 at 23:24 +0100, Grygoriy Dobrovolskyy wrote:
> > I had same problem in france, not much choice, i have ordered from
> > germany http://www.voipango.de
> >
> > This is not an advertisement i am not working for them.
> >
> > Are you sure about PRI ? in europe it's bri as i heard, PRI is in
> > usa??
> >
>
>
> No, it's also called PRI (or ISDN30), but its a different one...
> USA-pri is T1, 1,5 Mbps 23 * 64kbps channels + one 64 kbps D-channel.
>
> In Europe pri is E1, 2,048 Mbps
> made up of 32 * 64 kbps but two of those channels are not used for
> voice.
> One is used for signalling (also known as D-channel) and the other (used
> to be needed) for synchronisation.
>
> BRI is the isdn variant for home use: 144 kbps. Two 64kbps voice
> channels and a much smaller D-channel (16kbps) for signalling.
>
>
> hw
>
>
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Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.

[Grygoriy Dobrovolskyy]

paste output of
head -n 1 /proc/zaptel/*

2008/3/9, Fons van der Beek <[EMAIL PROTECTED]>:
>
> what clock?
> rxclock
> crystalclock
>
>
> I currently use
>
> "card=1,0x4"
>
>
>
>
>
>
> Grygoriy Dobrovolskyy schreef:
>
> > Well i have installed asterisk on spare system to replace old one,
> > with new tyan motherboard, surprise came when i installed digium
> > fxs/fxo and b410 p, system unstable, random bips on start, misdn
> > module Not loading, heh, old system worked on asus p5nd2 sli without a
> > problem with them. There were 2 reasons why i wanted change mobo 1:
> > chipset on asus card generated too much heat, 2: i had a new 3ware
> > controller. So you never know really.
> >
> > Loud scrathing sound? sometimes a card problem, try on other hardware.
> >
> > Pci interrupts, also maybe sync problem (you can enable b410 clock in
> > misdn-init.conf)
> >
> >
> > Also turn off all sound/usb/etc unused devices.
> >
> >
>
> > 2008/3/8, Royce Souther <[EMAIL PROTECTED] >:
>
> >
> > [EMAIL PROTECTED] lspci -v -s 01:07.0
> > pcilib: Cannot open /sys/bus/pci/devices
> > :01:07.0 Communication controller: Tiger Jet Network Inc.
> > Tiger3XX Modem/ISDN interface
> > Subsystem: Unknown device b1d9:0003
> > Flags: bus master, medium devsel, latency 32, IRQ 10
> > I/O ports at ac00 [size=256]
> > Memory at fbfff000 (32-bit, non-prefetchable) [size=4K]
> > Capabilities: [40] Power Management version 2
> >
> > This is all I know about it. The client bought them about a month
> > ago and installed them himself then asked me to setup the Asterisk
> > program for him. The problem motherboard was a four year old
> > Gigabyte with a Promise IDE controller.
> >
> > The new motherboard that works well is an ASUS but I don't know
> > anything else about it.
> >
> > On Sat, Mar 8, 2008 at 9:09 AM, Steve Totaro
> > <[EMAIL PROTECTED]
>
> > > wrote:
> >
> > Which revision of the Digium TDM400?
> >
> > On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther
>
> > <[EMAIL PROTECTED] > wrote:
> > > IRQ's seem to have been the problem. Thanks Steve Totaro for
> > that tip.
> > >
> > > The Digium cards were at the same IRQ as the IDE controller,
> > I moved the
> > > cards and hard drives to a different system and all is good
> now.
> > >
> > > Thanks.
> > >
> > >
> > >
> > > On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro
> > > <[EMAIL PROTECTED]
>
> > > wrote:
> > > > Check for IRQ issues, move the card to a different slot.
> > > >
> > > > You could ask permission to record calls so maybe you can
> > hear the
> > > > sound yourself.
> > > >
> > > > I would then go ahead and swap out cards.  I have had
> > TDM400 with bad
> > > > modules and also bad ports on the cards themselves, so it
> > could a
> > > > hardware issue.
> > > >
> > > > This is what I suspect, especially if you did not put any
> > surge
> > > > suppression on your telco lines.  Usually, at least in my
> > experience,
> > > > ticks or beeps indicate IRQ, hissing or loud static
> > indicate something
> > > > with/on the board is bad.  ALWAYS use surge suppression on
> > your lines!
> > > >
> > > > Thanks,
> > > > Steve Totaro
> > > >
> > > >
> > > >
> > > >
> > > > On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther
>
> > <[EMAIL PROTECTED] > wrote:
> > > > > I have setup a few Asterisk systems for customers using
> > Digium TDM400
> > > cards
> > > > > and Aastra phones. No problems with sound quality at all
> > except at this
> > > one
> > > > > site.
> > > > >
> > > > > Every time I try their system I don't hear any problems
> > but they tell me
> > > > > that it is really bad. They describe it a a loud
> > scratching sound.
> > > > >
> > > > > Are there any tests that can be done to pinpoint the
> > problem? Has anyone
> > > > > seen this before? Are there known causes for this?
> > > > >
> > > > > --
> > > > > Open Source: To innovate then create
> > > > > Proprietary: To imitate then litigate
> > > >
> > > >
> > > >
> > > > > ___
> > > > >  -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com --
> > > > >
> > > > >  asterisk-users mailing list
> > > > >  To UNSUBSCRIBE or update optio

[asterisk-users] How Do I continue after Dial Command ??

[Jim Duda]

How do I get a context to continue to execute commands after the caller 
hangs up after a Dial command?  I'm using the "e" option to the Dial 
application.  I though the "e" option would allow the context to 
continue.  This doesn't want to work for me.

I'm using asterisk-1.6.beta5

I never get to "3" below.  I get a message saying the "2" ended with a 
non-zero status.

[sphinx]
exten => s,1,AGI(MisterHouse.agi,"Sphinx Connect")
exten => s,2,Dial(CONSOLE/1,,e)
exten => s,3,AGI(MisterHouse.agi,"Sphinx Disconnect")
exten => s,4,Hangup

Thanks,

Jim


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Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.

[Fons van der Beek]

what clock?
rxclock
crystalclock


I currently use

"card=1,0x4"






Grygoriy Dobrovolskyy schreef:
> Well i have installed asterisk on spare system to replace old one, 
> with new tyan motherboard, surprise came when i installed digium 
> fxs/fxo and b410 p, system unstable, random bips on start, misdn 
> module Not loading, heh, old system worked on asus p5nd2 sli without a 
> problem with them. There were 2 reasons why i wanted change mobo 1: 
> chipset on asus card generated too much heat, 2: i had a new 3ware 
> controller. So you never know really.
>
> Loud scrathing sound? sometimes a card problem, try on other hardware.
>
> Pci interrupts, also maybe sync problem (you can enable b410 clock in 
> misdn-init.conf)
>
>
> Also turn off all sound/usb/etc unused devices.
>
>
> 2008/3/8, Royce Souther <[EMAIL PROTECTED] >:
>
> [EMAIL PROTECTED] lspci -v -s 01:07.0
> pcilib: Cannot open /sys/bus/pci/devices
> :01:07.0 Communication controller: Tiger Jet Network Inc.
> Tiger3XX Modem/ISDN interface
> Subsystem: Unknown device b1d9:0003
> Flags: bus master, medium devsel, latency 32, IRQ 10
> I/O ports at ac00 [size=256]
> Memory at fbfff000 (32-bit, non-prefetchable) [size=4K]
> Capabilities: [40] Power Management version 2
>
> This is all I know about it. The client bought them about a month
> ago and installed them himself then asked me to setup the Asterisk
> program for him. The problem motherboard was a four year old
> Gigabyte with a Promise IDE controller.
>
> The new motherboard that works well is an ASUS but I don't know
> anything else about it.
>
> On Sat, Mar 8, 2008 at 9:09 AM, Steve Totaro
> <[EMAIL PROTECTED]
> > wrote:
>
> Which revision of the Digium TDM400?
>
> On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther
> <[EMAIL PROTECTED] > wrote:
> > IRQ's seem to have been the problem. Thanks Steve Totaro for
> that tip.
> >
> > The Digium cards were at the same IRQ as the IDE controller,
> I moved the
> > cards and hard drives to a different system and all is good now.
> >
> > Thanks.
> >
> >
> >
> > On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro
> > <[EMAIL PROTECTED]
> > wrote:
> > > Check for IRQ issues, move the card to a different slot.
> > >
> > > You could ask permission to record calls so maybe you can
> hear the
> > > sound yourself.
> > >
> > > I would then go ahead and swap out cards.  I have had
> TDM400 with bad
> > > modules and also bad ports on the cards themselves, so it
> could a
> > > hardware issue.
> > >
> > > This is what I suspect, especially if you did not put any
> surge
> > > suppression on your telco lines.  Usually, at least in my
> experience,
> > > ticks or beeps indicate IRQ, hissing or loud static
> indicate something
> > > with/on the board is bad.  ALWAYS use surge suppression on
> your lines!
> > >
> > > Thanks,
> > > Steve Totaro
> > >
> > >
> > >
> > >
> > > On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther
> <[EMAIL PROTECTED] > wrote:
> > > > I have setup a few Asterisk systems for customers using
> Digium TDM400
> > cards
> > > > and Aastra phones. No problems with sound quality at all
> except at this
> > one
> > > > site.
> > > >
> > > > Every time I try their system I don't hear any problems
> but they tell me
> > > > that it is really bad. They describe it a a loud
> scratching sound.
> > > >
> > > > Are there any tests that can be done to pinpoint the
> problem? Has anyone
> > > > seen this before? Are there known causes for this?
> > > >
> > > > --
> > > > Open Source: To innovate then create
> > > > Proprietary: To imitate then litigate
> > >
> > >
> > >
> > > > ___
> > > >  -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
> > > >
> > > >  asterisk-users mailing list
> > > >  To UNSUBSCRIBE or update options visit:
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> > > >
> > >
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> > >
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Re: [asterisk-users] PRI suppliers in Switzerland

[Hans Witvliet]

On Sat, 2008-03-08 at 23:24 +0100, Grygoriy Dobrovolskyy wrote:
> I had same problem in france, not much choice, i have ordered from
> germany http://www.voipango.de
> 
> This is not an advertisement i am not working for them.
> 
> Are you sure about PRI ? in europe it's bri as i heard, PRI is in
> usa??
> 

No, it's also called PRI (or ISDN30), but its a different one...
USA-pri is T1, 1,5 Mbps 23 * 64kbps channels + one 64 kbps D-channel.

In Europe pri is E1, 2,048 Mbps
made up of 32 * 64 kbps but two of those channels are not used for
voice.
One is used for signalling (also known as D-channel) and the other (used
to be needed) for synchronisation.

BRI is the isdn variant for home use: 144 kbps. Two 64kbps voice
channels and a much smaller D-channel (16kbps) for signalling.

hw

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Re: [asterisk-users] PRI suppliers in Switzerland

[Gordon Henderson]

On Sat, 8 Mar 2008, Grygoriy Dobrovolskyy wrote:

> I had same problem in france, not much choice, i have ordered from germany
> http://www.voipango.de
>
> This is not an advertisement i am not working for them.
>
> Are you sure about PRI ? in europe it's bri as i heard, PRI is in usa??

We have both all over Europe, and they're both installed as required. If 
you need a small number of lines then you get a small number of BRIs. For 
more capacity, PRI is usually the way to go.

In the UK (From BT), PRI is commonly called ISDN30, (up to 30 channels), 
BRI is ISDN2. (2 channels, oddly enough :)

Gordon

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Re: [asterisk-users] Background: reading the digits correctly, buffering it, waiting the sound message to complete

[bilal ghayyad]

Dear Noah;

In that case, I can say that I have the following
issues need to be resolved:

1) When I press the digit (the first digit in the
entered number", asterisk recognize it as duplicated,
for example if I pressed "140" then asterisk recognize
it as "1140" or for example if I entered "2" then it
recognize it as "22" and so on. This happen if I did
not wait the sound message of the Background to be
completed, but if I let the sound message of the
Background to be completed, then this problem almost
does not happen (it might happen, but mostly does not
happen). So what is the solution to avoid this
duplication recognization in the first entered digit?

2) Is it possible in each menu to clear any entered
digit (in the begin of the senario) before start
playing any sound message? That help to avoid the
problem that might happen when I entered "1" and
asterisk recognize it as "11" so it will run the step
related to first 1 in the first menu and the step
related to second 1 in the second menu. How can I do
such clearance in the begin of the menu senario?

I can use Playback instead of Background, but also
sometimes we do not need to hear the sound message
completely because we already knows the complete
senario, so how can we avoid the duplication in the
first digit recognization when using Background
function and without letting the played sound file to
be completed (we need to start enter the digit once we
start hear the sound message).

Any help?
Regards
Bilal

-

Hi Bilal -

>  1) If I pressed 1 twice (11), so it runs the step
>  related to first 1 and then it runs the step
related
>  to second 1, so it does buffering for my input and
run
>  two steps, how can I make it run only the step
related
>  to first entered digit "1" and does not do
buffering
>  (so ignoring the second input)?

I don't think asterisk is doing any buffering here. 
It's just
reacting quickly enough that it sends the first '1' to
the first IVR
menu and then advances to the second IVR menu and
sends the next '1'
there.


>  2) If I waited the sound message to completed and
then
>  I entered my digits, it reads it correctly without
>  duplication in the DTMF and without any problem,
but
>  if i entered my digits from the beginning of the
sound
>  message (without waiting to complete it), then
>  asterisk might read the digits duplicated
(specially
>  first entered digit), how can I resolve it? Does
>  playback resolve my problem? Any solution for
>  Background to avoid this behaviour?

I can think of two things:

1. Use different extensions in the second IVR menu so
1 is not a valid
 response
2. As you suggested, insert a brief Playback() into
the second menu
that says "Please listen to the following options"


>  Note: I set the relaxdtmf=yes and I made the
>  toneduration=500.

If your DTMF is working correctly, I would not try to
make any changes
there to influence menu selections.


- Noah





  

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Re: [asterisk-users] PRI suppliers in Switzerland

[Grygoriy Dobrovolskyy]

I had same problem in france, not much choice, i have ordered from germany
http://www.voipango.de

This is not an advertisement i am not working for them.

Are you sure about PRI ? in europe it's bri as i heard, PRI is in usa??

2008/3/8, Chris Bagnall <[EMAIL PROTECTED]>:
>
> Greetings list,
>
> I posted this to the -biz list a few days ago. In hindsight, I think it
> would have been more appropriate posted here, so apologies to those on both
> lists who've now seen this twice.
>
> I have had a request to provide 2x PRIs to a site in Lausanne,
> Switzerland, but my knowledge of the Swiss Telco market is non-existent.
>
> Are there any folks on the list who've experience in this market who might
> be able to offer some hints, or companies to approach?
>
> Thanks in advance.
>
> Regards,
>
> Chris
> --
> C.M. Bagnall, Director, Minotaur I.T. Limited
> For full contact details visit http://www.minotaur.it
> This email is made from 100% recycled electrons
>
>
>
>
>
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Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.

[Grygoriy Dobrovolskyy]

Well i have installed asterisk on spare system to replace old one, with new
tyan motherboard, surprise came when i installed digium fxs/fxo and b410 p,
system unstable, random bips on start, misdn module Not loading, heh, old
system worked on asus p5nd2 sli without a problem with them. There were 2
reasons why i wanted change mobo 1: chipset on asus card generated too much
heat, 2: i had a new 3ware controller. So you never know really.

Loud scrathing sound? sometimes a card problem, try on other hardware.

Pci interrupts, also maybe sync problem (you can enable b410 clock in
misdn-init.conf)


Also turn off all sound/usb/etc unused devices.


2008/3/8, Royce Souther <[EMAIL PROTECTED]>:
>
> [EMAIL PROTECTED] lspci -v -s 01:07.0
> pcilib: Cannot open /sys/bus/pci/devices
> :01:07.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
> Modem/ISDN interface
> Subsystem: Unknown device b1d9:0003
> Flags: bus master, medium devsel, latency 32, IRQ 10
> I/O ports at ac00 [size=256]
> Memory at fbfff000 (32-bit, non-prefetchable) [size=4K]
> Capabilities: [40] Power Management version 2
>
> This is all I know about it. The client bought them about a month ago and
> installed them himself then asked me to setup the Asterisk program for him.
> The problem motherboard was a four year old Gigabyte with a Promise IDE
> controller.
>
> The new motherboard that works well is an ASUS but I don't know anything
> else about it.
>
> On Sat, Mar 8, 2008 at 9:09 AM, Steve Totaro <
> [EMAIL PROTECTED]> wrote:
>
> > Which revision of the Digium TDM400?
> >
> > On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther <[EMAIL PROTECTED]>
> > wrote:
> > > IRQ's seem to have been the problem. Thanks Steve Totaro for that tip.
> > >
> > > The Digium cards were at the same IRQ as the IDE controller, I moved
> > the
> > > cards and hard drives to a different system and all is good now.
> > >
> > > Thanks.
> > >
> > >
> > >
> > > On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro
> > > <[EMAIL PROTECTED]> wrote:
> > > > Check for IRQ issues, move the card to a different slot.
> > > >
> > > > You could ask permission to record calls so maybe you can hear the
> > > > sound yourself.
> > > >
> > > > I would then go ahead and swap out cards.  I have had TDM400 with
> > bad
> > > > modules and also bad ports on the cards themselves, so it could a
> > > > hardware issue.
> > > >
> > > > This is what I suspect, especially if you did not put any surge
> > > > suppression on your telco lines.  Usually, at least in my
> > experience,
> > > > ticks or beeps indicate IRQ, hissing or loud static indicate
> > something
> > > > with/on the board is bad.  ALWAYS use surge suppression on your
> > lines!
> > > >
> > > > Thanks,
> > > > Steve Totaro
> > > >
> > > >
> > > >
> > > >
> > > > On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther <[EMAIL PROTECTED]>
> > wrote:
> > > > > I have setup a few Asterisk systems for customers using Digium
> > TDM400
> > > cards
> > > > > and Aastra phones. No problems with sound quality at all except at
> > this
> > > one
> > > > > site.
> > > > >
> > > > > Every time I try their system I don't hear any problems but they
> > tell me
> > > > > that it is really bad. They describe it a a loud scratching sound.
> > > > >
> > > > > Are there any tests that can be done to pinpoint the problem? Has
> > anyone
> > > > > seen this before? Are there known causes for this?
> > > > >
> > > > > --
> > > > > Open Source: To innovate then create
> > > > > Proprietary: To imitate then litigate
> > > >
> > > >
> > > >
> > > > > ___
> > > > >  -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com --
> > > > >
> > > > >  asterisk-users mailing list
> > > > >  To UNSUBSCRIBE or update options visit:
> > > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > >
> > > >
> > > > ___
> > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com--
> > > >
> > > > asterisk-users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > >
> > >
> > > --
> > > http://www.Radados.org
> > > ___
> > >  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > >
> > >  asterisk-users mailing list
> > >  To UNSUBSCRIBE or update options visit:
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> > >
> >
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
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Re: [asterisk-users] replace astdb with a cluster-capable sql database engine

[Bruce Reeves]

Vieri,

What values are you looking to move from astdb?

I have used realtime to store values for call features and other
functions in the dial plan. I'm curious what you are looking to do.

On Sat, Mar 8, 2008 at 12:01 PM, Vieri <[EMAIL PROTECTED]> wrote:
> I've been searching the Internet for information
>  regarding the replacement of astdb with a modern sql
>  engine.
>
>  There are several reasons one would like to do this.
>  First of all, external applications have a hard time
>  reading/writing to the now-old astdb format.
>  Also (and this is what interests me most), the sql
>  astdb could easily be clustered throughout several
>  servers (I'm looking for a master-master MySQL
>  2-server cluster solution).
>
>  Asterisk has brought up Realtime which is very
>  powerful but, correct me if I'm wrong, it still
>  requires astdb internally. In other words, if I call
>  Set(DB) in the dialplan then it will always be using
>  astdb regardless of realtime.
>
>  Some projects like Callweaver have forked from
>  Asterisk 1.2 and replaced astdb with sqlite.
>
>  I'm wondering if Asterisk has plans to allow the user
>  to choose the astdb backend: standard db1, sqlite,
>  MySQL (which I would use with nbcluster for my
>  clustering purposes), Postgresql with Slony-II,
>  PGcluster, etc.
>
>  Or is it already possible?
>
>  There has been some talk on this before:
>  http://lists.digium.com/pipermail/asterisk-dev/2004-December/007846.html
>
>  Also, the func_odbc feature seems to be very powerful:
>  http://www.asteriskpbx.org/func_odbc
>  but:
>  1) would there be potential issues with db handles on
>  a very busy asterisk system after a relatively long
>  run time?
>  2) would there be a way to "map" the odbc function(s)
>  to the DB functions (Set(DB), read and write, DBdel,
>  etc) so that rewriting the whole dialplan would not be
>  necessary? (that's the whole point of defining a
>  different astdb "backend")
>
>  If there are known
>  problems/issues/projects/alternatives then please let
>  me know.
>
>  Thanks
>
>
>
>
>   
> 
>  Be a better friend, newshound, and
>  know-it-all with Yahoo! Mobile.  Try it now.  
> http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
>
>
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-- 
*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com


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Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.

[Royce Souther]

[EMAIL PROTECTED] lspci -v -s 01:07.0
pcilib: Cannot open /sys/bus/pci/devices
:01:07.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
Subsystem: Unknown device b1d9:0003
Flags: bus master, medium devsel, latency 32, IRQ 10
I/O ports at ac00 [size=256]
Memory at fbfff000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2

This is all I know about it. The client bought them about a month ago and
installed them himself then asked me to setup the Asterisk program for him.
The problem motherboard was a four year old Gigabyte with a Promise IDE
controller.

The new motherboard that works well is an ASUS but I don't know anything
else about it.

On Sat, Mar 8, 2008 at 9:09 AM, Steve Totaro <[EMAIL PROTECTED]>
wrote:

> Which revision of the Digium TDM400?
>
> On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther <[EMAIL PROTECTED]> wrote:
> > IRQ's seem to have been the problem. Thanks Steve Totaro for that tip.
> >
> > The Digium cards were at the same IRQ as the IDE controller, I moved the
> > cards and hard drives to a different system and all is good now.
> >
> > Thanks.
> >
> >
> >
> > On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro
> > <[EMAIL PROTECTED]> wrote:
> > > Check for IRQ issues, move the card to a different slot.
> > >
> > > You could ask permission to record calls so maybe you can hear the
> > > sound yourself.
> > >
> > > I would then go ahead and swap out cards.  I have had TDM400 with bad
> > > modules and also bad ports on the cards themselves, so it could a
> > > hardware issue.
> > >
> > > This is what I suspect, especially if you did not put any surge
> > > suppression on your telco lines.  Usually, at least in my experience,
> > > ticks or beeps indicate IRQ, hissing or loud static indicate something
> > > with/on the board is bad.  ALWAYS use surge suppression on your lines!
> > >
> > > Thanks,
> > > Steve Totaro
> > >
> > >
> > >
> > >
> > > On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther <[EMAIL PROTECTED]>
> wrote:
> > > > I have setup a few Asterisk systems for customers using Digium
> TDM400
> > cards
> > > > and Aastra phones. No problems with sound quality at all except at
> this
> > one
> > > > site.
> > > >
> > > > Every time I try their system I don't hear any problems but they
> tell me
> > > > that it is really bad. They describe it a a loud scratching sound.
> > > >
> > > > Are there any tests that can be done to pinpoint the problem? Has
> anyone
> > > > seen this before? Are there known causes for this?
> > > >
> > > > --
> > > > Open Source: To innovate then create
> > > > Proprietary: To imitate then litigate
> > >
> > >
> > >
> > > > ___
> > > >  -- Bandwidth and Colocation Provided by http://www.api-digital.com--
> > > >
> > > >  asterisk-users mailing list
> > > >  To UNSUBSCRIBE or update options visit:
> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > > ___
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> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> >
> > --
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[asterisk-users] PRI suppliers in Switzerland

[Chris Bagnall]

Greetings list,

I posted this to the -biz list a few days ago. In hindsight, I think it would 
have been more appropriate posted here, so apologies to those on both lists 
who've now seen this twice.

I have had a request to provide 2x PRIs to a site in Lausanne, Switzerland, but 
my knowledge of the Swiss Telco market is non-existent.

Are there any folks on the list who've experience in this market who might be 
able to offer some hints, or companies to approach?

Thanks in advance.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons





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Re: [asterisk-users] NIN Ghosts music (free download) safe for MOH?

[Jay R. Ashworth]

On Fri, Mar 07, 2008 at 09:12:52PM -0600, Tilghman Lesher wrote:
> > Pretty much, unless it's music developed in-house, I wouldn't put it on the
> > hold line unless you're willing to risk a fight with them (and even then,
> > they're likely to make a fuss just for the heck of it).
> 
> The default hold music is licensed from another publishing house, Free Play
> Music.  I know the default music tends to sound crappy, but that's entirely
> due to it being compressed down to single channel 8000Hz audio.  If you listen
> to it in the full, native format, it sounds pretty good.
> 
> Free Play Music will license 10 of their works for music on hold for $125 per
> year.  If you really want another style, I'd invite you to take a look at
> their website.  ALL of their works, in the entirety, are available online at
> their website, and not just a preview of each work, the whole thing.
> 
> http://www.freeplaymusic.com/
> 
> The three works distributed with Asterisk:
> http://www.freeplaymusic.com/search/download_file.php?id=303&dur=0&type=mp3
> http://www.freeplaymusic.com/search/download_file.php?id=22&dur=0&type=mp3
> http://www.freeplaymusic.com/search/download_file.php?id=313&dur=0&type=mp3

But, just to clarify, please remember that using music as MoH is
considered a "public performance", and if the pieces in question do not
include a buyout license *for the performance rights* (not just
synchronization rights for the recordings, as many "buyout production
libraries" do, then you *still* have to have ASCAP or BMI licensing to
use them.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)


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Re: [asterisk-users] Sync Problem (astribank)

[Grygoriy Dobrovolskyy]

I got that problem! It was bad contact in astribank usb port.
Have fun

Thank you tzafrir (angryuser)

2008/3/8, Tzafrir Cohen <[EMAIL PROTECTED]>:
>
> Hi
>
>
> On Fri, Mar 07, 2008 at 11:23:03PM +0100, Grygoriy Dobrovolskyy wrote:
> > My equipement
> > : 2x tdm 400p /4FXO/4FXS
>
>
> Any other Zaptel devices on that box? (you mentioned "isdn cards")
>
>
> > 16 PORT ASTRIBANK
> > My first TDM400p is the sync master, so i set astribank to sync to it,
> but
> > the quality is bad, like it is going fine and second after >>ROBOVOICE
> ;)
> > any other devices (isdn/sip/tdm cards) are working fine, looks like
> > astribank dont like to be sync 'slave'.
>
>
> What is the output of:
>
>   xpp_sync
>
> What version of Zapetl is it?
>
>
> >
> > Is there any way to change sync master manually, so every another
> hardware
> > sync to it ?
>
>
> Generally it is set with xpp_sync, which essentially writes a string to
> /proc/xpp/sync ('cat /proc/xpp/sync' to see the outpions).
>
>
> >
> > Also i know that tdm is my sync master from generated config. I dont
> > remember in what file it is written who is sync master, i you know it
> let me
> > know.
>
>
> --
>Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] I am now on ___________

[Tzafrir Cohen]

On Sat, Mar 08, 2008 at 06:36:39PM +, Karthik Arumugam wrote:
> Hi!
> I would like to invite you to visit my 

[snipped]

Yet another one sending a private invitiation to our mailing list?

We're all your friends :-)

Anyway, note the double From: field. The mailer responsible for that
horror:

  X-Mailer: PHPMailer [version 1.73]

-- 
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icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
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[asterisk-users] replace astdb with a cluster-capable sql database engine

[Vieri]

I've been searching the Internet for information
regarding the replacement of astdb with a modern sql
engine.

There are several reasons one would like to do this.
First of all, external applications have a hard time
reading/writing to the now-old astdb format.
Also (and this is what interests me most), the sql
astdb could easily be clustered throughout several
servers (I'm looking for a master-master MySQL
2-server cluster solution).

Asterisk has brought up Realtime which is very
powerful but, correct me if I'm wrong, it still
requires astdb internally. In other words, if I call
Set(DB) in the dialplan then it will always be using
astdb regardless of realtime.

Some projects like Callweaver have forked from
Asterisk 1.2 and replaced astdb with sqlite.

I'm wondering if Asterisk has plans to allow the user
to choose the astdb backend: standard db1, sqlite,
MySQL (which I would use with nbcluster for my
clustering purposes), Postgresql with Slony-II,
PGcluster, etc.

Or is it already possible?

There has been some talk on this before:
http://lists.digium.com/pipermail/asterisk-dev/2004-December/007846.html

Also, the func_odbc feature seems to be very powerful:
http://www.asteriskpbx.org/func_odbc
but:
1) would there be potential issues with db handles on
a very busy asterisk system after a relatively long
run time?
2) would there be a way to "map" the odbc function(s)
to the DB functions (Set(DB), read and write, DBdel,
etc) so that rewriting the whole dialplan would not be
necessary? (that's the whole point of defining a
different astdb "backend")

If there are known
problems/issues/projects/alternatives then please let
me know.

Thanks




  

Be a better friend, newshound, and 
know-it-all with Yahoo! Mobile.  Try it now.  
http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ 


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[asterisk-users] I am now on Refriendz!

[Karthik Arumugam]

Hi!
I would like to invite you to visit my Refriendz page and see my latest photos.

In order to visit my space, you must go to:
http://www.refriendz.com/?do=Login.Invite&rid=rtskarthik&[EMAIL PROTECTED]

(If this link does not work, please copy and paste it into your browser or go 
to www.refriendz.com and enter 'rtskarthik' as Invitation ID to Login to the 
web site.)

P.S. Refriendz is Invitation-Only, so do not miss your chance to visit my page!

Cheers!

Karthik






~~~
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Please do not reply directly to this email.  This mailbox is not monitored and 
you will not receive a response.

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Re: [asterisk-users] FXS channel banks

[Steve Totaro]

On Sat, Mar 8, 2008 at 11:35 AM, Dumpolid Exeplish <[EMAIL PROTECTED]> wrote:
> Why not get a TDMoE multiplexer, check out http://spidermux.com/

Maybe I am missing the concept, but why would you get a TDMoE
multiplexer for the OP's usage?

I can't really think under what circumstances this would be valuable.

Thanks,
Steve Totaro

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Re: [asterisk-users] FXS channel banks

[Dumpolid Exeplish]

Why not get a TDMoE multiplexer, check out http://spidermux.com/



On Sat, Mar 8, 2008 at 4:03 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Sat, Mar 08, 2008 at 07:41:50PM +0500, Faraz Khan wrote:
>  >
>  > > Just think of a different alternative: If you consider the cost of a
>  > > 24/32/48 FXS channel bank, vs. the cost of the PC used to driver it:
>  > > just put a simple PC at that end of the campus and attach the Astribank
>  > > to it.
>  > >
>  >
>  > A simple PC? Thats just asking for trouble. In my experience if this
>  > is an enterprise, you would need atleast a HP ML 150 with redundant
>  > PSU and Raid 5  + 1 spare disk, redundant fans EVERYWHERE! :)
>
>  For such a satelite server?
>
>  Get a small system with no moving parts. A bit more reliable (and less
>  noisy :-) ) than such a server. Do use a reliable system as your main
>  server.
>
>
>  --
>Tzafrir Cohen
>  icq#16849755  jabber:[EMAIL PROTECTED]
>  +972-50-7952406   mailto:[EMAIL PROTECTED]
>  http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.

[Steve Totaro]

Which revision of the Digium TDM400?

On Sat, Mar 8, 2008 at 10:24 AM, Royce Souther <[EMAIL PROTECTED]> wrote:
> IRQ's seem to have been the problem. Thanks Steve Totaro for that tip.
>
> The Digium cards were at the same IRQ as the IDE controller, I moved the
> cards and hard drives to a different system and all is good now.
>
> Thanks.
>
>
>
> On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro
> <[EMAIL PROTECTED]> wrote:
> > Check for IRQ issues, move the card to a different slot.
> >
> > You could ask permission to record calls so maybe you can hear the
> > sound yourself.
> >
> > I would then go ahead and swap out cards.  I have had TDM400 with bad
> > modules and also bad ports on the cards themselves, so it could a
> > hardware issue.
> >
> > This is what I suspect, especially if you did not put any surge
> > suppression on your telco lines.  Usually, at least in my experience,
> > ticks or beeps indicate IRQ, hissing or loud static indicate something
> > with/on the board is bad.  ALWAYS use surge suppression on your lines!
> >
> > Thanks,
> > Steve Totaro
> >
> >
> >
> >
> > On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther <[EMAIL PROTECTED]> wrote:
> > > I have setup a few Asterisk systems for customers using Digium TDM400
> cards
> > > and Aastra phones. No problems with sound quality at all except at this
> one
> > > site.
> > >
> > > Every time I try their system I don't hear any problems but they tell me
> > > that it is really bad. They describe it a a loud scratching sound.
> > >
> > > Are there any tests that can be done to pinpoint the problem? Has anyone
> > > seen this before? Are there known causes for this?
> > >
> > > --
> > > Open Source: To innovate then create
> > > Proprietary: To imitate then litigate
> >
> >
> >
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>
>
>
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Re: [asterisk-users] Background: reading the digits correctly, buffering it, waiting the sound message to complete

[Noah Miller]

Hi Bilal -

>  1) If I pressed 1 twice (11), so it runs the step
>  related to first 1 and then it runs the step related
>  to second 1, so it does buffering for my input and run
>  two steps, how can I make it run only the step related
>  to first entered digit "1" and does not do buffering
>  (so ignoring the second input)?

I don't think asterisk is doing any buffering here.  It's just
reacting quickly enough that it sends the first '1' to the first IVR
menu and then advances to the second IVR menu and sends the next '1'
there.


>  2) If I waited the sound message to completed and then
>  I entered my digits, it reads it correctly without
>  duplication in the DTMF and without any problem, but
>  if i entered my digits from the beginning of the sound
>  message (without waiting to complete it), then
>  asterisk might read the digits duplicated (specially
>  first entered digit), how can I resolve it? Does
>  playback resolve my problem? Any solution for
>  Background to avoid this behaviour?

I can think of two things:

1. Use different extensions in the second IVR menu so 1 is not a valid response
2. As you suggested, insert a brief Playback() into the second menu
that says "Please listen to the following options"


>  Note: I set the relaxdtmf=yes and I made the
>  toneduration=500.

If your DTMF is working correctly, I would not try to make any changes
there to influence menu selections.


- Noah

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Re: [asterisk-users] Customer complains of noise on line I cannot reproduce.

[Royce Souther]

IRQ's seem to have been the problem. Thanks Steve Totaro for that tip.

The Digium cards were at the same IRQ as the IDE controller, I moved the
cards and hard drives to a different system and all is good now.

Thanks.

On Wed, Feb 27, 2008 at 12:11 PM, Steve Totaro <
[EMAIL PROTECTED]> wrote:

> Check for IRQ issues, move the card to a different slot.
>
> You could ask permission to record calls so maybe you can hear the
> sound yourself.
>
> I would then go ahead and swap out cards.  I have had TDM400 with bad
> modules and also bad ports on the cards themselves, so it could a
> hardware issue.
>
> This is what I suspect, especially if you did not put any surge
> suppression on your telco lines.  Usually, at least in my experience,
> ticks or beeps indicate IRQ, hissing or loud static indicate something
> with/on the board is bad.  ALWAYS use surge suppression on your lines!
>
> Thanks,
> Steve Totaro
>
> On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther <[EMAIL PROTECTED]> wrote:
> > I have setup a few Asterisk systems for customers using Digium TDM400
> cards
> > and Aastra phones. No problems with sound quality at all except at this
> one
> > site.
> >
> > Every time I try their system I don't hear any problems but they tell me
> > that it is really bad. They describe it a a loud scratching sound.
> >
> > Are there any tests that can be done to pinpoint the problem? Has anyone
> > seen this before? Are there known causes for this?
> >
> > --
> > Open Source: To innovate then create
> > Proprietary: To imitate then litigate
> > ___
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> >
>
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Re: [asterisk-users] FXS channel banks

[Tzafrir Cohen]

On Sat, Mar 08, 2008 at 07:41:50PM +0500, Faraz Khan wrote:
> 
> > Just think of a different alternative: If you consider the cost of a
> > 24/32/48 FXS channel bank, vs. the cost of the PC used to driver it:
> > just put a simple PC at that end of the campus and attach the Astribank
> > to it.
> >
> 
> A simple PC? Thats just asking for trouble. In my experience if this  
> is an enterprise, you would need atleast a HP ML 150 with redundant  
> PSU and Raid 5  + 1 spare disk, redundant fans EVERYWHERE! :)

For such a satelite server?

Get a small system with no moving parts. A bit more reliable (and less
noisy :-) ) than such a server. Do use a reliable system as your main
server.

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

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Re: [asterisk-users] FXS channel banks

[Faraz Khan]

> Just think of a different alternative: If you consider the cost of a
> 24/32/48 FXS channel bank, vs. the cost of the PC used to driver it:
> just put a simple PC at that end of the campus and attach the Astribank
> to it.
>

A simple PC? Thats just asking for trouble. In my experience if this  
is an enterprise, you would need atleast a HP ML 150 with redundant  
PSU and Raid 5  + 1 spare disk, redundant fans EVERYWHERE! :)

Dont get me wrong. I think the idea of having a USB channel bank is  
great- but deployment in distributed or a 'campus' network would be  
very problematic compared to a quintum/grandstream SIP gateway  
(completely solid state / no pc required/ consumes little power)


-- 
Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
www.emergen.biz


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Re: [asterisk-users] Fwd: {s} - extension

[Tilghman Lesher]

On Saturday 08 March 2008 02:52:39 Daniel Suleyman wrote:
> Here is the log, my extensions is in the default section

> INVITE sip:[EMAIL PROTECTED] SIP/2.0

> [ Context 'default' created by 'pbx_config' ]
>   '7007' => 1. Goto(7007-${CNT}|1)   
> [pbx_config] '7007-2' =>   1. Set(GLOBAL(CNT)=1)   
>  [pbx_config] 2. Answer()   [pbx_config] 3.
> Playback(hello-world|skip) [pbx_config] 4. Hangup()
>   [pbx_config] '7008' => 1.
> Set(GLOBAL(connid)=0)  [pbx_config] 2.
> Set(GLOBAL(resultid)=0)[pbx_config] 3.
> Set(GLOBAL(fetchid)=0) [pbx_config] 4. MYSQL(Connect
> connid localhost root test test) [pbx_config]
> 5. MYSQL(Query resultid ${connid} Select a from a)
> [pbx_config]
> 6. MYSQL(Fetch fetchid ${resultid} a)
> [pbx_config] 7. MYSQL(Clear ${resultid})   [pbx_config] 8.
> MYSQL(Disconnect ${connid})[pbx_config] 9. goto(${a}|1)
>   [pbx_config] 's' =>1. Answer()   
>[pbx_config] '_7007-.' =>  1.
> Set(GLOBAL(CNT)=$[${CNT}+1])   [pbx_config] 2. Answer()
>   [pbx_config] 3. Playback(tt-weasels|skip)
>  [pbx_config] 4. Hangup()  
> [pbx_config]


I don't see extension "999" anywhere in your default context.  "s" is ONLY
used when an extension is NOT sent, not as a default when nothing matches.

-- 
Tilghman

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Re: [asterisk-users] FXS channel banks

[Raúl Gómez C.]

On Sun, Mar 9, 2008 at 9:24 AM, Faraz Khan <[EMAIL PROTECTED]> wrote:

> Steve- I thought Quintum doesnt do 48 Port FXS gateways? Last I found
> out from quintum was that that max is 48 port FXO or 24 Port FXS. Is
> this correct?


Yep, just up yo 24 FXS and up to 48 FXO...

http://www.quintum.com/enterprise/entspecs.html?id=21


-- 
Raul
Linux Counter #156439
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[asterisk-users] Experiences with grandstream GXW 4024 FXS?

[Faraz Khan]

Dear all,

Just wanted to know if any one had deployed the Grandstream GXW 4024  
yet. Wanted to hear any feedback and/or problems with this unit that  
you may have experienced.

Thank you.



-- 
Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
www.emergen.biz


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Re: [asterisk-users] FXS channel banks

[Faraz Khan]

I would however be interested in knowing how these USB channel banks  
work out in a extremely large environment. Cost/Reliability and  
management wise.Keep in mind that grandstream now has a 24 port FXS  
gateway which retails for $700- and their newer 8 port gateways are  
extremely good.


Quoting Steve Totaro <[EMAIL PROTECTED]>:

> On Fri, Mar 7, 2008 at 8:02 PM, Tzafrir Cohen  
> <[EMAIL PROTECTED]> wrote:
>> On Fri, Mar 07, 2008 at 03:00:03PM -0500, Jay R. Ashworth wrote:
>>  > On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote:
>>  > > > 
>> http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html
>>  > > >
>>  > > > Trouble is, you'll need 7 32-port units to cover your needs  
>> and I'm not
>>  > > > sure if USB2 is up to driving that many ... Tzafrir?
>>  > >
>>  > > One USB connector can take a number close to that easily. But even if
>>  > > USB were the bottleneck, you would just add another USB controller in
>>  > > the form of PCI card and get extra bandwidth.
>>  >
>>  > Is there any reason you'd want to do that on a system of that scale
>>  > instead of just using Ethernetted FXS boxes on a dedicated 100Base?
>>  >
>>  > Even if you didn't want to use reinvite, seems you'd still win just
>>  > from the less expensive host interface (I can't understand people using
>>  > T-1 interfaces for FXS channels either, honestly, in the current
>>  > environment).
>>
>>  USB is very cheap. It's in every computer. A dedicated ethernet segment
>>  costs more to set up that an extra USB segment (a 10$ for an extra USB
>>  controller? 20$ for a USB hub? a bit more for the wiring?).
>>
>>  TDMoE is more complicated as the latency is higher and the jitter is
>>  larger.
>>
>>
>>  Now both thing have been (T1 channel banks, and TDMoE) have been done by
>>  others. People do use and buy them. I don't intend to say that they
>>  don't. But ours does as well :-)
>>
>>
>>  --
>>Tzafrir Cohen
>>  icq#16849755  jabber:[EMAIL PROTECTED]
>>  +972-50-7952406   mailto:[EMAIL PROTECTED]
>>  http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>>
>
> Ethernet/SIP is going to be by far the most flexible.
>
> You can have much longer cable runs without some kind of USB repeater
> device.  Switches are cheap, CAT5/6 is cheap.
>
> You could put a Quintum Tenor AX 48 Port (for instance) in one section
> of a building, campus, LAN (WAN if you are daring) and the server
> could be anywhere, not tied by 15 or 30 foot USB cables.  Then if you
> are doing new wiring, you can run the shortest distance from the
> location of the SIP FXS device to the phones.
>
> You can have redundant, self healing links as well as link aggregation.
>
> I cannot see how TDMoE or USB come anywhere close to this flexibility
> and certainly don't see it being a fit for high port densities like
> discussed.
>
> I see TDM0E as something that a tech guy thought would be cool (and it
> is but not very practical) and a USB device something suited for the
> SoHo (but missing the scalability, redundancy, and flexibility that IP
> gives.)
>
> Thanks,
> Steve Totaro
>
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>



-- 
Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
www.emergen.biz

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Re: [asterisk-users] FXS channel banks

[Faraz Khan]

To support the quintum viewpoint we have deployed the Tenor AX 24-Port  
FXS in mass configurations (200-300 extensions) without issues. In a  
newer project we are going to do 1000 FXS extensions. They are  
exceptionally reliable.

Steve- I thought Quintum doesnt do 48 Port FXS gateways? Last I found  
out from quintum was that that max is 48 port FXO or 24 Port FXS. Is  
this correct?



Quoting Steve Totaro <[EMAIL PROTECTED]>:

> On Fri, Mar 7, 2008 at 8:02 PM, Tzafrir Cohen  
> <[EMAIL PROTECTED]> wrote:
>> On Fri, Mar 07, 2008 at 03:00:03PM -0500, Jay R. Ashworth wrote:
>>  > On Fri, Mar 07, 2008 at 02:14:57AM +0200, Tzafrir Cohen wrote:
>>  > > > 
>> http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html
>>  > > >
>>  > > > Trouble is, you'll need 7 32-port units to cover your needs  
>> and I'm not
>>  > > > sure if USB2 is up to driving that many ... Tzafrir?
>>  > >
>>  > > One USB connector can take a number close to that easily. But even if
>>  > > USB were the bottleneck, you would just add another USB controller in
>>  > > the form of PCI card and get extra bandwidth.
>>  >
>>  > Is there any reason you'd want to do that on a system of that scale
>>  > instead of just using Ethernetted FXS boxes on a dedicated 100Base?
>>  >
>>  > Even if you didn't want to use reinvite, seems you'd still win just
>>  > from the less expensive host interface (I can't understand people using
>>  > T-1 interfaces for FXS channels either, honestly, in the current
>>  > environment).
>>
>>  USB is very cheap. It's in every computer. A dedicated ethernet segment
>>  costs more to set up that an extra USB segment (a 10$ for an extra USB
>>  controller? 20$ for a USB hub? a bit more for the wiring?).
>>
>>  TDMoE is more complicated as the latency is higher and the jitter is
>>  larger.
>>
>>
>>  Now both thing have been (T1 channel banks, and TDMoE) have been done by
>>  others. People do use and buy them. I don't intend to say that they
>>  don't. But ours does as well :-)
>>
>>
>>  --
>>Tzafrir Cohen
>>  icq#16849755  jabber:[EMAIL PROTECTED]
>>  +972-50-7952406   mailto:[EMAIL PROTECTED]
>>  http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>>
>
> Ethernet/SIP is going to be by far the most flexible.
>
> You can have much longer cable runs without some kind of USB repeater
> device.  Switches are cheap, CAT5/6 is cheap.
>
> You could put a Quintum Tenor AX 48 Port (for instance) in one section
> of a building, campus, LAN (WAN if you are daring) and the server
> could be anywhere, not tied by 15 or 30 foot USB cables.  Then if you
> are doing new wiring, you can run the shortest distance from the
> location of the SIP FXS device to the phones.
>
> You can have redundant, self healing links as well as link aggregation.
>
> I cannot see how TDMoE or USB come anywhere close to this flexibility
> and certainly don't see it being a fit for high port densities like
> discussed.
>
> I see TDM0E as something that a tech guy thought would be cool (and it
> is but not very practical) and a USB device something suited for the
> SoHo (but missing the scalability, redundancy, and flexibility that IP
> gives.)
>
> Thanks,
> Steve Totaro
>
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>



-- 
Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
www.emergen.biz

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Re: [asterisk-users] NIN Ghosts music (free download) safe for MOH?

[Tzafrir Cohen]

On Fri, Mar 07, 2008 at 09:12:52PM -0600, Tilghman Lesher wrote:
> On Wednesday 05 March 2008 12:05:40 Joshua Kinard wrote:
> > That'd be ASCAP (I think there's another one too).  They're the ones known
> > for calling up places, asking to be put on hold to listen to the hold
> > music, then querying on whether it's been licensed or not (among other
> > tactics).
> 
> BMI (Broadcast Music International) and ASCAP (American Society of Composers,
> Authors, and Publishers) are the two major licensing houses in this country.
> There are others, but these are the two 800-lb gorillas in the industry.
> 
> > Pretty much, unless it's music developed in-house, I wouldn't put it on the
> > hold line unless you're willing to risk a fight with them (and even then,
> > they're likely to make a fuss just for the heck of it).

Just to clarify one thing here - if you got a sound file with a license
that is liberal enough, you *can* use it for on-hold music. This is
stated clearly in the license.

First-off, I'm not a lawyer. I'm just saying here things that make sense
to me. Consult a lawyer of your own in case of doubt.

It seems that on-hold music is considered as a sort of public
performance. So if I can use some music in my public, commercial, shows.
If the license permits me to do "anything execept", and no exceptions
for public performances or alike, then the lices permits me to use it.

Such a license can be GPL or BSD or MIT. For various reasons
those licenses are not so often used for music. But if you want to,
e.g., use music from the game The Battle of Wesnoth as your music, feel
free:

http://packages.debian.org/sid/wesnoth-music
http://packages.debian.org/changelogs/pool/main/w/wesnoth/wesnoth_1.4-1/wesnoth-music.copyright


Creative Commons did a good job at simplifying the licenses. The only
confusion is to consider all the CC licenses as one. There are a number
of CC licenses and they clearly differ by the name. If the license if
"nc" you cannot use the work for commercial uses.

Various Creative Commons licenses explicitly address the right to public
performance (explicitly grant it).

Something I can't see how to satisfy in a simple IVR system is the
requirement for attribution. If you call into an IVR, and playing the
work is considered a public performance, you still have no idea who's it
was originally.

One way to provide it is an "about this system" IVR menu item :-)

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

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Re: [asterisk-users] Fwd: {s} - extension

[Daniel Suleyman]

Here is the log, my extensions is in the default section

*CLI> core set verbose 3
Verbosity is at least 3
*CLI> [Mar  5 15:21:43] NOTICE[15870]: chan_sip.c:13879
handle_request_invite: Call from '7007' to extension '999' rejected
because extension not found.
*CLI> sip set debug
SIP Debugging enabled
*CLI>
<--- SIP read from 192.168.85.27:5060 --->
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
192.168.85.27:5060;branch=z9hG4bK-d87543-35763506b9430b2b-1--d87543-;rport
Max-Forwards: 70
Contact: 
To: "999"
From: "dan";tag=4773d83f
Call-ID: NjRjMmE1MDQzYjc5YjM3OTJiYmJkN2YzMGY2ZDMzNDQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 371

v=0
o=- 7 2 IN IP4 192.168.85.27
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.85.27
t=0 0
m=audio 5062 RTP/AVP 107 119 100 106 0 105 98 8 101
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<->
--- (12 headers 15 lines) ---
Sending to 192.168.85.27 : 5060 (NAT)
Using INVITE request as basis request -
NjRjMmE1MDQzYjc5YjM3OTJiYmJkN2YzMGY2ZDMzNDQ.
Found user '7007'
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.85.27:5062
Found unknown media description format BV32 for ID 107
Found unknown media description format BV32-FEC for ID 119
Found audio description format SPEEX for ID 100
Found unknown media description format SPEEX-FEC for ID 106
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x60c
(ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.85.27:5062
Looking for 999 in default (domain 192.168.85.29)

<--- Reliably Transmitting (no NAT) to 192.168.85.27:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.85.27:5060;branch=z9hG4bK-d87543-35763506b9430b2b-1--d87543-;received=192.168.85.27;rport=5060
From: "dan";tag=4773d83f
To: "999";tag=as2eff0a82
Call-ID: NjRjMmE1MDQzYjc5YjM3OTJiYmJkN2YzMGY2ZDMzNDQ.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<>
[Mar  5 15:22:17] NOTICE[15870]: chan_sip.c:13879
handle_request_invite: Call from '7007' to extension '999' rejected
because extension not found.
Scheduling destruction of SIP dialog
'NjRjMmE1MDQzYjc5YjM3OTJiYmJkN2YzMGY2ZDMzNDQ.' in 32000 ms (Method:
INVITE)

<--- SIP read from 192.168.85.27:5060 --->
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
192.168.85.27:5060;branch=z9hG4bK-d87543-35763506b9430b2b-1--d87543-;rport
To: "999";tag=as2eff0a82
From: "dan";tag=4773d83f
Call-ID: NjRjMmE1MDQzYjc5YjM3OTJiYmJkN2YzMGY2ZDMzNDQ.
CSeq: 1 ACK
Content-Length: 0


<->
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'NjRjMmE1MDQzYjc5YjM3OTJiYmJkN2YzMGY2ZDMzNDQ.' Method: ACK
sip sedialplan show
[ Context 'ael-default' created by 'pbx_ael' ]
  Include =>'ael-demo'[pbx_ael]

[ Context 'ael-demo' created by 'pbx_ael' ]
  '#' =>1. Playback(demo-thanks)  [pbx_ael]
2. Hangup()   [pbx_ael]
  '1000' => 1. Goto(ael-default|s|1)  [pbx_ael]
  '2' =>1. Background(demo-moreinfo)  [pbx_ael]
2. Goto(s|instructions)   [pbx_ael]
  '3' =>1. Set(LANGUAGE()=fr) [pbx_ael]
2. Goto(s|restart)[pbx_ael]
  '500' =>  1. Playback(demo-abouttotry)  [pbx_ael]
2. Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) [pbx_ael]
3. Playback(demo-nogo)[pbx_ael]
4. Goto(s|instructions)   [pbx_ael]
  '600' =>  1. Playback(demo-echotest)[pbx_ael]
2. Echo() [pbx_ael]
3. Playback(demo-echodone)[pbx_ael]
4. Goto(s|instructions)   [pbx_ael]
  '8500' => 1. VoicemailMain()[pbx_ael]
2. Goto(s|instructions)   [pbx_a

Re: [asterisk-users] FXS channel banks

[Tzafrir Cohen]

On Fri, Mar 07, 2008 at 08:28:14PM -0500, Steve Totaro wrote:
> 
> Ethernet/SIP is going to be by far the most flexible.
> 
> You can have much longer cable runs without some kind of USB repeater
> device.  Switches are cheap, CAT5/6 is cheap.
> 

> You could put a Quintum Tenor AX 48 Port (for instance) in one section
> of a building, campus, LAN (WAN if you are daring) and the server
> could be anywhere, not tied by 15 or 30 foot USB cables.  Then if you
> are doing new wiring, you can run the shortest distance from the
> location of the SIP FXS device to the phones.

Just think of a different alternative: If you consider the cost of a
24/32/48 FXS channel bank, vs. the cost of the PC used to driver it:
just put a simple PC at that end of the campus and attach the Astribank
to it.

> 
> You can have redundant, self healing links as well as link aggregation.
> 
> I cannot see how TDMoE or USB come anywhere close to this flexibility
> and certainly don't see it being a fit for high port densities like
> discussed.
> 
> I see TDM0E as something that a tech guy thought would be cool (and it
> is but not very practical) and a USB device something suited for the
> SoHo (but missing the scalability, redundancy, and flexibility that IP
> gives.)

As for USB: this is also what I thought before actually starting to work
with it. Sure, there are limitations. But the Linux USB stack is a nice
one.

As for TDMoE, I know that at least the current ztd-eth in Zaptel is 
considered "broken". Fixing it would be appreciated if you actually want 
to use it :-)

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

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Re: [asterisk-users] Fwd: {s} - extension

[Tzafrir Cohen]

On Sat, Mar 08, 2008 at 11:45:14AM +0400, Daniel Suleyman wrote:
> Even if I have s in defult it is not work.

So please provide a trace of that case:

core set verbose 3


And see what happens.

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

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