[Asterisk-Users] least sucky FXO interface?

[Dorn Hetzel]

Would anyone care to offer opinions as to the FXO interface which sucks
the least :)  I have an application in which it appears I must route
certain calls out an analog PSTN line.  Presently, I am testing an 
SPA-3000, but I can't seem to get the echo heard on the IP end of the
call down to a non-annoying level.

Any suggestions welcomed :)

-Dorn
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Re: [Asterisk-Users] least sucky FXO interface?

[Dorn Hetzel]

On Tue, Dec 14, 2004 at 06:44:04PM +, Jean-Michel Hiver wrote:
> Dorn Hetzel wrote:
> 
> >Would anyone care to offer opinions as to the FXO interface which sucks
> >the least :)
> >
> So far, for me, using VoIP -> PSTN termination provider has been the 
> solution which "sucked the least".
> My FXO card doesn't seem to work so well. Never tried my SIPURA as an 
> FXO device though... I use it as an FXS and for that usage it gives me 
> no echo.
>

Unfortunately, for this application, it appears that some calls
must be sent directly into an analog line, so I am trying to
find the hardware that will do that with the least crudifying
of the call :)

-Dorn
 
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Re: [Asterisk-Users] least sucky FXO interface?

[Dorn Hetzel]

On Tue, Dec 14, 2004 at 05:20:02PM +0100, Dave Cotton wrote:
> On Tue, 2004-12-14 at 18:44 +, Jean-Michel Hiver wrote:
> 
> > To the list: Am I right understanding that Fritz + BRI line = no echo 
> > issues?
> 
> I have two systems using AVM cards, one uses a C2 and the other 2 Fritz
> cards neither have any problems of echo.
>

Unfortunately, I cannot use a BRI for this part of my application,
only analog :(
 
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Re: [Asterisk-Users] VoIP Termination

[Dorn Hetzel]

This is the sort of language you are likely to find from anyone offering
"unlimited" plans.  It's just the reality of the fact that there is 
no such thing as unlimited, really...  Everything has it's limits :)

If you go with a per-minute plan, like from NuFone, Voipjet, etc., you
will not find any of this sort of hoo-hah, since they WANT you to use
more minutes, since that's how they get paid :)  

Any since the per minute rates can be as low as $0.013/minute last time
I looked, you have to use a LOT of minutes before you spend as much as
you would have with that "unlimited" plan... 

Regards,

-Dorn

p.s. I use both NuFone and VoipJet and am reasonably happy with both.



On Thu, Dec 16, 2004 at 01:53:44PM -0700, Mike Diehl (Encrypted email 
preferred) wrote:
> On Thursday 16 December 2004 05:17 am, Andrew Kohlsmith wrote:
> > On December 16, 2004 12:03 am, Mike Diehl (Encrypted email preferred) wrote:
> > > One of the catches is that I often telecommute and sometimes I do some
> > > side business; these practices violate many provider's acceptable use
> > > policies. So, I need a provider who doesn't care how I use the phone, and
> > > one that works well with Asterisk.
> >
> > You've gotta be kidding, VOIP providers are trying to regulate who you can
> > call?  Go with Nufone or iax.cc or even voicepulse connect -- use IAX2 over
> > SIP, IMO it's just better.
> 
> Thanx, I will look into these providers.
> 
> This is an exerpt from Packet8's Terms of Use statement.  I've edited it for 
> space, but I've tried to retain the context:
> --
> PERSONAL USE. 8x8's Service Plans for residential subscribers that offer 
> unlimited minutes of PSTN calls ("Unlimited PSTN Plans") are for the 
> reasonable personal residential use of End User only. End Users of Unlimited 
> PSTN Plans shall not use the Services for commercial or governmental purposes 
> or for profit or non-profit activities, including, but not limited to, home 
> office, business, sales, tele-commuting, autodialing, continuous or extensive 
> call forwarding, continuous connectivity, fax broadcast, fax blasting, 
> telemarketing or any other activity that would be inconsistent with personal 
> and residential usage. 8x8 reserves the right to immediately terminate or 
> modify the Services of any End User using Unlimited PSTN Plans if 8x8 
> determines, in its sole discretion, that End User is not using the Unlimited 
> PSTN Plans for End User's reasonable personal residential use.
> --
> 
> Now I agree with their policy on fax-blasting, etc.  But according to them, I 
> can't use my own phone for charity work?  I work at a national lab; would my 
> wife be alowed to call me at work?  Or would the be a "governmental purpose?"
> 
> It gets better... If Packet8 decides, in THEIR SOLE DISCRETION, that I'm 
> conducting a business with my phone, they can terminate my service, or 
> increase the price of it.
> 
> I'm trying to make an issue out of this because I think it needs to change 
> and 
> I'm hoping people who are affiliated with these providers are reading this.  
> I was going to go with Packet8.  I was going through the "final checklist" 
> before subscribing when I came accross this fascist policy.
> 
> Sure, I can go with a business plan, but that would cost me $39.95.  That's 
> $5 
> more than I'm spending for an analog phone line!  Part of the reason for me 
> to go with VoIP is to become "Quest Free."  But suddenly, Quest is starting 
> to resemble the Boy Scouts when compared to the types of usage policies I'm 
> seeing from some of the VoIP providers.
> 
> Sorry for the rant, but I hope you understand.
> 
> -- 
> Mike
> gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc
> 83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB
> 
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Re: [Asterisk-Users] VoIP Termination

[Dorn Hetzel]

On Thu, Dec 16, 2004 at 05:58:08PM -0500, Gary Carr wrote:
> So they offer termination via SIP for $0.013/minute?
>

Most of the good deals I have found are for IAX termination,
but maybe the same deal are available for SIP.  

-Dorn
 
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Re: [Asterisk-Users] VoIP Termination

[Dorn Hetzel]

I wouldn't say I hate SIP, it sucks less than H.323 and
so on by a large margin.  But, having said that, if you
can use IAX, it sucks even much than SIP does :)

On Thu, Dec 16, 2004 at 08:53:01PM -0500, Andrew Kohlsmith wrote:
> On December 16, 2004 08:47 pm, Gary Carr wrote:
> > Why is IAX termination better?
> 
> Becuase it's lean and mean and doesn't rely on the "too many cooks in the 
> kitchen" implementation that is SIP.  IAX2 works seamlessly behind NAT and 
> doesn't have any of the weird issues that SIP tends to have (STUN, SER, SIP 
> Proxies, etc., etc.).
> 
> I'm biased though, I just hate SIP.  :-)
> 
> -A.
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Re: [Asterisk-Users] VoIP Termination

[Dorn Hetzel]

On Fri, Dec 17, 2004 at 07:51:00AM +0100, Wilson Pickett wrote:
> > I'm looking to change from a standard telephone line to a VoIP phone line at
> > home.  I'm looking for recommendations for VoIP providers that I can use 
> > with
> > Asterisk.
> 
> Don't forget about emergency services (lack of) with voIP. 
>
For myself, I would NEVER give up my last lone 1FR (analog home line).
I route it through *, but at least one phone in the house is an old
fashioned analog phone that has direct access to the analog line for
when the poop is really hitting the fan ...  That's the phone all the
kids know to use for 911.  We just never use it for much of anything
else :)
  
> > One of the catches is that I often telecommute and sometimes I do some side
> > business; these practices violate many provider's acceptable use policies.
> > So, I need a provider who doesn't care how I use the phone, and one that
> > works well with Asterisk.
> 
> I agree with those who have said you should use metered services, but
> you need to measure the time you spend on the phone to see.
> Fortunately, asterisk will do that for you to the second, looking at
> the cdr records and totalling up the duration column for a specific
> period will tell you what your bill would be at the few cents a minute
> you'll be charged.
>
If you use so much time that at $0.013/minute you will spend more than
the $20.00 most unlimited plans cost, then no unlimited provider is
going to keep you as a customer for long, since they are almost 
certain to lose money on you.  Just accept the fact that if you
spend 2000+ minutes/month on outbound calls, your LD bill WILL
eventually exceed $20/month :)
 
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Re: [Asterisk-Users] My Boss wants background music!!!!

[Dorn Hetzel]

On Fri, Dec 17, 2004 at 07:54:19AM +0100, Wilson Pickett wrote:
> > I am searching for a new PBX for the company. My choice is Astrisk. My Boss
> > wants background music via all the telephones. This is done in a
> > conventional PBX that he wants, but I can use the Asterisk PBX if it can do
> 
> What a waste of resources though, like installing video games on the
> asterisk server... Ther must be a powerline intercom that would handle
> this (adding a speaker per music distribution point.)

Or get a couple of these -->  http://www.slimdevices.com/  or similar :)
and have REAL music in your office :)

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Re: [Asterisk-Users] VoIP Termination

[Dorn Hetzel]

On Sat, Dec 18, 2004 at 06:28:54PM +, Antony Stone wrote:
> On Saturday 18 December 2004 18:07, Dorn Hetzel wrote:
> 
> > I wouldn't say I hate SIP, it sucks less than H.323 and
> > so on by a large margin.  But, having said that, if you
> > can use IAX, it sucks even much than SIP does :)
> 
> Um, are you saying IAX is good, or that it is not good?   I'm not sure I 
> understand your statement above.
> 
> If you are saying that IAX is bad, why?  And what's better?
>

EEK!!!  Darn rented fingers :)

s/even much than/even much less than/ 

For my money, IAX is the best solution if you can use it,
or put another way, It sucks the least of all available
options :)

-Dorn
 
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[Asterisk-Users] IAX hardphone

[Dorn Hetzel]

Are there any IAX speaking "hardphones" out there?

If so, can anyone offer comment on their quality?

Thanks!

-Dorn

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Re: [Asterisk-Users] IAX hardphone

[Dorn Hetzel]

I can't get the link to work.  Does this mean that there is
some IP phone available which if loaded with the right
firmware can do IAX?  If so, where can I buy one and where
can I get the code?

-Dorn

On Wed, Dec 22, 2004 at 05:30:48PM +0100, Wilson Pickett wrote:
> This just in:
> 
> Centrality has released Version 1.4 for PA168x based phones.
>  
> Firmwarefiles for different protocols like SIP, IAX etc. can be
> downloaded from Centrality Website. Firmware for different brands is
> already available. I've tried out with my HOP-1002 which is actually a
> WuChuan with PA168S. It connects to my Asterisk and I succeeded to
> connect to German provider Purtel.com with IAX.
> 
> 
> Original source: 
> http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=HW-phone;action=display;num=1103573331
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[Asterisk-Users] TDM400P install on Debian 2.6.10

[Dorn Hetzel]

I just installed a new TDM400P with one FXO interface
in slot 4 (how it came from Digium).  This box is
running Debian with a 2.6.10-rc2-mm3 kernel.  After
the make linux26 and make install in /usr/local/src/zaptel,
I can see contents in /dev/zap but any attemp to 
touch for example /dev/zap/ctl gets a no such
device or address ...

Any suggestions?

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Re: [Asterisk-Users] TDM400P install on Debian 2.6.10

[Dorn Hetzel]

Not such a stupid question :)

The paper instructions included with the card failed
to mention that needed to be done manually :)

Up and running :)

-Dorn

On Wed, Dec 22, 2004 at 10:34:17PM -0800, Shahed wrote:
> 
> >I can see contents in /dev/zap but any attemp to 
> >touch for example /dev/zap/ctl gets a no such
> >device or address ...
> 
> May be a stupid suggestion, but have you loaded the
> wctdm driver by doing a "modprobe wctdm" ??
> 
> Regards
> Shahed
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Re: [Asterisk-Users] Why use 'Answer'?

[Dorn Hetzel]

On Thu, Dec 23, 2004 at 11:45:04AM +0100, Michael Vogel wrote:
> Rich Adamson schrieb:
> 
> >Rephrased: Why do folks think they have to use Answer in the sequence
> >when Playback (etc) is _not_ used?
> 
> Because they don't think or they love the telephone companies ...
>

Ok, this is probably a stupid question ;)

If I have a setup like the following:

One TDM400P with one FXO interface in slot 4.

zapata.conf chunk:

signalling=fxs_ks
language=en
context=in5100
channel => 4

extensions.conf chunk:

[in5100]
exten => s,1,Wait,2
exten => s,n,Answer
exten => s,n,DigitTimeout,5
exten => s,n,ResponseTimeout,10
exten => s,n(restart),BackGround(demo-congrats)
exten => s,n,WaitExten
exten => 401,1,Dial(SIP/sip1,20,tr)

///

I'm just playing demo-congrats so I can hear something when I call in
to know it's working.  What I would really like to have happen for
now is to ring the sip1 phone when the incoming line rings and only
answer the incoming line if the sip1 phone gets answered.

I've been playing with * for a while,
but this is my first zap device :)
[been all sip and iax til now]

Clues?

Regards,

-Dorn

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Re: [Asterisk-Users] Asterisk billing solution

[Dorn Hetzel]

On Thu, Dec 23, 2004 at 02:10:12PM +0100, E. Versaevel wrote:
> Qui, mais je ne parle pas français ;)
> 
> On Thu, 2004-12-23 at 00:12 +0100, Thierry wrote:
> > Hi
> > 
> > I have something like this but it's in french and it uses teh res_config
> > 
> > Best regards
> > Thierry wehr 
> 
> Thierry,
> 
> If you are willing to share your billing solution with the community,
> I'm sure there will be people pitching in to translate it from french to
> english and any other language they feel is important. My french is not
> that good but I will definitely have a look and help where possible.
> 
> Regards,
> Patrick
>
mon Francais est plus mauvais que pathetique.

But I'd still be willing to try and help also :)
[above remark compliments of machine translation :)]

-Dorn
 
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Re: [Asterisk-Users] Why use 'Answer'?

[Dorn Hetzel]

Thanks! :)

On Thu, Dec 23, 2004 at 08:57:22AM -0600, Rich Adamson wrote:
> 
> As in many cases with *, there are usually multiple ways to accomplish
> a task. Here's a couple that you'll need to tailor to your environment.
> 
>  [in5100]
>  exten => s,1,Dial(SIP/sip1,20,tr)
> 
> The above assumes the pstn line is _not_ sending any digits to you. If
> it does, then replace "s" with whatever they are sending to you.
>
If the PSTN line is just an analog line connected to the TDM400P FXO
interface, is there any way it will be sending me digits?
 
> Or, if you don't want to maintain multiple instances of the Dial command,
> then do something like:
> 
>  [in5100]
>  exten => s,1,Goto(local-extns,sip1,1)
> 
> The above assumes your sip phone dialplan reside in the [local-extns]
> context, that sip1 is a valid extn number within that context, and "1" 
> is the first priority in that sip phone definition. 
>
so I would need something like:
[local-extns]
exten => sip1,1,Dial(SIP/sip1,,tr)

??
 
> When someone answers the sip phone, the zap pstn line will be answered.
> If no one answers the sip phone, the "20" second timer will expire and
> ordinarily it would jump to the next priority. So, you probably want to
> remove the timeout altogether from that exten definition and just let 
> it ring forever.
>
If I want to timeout to voicemail then what would the
line(s) in [local-extns] look like?

-Dorn
 
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Re: [Asterisk-Users] TDM400 success?

[Dorn Hetzel]

On Thu, Dec 23, 2004 at 09:58:19AM -0700, Damon Estep wrote:
> Has anyone had success with the TDM400 in production? I have multiple
> boxes where these cards lock up and the only thing that will fix them is
> to unload *, modprobe -r wctdm, modprobe wctdm, load asterisk. Does not
> matter if it is a FXS/FXO module.
>
I have a recently installed TDM400P with one FXO in slot 4 which 
hasn't locked up yet, but it's only been a day or so at this
point and usage is light so far...

-Dorn
 
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[Asterisk-Users] Goto and exten => syntax

[Dorn Hetzel]

I understand some of the basic Goto() forms,
such as Goto(context,extension,priority) and
Goto(extension,priority) [within context I presume].

Can someone Explain Goto(6275|1) as found in the
sample extensions.conf?  Is this the same as
Goto(6275,1) just with a different delimiting
character?

In a use like:

exten => s,1,Wait,2
exten => s,n(dial),Dial(SIP/sip1,20,tr)

two questions:

does the "n" mean priority value of the previous line + 1 ?

does the (dial) mean create a variable? named dial which
can be referenced as priority in a Goto to get to this
line?


I looked at the Wiki but couldn't determine these points
for sure...

Regards,

-Dorn

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Re: [Asterisk-Users] Asterisk with Dialogic VFX/40ESC plus

[Dorn Hetzel]

On Thu, Dec 23, 2004 at 12:05:21PM -0600, Steven Critchfield wrote:
> 
> Close on the complete reason. There is also a licensing conflict with
> Dialogic drivers and GPL software. You have to get a commercial license
> for asterisk to clear the licensing issue. Beware that I think as soon
> as you get the commercial license for Dialogic, you may not be able to
> use any mysql functions due to license conflicts unless you bought a
> commercial mysql license as well.

Would it be correct that these licensing conflict only come into
play if the combination is further distributed.  It was my understanding
that nothing in the GPL prevents you from mixing GPL code and
proprietary code for your own private use;  you just can't
further distribute the combination since the GPL would require
you to distribute source to the other pieces and you don't 
have that to do so..?  Or maybe I'm all confused and should
be slapped :)

-Dorn

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[Asterisk-Users] changethread: can't change device with no technology!

[Dorn Hetzel]

After I leave a voicemail for an extension and hangup,
my asterisk console (with debug turned up quite high)
shows two error messages like:

WARNING[7664]: app_queue.c:341 changethread: Can't change device with no 
technology!
WARNING[7668]: app_queue.c:341 changethread: Can't change device with no 
technology!


Clues?

Thanks!

-Dorn

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Re: [Asterisk-Users] Asterisk with Dialogic VFX/40ESC plus

[Dorn Hetzel]

On Thu, Dec 23, 2004 at 01:26:38PM -0600, Steven Critchfield wrote:
> 
> I seem to remember that mysql made a stink recently about the dual
> licensing of asterisk. That was the cause of the mysql code getting
> pulled and placed in the add-ons sections so the core didn't have any
> licensing issues.

That's fair.  the MySQL folks do have the right to object to their
GPL code being distributed packaged with non-GPL copies of *.

They, just like Digium, offer non-GPL flavors of their code for
such purposes.  Of course, Digium, or anyone else, is free to
offer someone * and separately let them download MySQL and the
end user couple the two, and no licensing problem should
result...

-Dorn

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Re: [Asterisk-Users] Asterisk in parallel with PSTN

[Dorn Hetzel]

I've got a configuration with PSTN line connected to FXO
on TDM400P ringing through to a phone connected on a
Sipura SPA-3000.  The phone *does* ring before the
caller-id is available.  In fact, it shoes some 
alternate message like "waiting for caller id info"
right after the first ring and then changes to
the real caller-id after the 2nd ring.

-Dorn

On Thu, Dec 23, 2004 at 03:44:14PM -0500, Steve Prior wrote:
> Andrei (MPI) wrote:
> 
> >richard wrote:
> >>Asterisk also ring (so that all three of them are ringing), and then 
> >>someone can then choose which phone they want to answer?
> >
> >
> >Hi Richard,
> >
> >Absolutely, you can do that with Asterisk. Though VoiP telephone 
> >(Asterisk) may start ringing a second later than analog phones connected 
> >to the line directly. I do not think that would be a problem.
> 
> I believe you should note that if Asterisk is configured for callerid
> that the SIP phone won't ring until your non-Asterisk phones have rung
> twice.  This is because Asterisk gets the CID before it passes the call
> through.  I'd like to be wrong about this, but I believe that's what I've
> been told.
> 
> Steve
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Re: [Asterisk-Users] Asterisk in parallel with PSTN [OT]

[Dorn Hetzel]

On Thu, Dec 23, 2004 at 04:25:55PM -0600, Christopher L. Wade wrote:
> Dorn Hetzel wrote:
> >I've got a configuration with PSTN line connected to FXO
> >on TDM400P ringing through to a phone connected on a
> >Sipura SPA-3000.  The phone *does* ring before the
> >caller-id is available.  In fact, it shoes some 
> >alternate message like "waiting for caller id info"
> >right after the first ring and then changes to
> >the real caller-id after the 2nd ring.
> >
> >-Dorn
> 
> I've always wondered if certain IP (regardless of proto) phones could do 
> the same?  Basically initiate the call with fake callerid info and then 
> send an 'update' packet later to inform the phone of the new callerid? 
> Is this possible - even if it is only supported on certain phones?
> 
> If this is possible, then we could modify * to allow the dialplan to 
> (optionally) start before callerid is received and then update the 
> ${CALLERID} variable(s) once the information is available.  There are 
> situations where this is VERY desirable (obviously this only applies to 
> POTS though).
>
Seems like something similar must be going on in my setup,
because * is clearly taking the inbound call from the 
TDM400P/FXO and ringing it through to the Sipura FXS port
before the caller-id info is available.

-Dorn
 
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[Asterisk-Users] Voicemail email notification

[Dorn Hetzel]

Are there any common silent failure modes for email
notification from the Voicemail module.  I put the
email and pager email addresses in my entry in
voicemail.conf but no mail gets sent when I leave
a voicemail.  No obvious error messages either,
unless I'm just not looking in the right place.

Thanks for any clues :)

-Dorn

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Re: [Asterisk-Users] Voicemail email notification

[Dorn Hetzel]

On Thu, Dec 23, 2004 at 04:51:34PM -0600, Rich Adamson wrote:
> > Are there any common silent failure modes for email
> > notification from the Voicemail module.  I put the
> > email and pager email addresses in my entry in
> > voicemail.conf but no mail gets sent when I leave
> > a voicemail.  No obvious error messages either,
> > unless I'm just not looking in the right place.
> > 
> > Thanks for any clues :)
> 
> Nop, that's it other then you have to have sendmail configured
> and running on the system (or have a substitute mail handler).
>
sendmail is running (well, actually, it's postfix, but it
responds to /usr/sbin/sendmail) ...  still no mail gets
sent.  is there any way to get * to log what happens when
it tries to call sendmail?

-Dorn
 
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Re: [Asterisk-Users] What do I need to build up DID services?

[Dorn Hetzel]

On Sat, Dec 25, 2004 at 04:27:52PM +0800, Ronald Wiplinger wrote:
> Greg Hill wrote:
> 
> I am looking for a small device with four FXO and one WAN connection. 
> Simple, so that the cleaning woman can make a hardware reset if 
> necessary. This device should be connnected to my Asterisk box. The box 
> will be used in areas where DIDs are not available yet, and where you 
> not even can make ads for it ;-(
> It should be cheap, and it should connect either with SIP or IAX.
> No FXS is needed !!!
>
I'm not sure, but I think your best price/performance solution
might be a very small linux system (1U or smaller) with one
Digium TDM400P+4*FXO daughtercards.  At least the the size
point of 4*FXO, this may turn out to be your winner.  If it
boots from flash, your cleaning lady can always apply power
cycle for anything you can't fix via ssh...  

By using this solution, you would also get the ongoing 
bandwidth benefit of being able to trunk all 4 FXO calls
out in a single IAX frame.  This will save you money on
an on-going basis if your bandwidth costs are significant.

Perhaps Soekris would be a good box for this solution,
but there may be even smaller devices with the needed
1*PCI interface, I don't know.

Regards,

-Dorn
 
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Re: [Asterisk-Users] Linux Distribution

[Dorn Hetzel]

On Sat, Dec 25, 2004 at 12:29:21PM +, Jean-Michel Hiver wrote:
> Seth Ueland Chancy wrote:
> 
> Probably your best bet is debian + 2.4 kernel + X100P card + apt-get 
> install asterisk
> 
> Cheers,
> Jean-Michel.

I can also confirm that * works fine on Debian w/2.6.10-rc2-mm3
for the adventurous :)

-Dorn

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[Asterisk-Users] VM_CALLERID (how to get name+number)

[Dorn Hetzel]

I'd like to get VM_CALLERID to include number in addition to name
since often when calls come from cell lines or various other,
the name is just a city, state and the number would be more
usefull.  Is there a way to get the number in the VM_CALLERID
string, or is there a second variable I can use in formatting
email vmail notifications to get the number?

Regards,

-Dorn

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Re: [Asterisk-Users] What do I need to build up DID services?

[Dorn Hetzel]

On Sun, Dec 26, 2004 at 12:16:24PM -0600, James Taylor wrote:
> On Sat, 25 Dec 2004 10:43:00 -0500, Dorn Hetzel  
> <[EMAIL PROTECTED]> wrote:
> 
> >On Sat, Dec 25, 2004 at 04:27:52PM +0800, Ronald Wiplinger wrote:
> >>Greg Hill wrote:
> >>
> >>I am looking for a small device with four FXO and one WAN connection.
> >>Simple, so that the cleaning woman can make a hardware reset if
> >>necessary. This device should be connnected to my Asterisk box. The box
> >>will be used in areas where DIDs are not available yet, and where you
> >>not even can make ads for it ;-(
> >>It should be cheap, and it should connect either with SIP or IAX.
> >>No FXS is needed !!!
> >>
> >I'm not sure, but I think your best price/performance solution
> >might be a very small linux system (1U or smaller) with one
> >Digium TDM400P+4*FXO daughtercards.  At least the the size
> >point of 4*FXO, this may turn out to be your winner.  If it
> >boots from flash, your cleaning lady can always apply power
> >cycle for anything you can't fix via ssh...
> >
> >By using this solution, you would also get the ongoing
> >bandwidth benefit of being able to trunk all 4 FXO calls
> >out in a single IAX frame.  This will save you money on
> >an on-going basis if your bandwidth costs are significant.
> >
> >Perhaps Soekris would be a good box for this solution,
> >but there may be even smaller devices with the needed
> >1*PCI interface, I don't know.
> >
> >Regards,
> >
> >-Dorn
> 
> A MAX 4096 will work. And they're cheap.
> I'm using the TNT's.
> Contact me  off list.
>

That seems kind of overkill to service 4 pots lines ...

-Dorn
 
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[Asterisk-Users] caller id NUMBER in addition to or in place of NAME

[Dorn Hetzel]

On Sat, Dec 25, 2004 at 11:12:22PM -0500, Dorn Hetzel wrote:
> I'd like to get VM_CALLERID to include number in addition to name
> since often when calls come from cell lines or various other,
> the name is just a city, state and the number would be more
> usefull.  Is there a way to get the number in the VM_CALLERID
> string, or is there a second variable I can use in formatting
> email vmail notifications to get the number?

Maybe I could ask this more clearly.  Is there any way to get
the callers NUMBER in addition to or in place of NAME in the
voicemail notification emails?

Regards,

-Dorn

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Re: [Asterisk-Users] What do I need to build up DID services?

[Dorn Hetzel]

On Sun, Dec 26, 2004 at 02:22:43PM -0600, James Taylor wrote:
> On Sun, 26 Dec 2004 14:17:59 -0500, Dorn Hetzel  
> <[EMAIL PROTECTED]> wrote:
> 
> >On Sun, Dec 26, 2004 at 12:16:24PM -0600, James Taylor wrote:
> >>On Sat, 25 Dec 2004 10:43:00 -0500, Dorn Hetzel
> >><[EMAIL PROTECTED]> wrote:
> >>
> >>>On Sat, Dec 25, 2004 at 04:27:52PM +0800, Ronald Wiplinger wrote:
> >>>>Greg Hill wrote:
> >>>>
> >>>>I am looking for a small device with four FXO and one WAN connection.
> >>>>Simple, so that the cleaning woman can make a hardware reset if
> >>>>necessary. This device should be connnected to my Asterisk box. The  
> >>box
> >>>>will be used in areas where DIDs are not available yet, and where you
> >>>>not even can make ads for it ;-(
> >>>>It should be cheap, and it should connect either with SIP or IAX.
> >>>>No FXS is needed !!!
> >>>>
> >>>I'm not sure, but I think your best price/performance solution
> >>>might be a very small linux system (1U or smaller) with one
> >>>Digium TDM400P+4*FXO daughtercards.  At least the the size
> >>>point of 4*FXO, this may turn out to be your winner.  If it
> >>>boots from flash, your cleaning lady can always apply power
> >>>cycle for anything you can't fix via ssh...
> >>>
> >>>By using this solution, you would also get the ongoing
> >>>bandwidth benefit of being able to trunk all 4 FXO calls
> >>>out in a single IAX frame.  This will save you money on
> >>>an on-going basis if your bandwidth costs are significant.
> >>>
> >>>Perhaps Soekris would be a good box for this solution,
> >>>but there may be even smaller devices with the needed
> >>>1*PCI interface, I don't know.
> >>>
> >>>Regards,
> >>>
> >>>-Dorn
> >>
> >>A MAX 4096 will work. And they're cheap.
> >>I'm using the TNT's.
> >>Contact me  off list.
> >>
> >
> >That seems kind of overkill to service 4 pots lines ...
> >
> And, you can't do DID on pots lines.
> You can use BRI.
> or You can use 4 wire trunks.
> It's cheaper to do T1's and only use a few channels.
>

(a) there are definitely analog DID implementations out there.
not saying they're pretty, but they exist...

(b) are you really sure it's cheaper with only 4 channels to
do a T1?  including local loop? 
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Re: [Asterisk-Users] caller id NUMBER in addition to or in place of NAME

[Dorn Hetzel]

On Sun, Dec 26, 2004 at 02:40:20PM -0600, Eric Wieling aka ManxPower wrote:
> Dorn Hetzel wrote:
> >
> >Maybe I could ask this more clearly.  Is there any way to get
> >the callers NUMBER in addition to or in place of NAME in the
> >voicemail notification emails?
> 
> I seem to recall a bug regarding this.  Are you using 1.0.3, 1.0.x CVS 
> STABLE, or CVS-HEAD.  The problem is that what was listed in the 
> voicemail.conf.sample as the default e-mail message was, in fact, not 
> the default e-mail message.  Uncommenting out the example message fixed 
> the problem.
>
That helped!  Now the regular emails have the name and  ...

However, the pager emails still have name only...  Is there a string
variable to redefine the subject/body sent to the pageremail?

Regards,

-Dorn
 
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Re: [Asterisk-Users] Linux Distribution

[Dorn Hetzel]

On Tue, Dec 28, 2004 at 09:13:02AM -0800, Geoff Nordli wrote:
> [EMAIL PROTECTED] <> scribbled on :
> 
> > On Sat, Dec 25, 2004 at 12:29:21PM +, Jean-Michel Hiver wrote:
> >> Seth Ueland Chancy wrote:
> >> 
> >> Probably your best bet is Debian + 2.4 kernel + X100P card + apt-get
> >> install asterisk 
> >> 
> >> Cheers,
> >> Jean-Michel.
> > 
> > I can also confirm that * works fine on Debian w/2.6.10-rc2-mm3 for
> > the adventurous :) 
> > 
> > -Dorn
> > 
> 
> Jean-Michel or Dorn, did you have any issues compiling the zaptel-source
> with the 2.4.27-1 or 2.6.x kernels on Sarge?
>
Well, I think I'm acually running Woody boosted to 2.6.10-rc2-mm3, but
I had no problems at all with "make linux26" on zaptel after reading
README.Linux26 ...  Mostly just making sure make can find your kernel
source.

-Dorn
 
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Re: [Asterisk-Users] Compile Error

[Dorn Hetzel]

On Tue, Dec 28, 2004 at 10:02:12PM +0200, David Norton wrote:
> Hi,
>  
> I have been running asterisk for about a week though on a debian system
> through apt-get. I am now trying to compile it use the CVS and im getting
> this error.
>  
> /usr/bin/ld: cannot find -lssl
>  
> What do I need to install to get rid of this message?
>
I built openssl from source to deal with that requirement,
your mileage may vary :)

-Dorn
  
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Re: [Asterisk-Users] Hardware opinions?

[Dorn Hetzel]

On Tue, Dec 28, 2004 at 02:17:47PM -0600, Me wrote:
> Hello, I am trying to build up a pretty meaty Asterisk box after doing our 
> initial testing and playing on a 1ghz system.
> 
> Right now I have decided on a prebuilt system which I normally don't do but 
> thought it seemed like a good deal.
> 
> I have included the initial specs below, I will be adding another 1 GB of RAM 
> for a total of 2 GB. 
> 
> My first question is regarding the serial ATA drives... I will be using 
> Fedora and considering FC1 seems to be the smartest of the builds when it 
> comes to the digium hardware, will I have to scrap the SATA drives because 
> FC1 doesn't support them or do I have bad information?
> 
> If I need to scrap the SATA drives and let's say I didn't care about the Raid 
> functionality, would you folks think that IDE drives would be fine or would 
> the speed of SCSI really make much of a difference when it comes to Asterisk? 
> If speed of drives does matter, can someone tell me why Asterisk might need 
> fast drives vs. say 7200 IDE drives?
> 
> Next and last question is, how many simultaneous calls do you folks figure I 
> can run on this in the following two scenarios:
> 
> 1- All clients would be using SIP devices like SPA-2000's and all calls would 
> originate/terminate using an IAX termination partner.
> 2- All clients would be using IAX like Asterisk or an IAXy and all calls 
> would originate/terminate using an IAX termination partner.
> 
> 
> Here are the specs:
> 
>   a.. Dual (2) Intel Xeon 2.66GHz CPUs w/ HT Technology
>   b.. 533MHz Front Side Bus 
>   c.. 1GB PC2100 DDR ECC Registered Memory
>   d.. Two (2) 120GB SATA 7200RPM Hard Drives with 8MB Cache 
>   e.. 52X CD-RW Drive w/Burning Software 
>   f.. 3.5" 1.44MB Floppy Drive 
>   g.. ATI Rage XL with 8MB Onboard
>   h.. Onboard RAID controller 
>   i.. (2) Intel Ethernet Controllers (1x1000BT Gigabit & 1x10/100) 
>   j.. 2U Rackmount Chassis w/ 500-Watt Power Supply 
> 
> Thanks!
>

You don't need to scrap the SATA drives, they are very nice.
You might want to give Debian a try.
Don't use the "RAID" mode on the motherboard as it's likely
"fake raid", instead use linux md driver for software raid,
it's smoking fast for SATA drives and on my 3.2ghz box
barely scratches the CPU resyncing.

-Dorn
 
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Re: [Asterisk-Users] Hardware opinions?

[Dorn Hetzel]

On Tue, Dec 28, 2004 at 04:12:26PM -0600, Me wrote:
> Dorn,
> 
>  Can you give me some details on this linux md driver you mentioned?
> 
> Also, you say not to scrap the SATA drives, is this because you think I can
> use them with FC1 or because you think I should try Debian? I really don't
> want to venture away from Fedora at the moment for a few reasons.
>
It's likely you can make the SATA drives work with Fedora, I just
can't say from personal experience.  The md driver is a software
raid implementation.  check out mdadm (the setup command) man pages
for more info.  I'm using three different flavors on the last
server I built, raid0 for speed /tmp type space, raid5 for speed
and security, and a triple-copy raid1 for really important stuff.

What sort of chipset is your SATA controller interface?  Intel
ICH6R? 

-Dorn
 
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Re: [Asterisk-Users] OT: Linux routing with T100P problems

[Dorn Hetzel]

On Tue, Dec 28, 2004 at 11:18:42PM -0600, Matthew Boehm wrote:
> Hey gang,
>  I was successful in recompiling my 2.4.20 kernel to support HDLC. I was
> successful in hooking up our T1 line into the zap card. I was successful in
> being able to ping equipment on the other end of the T1. I was unsuccessful
> in pinging the outside world from the other end of the T1.
> 
> I've attached a cheezy image of the network. Here is the routing table:
> 
> [EMAIL PROTECTED] root]# route
> Kernel IP routing table
> Destination Gateway Genmask Flags Metric RefUse
> Iface
> 10.0.5.2   * 255.255.255.255 UH0 00
> hdlc0
> 10.0.0.0   *   255.255.255.0   U   0 0
> 0eth1
> 10.0.3.0   *   255.255.255.0   U   0 0
> 0eth1
> 65.78.109.0 *   255.255.255.0   U   0 00
> eth0
> 127.0.0.0 *   255.0.0.0   U   0 0
> 0lo
> default   65.78.109.2 0.0.0.0   UG0 0
> 0eth0
> 
> There are 2 NICs (10.0.3.10, 65.78.109.10) and 1 T100P (10.0.5.1) on this
> box.
> 
> Like I said above, from this machine I can ping everything in every attached
> network and the outside world. For some reason, I cannot ping the outside
> world if I am comming from the 10.0.0.* network on the diagram. From that
> network, I can ping 10.0.5.1 (this box) but nothing else.
> 
> I'm a little stumped. My iptables are completly empty. If this is waaayyy
> off topic, please contact me off list. But I figured since it was related to
> the T100P it might be relevant.
> 
> What can I use to find out why packets destined for the outside world (via
> 65.78.109.2) are not being routed?
>

Since 10.x.x.x is RFC1918 private space which no real-world addresses
will/can reply to, you need to use masquerading (NAT) so that all of 
the packets to the "outside world" appear to come from a public
routable address on the outside of your gateway box.

-Dorn
 
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Re: [Asterisk-Users] Hardware opinions?

[Dorn Hetzel]

On Tue, Dec 28, 2004 at 11:37:42PM -0600, Me wrote:
> > What sort of chipset is your SATA controller interface?  Intel
> > ICH6R?
> 
> Adaptec ICH5R SATA controller according to SuperMicro which makes the Mobo.
> The board has an Intel® E7501 main chipset.
>
That should probably work.  You may need to reconfigure your kernel,
or maybe not.  I can't say for Fedora.

run "make menuconfig", go into "Device Drivers", then into
"SCSI Device Support" (yes, that's where the good SATA stuff
hides), then into "SCSI low-level drivers" (at the bottom),
where you will find a section that starts with 
"Serial ATA (SATA) Support".

I am using "AHCI SATA Support", which is very nice, but
depends on your motherboard bios having an AHCI mode for
the SATA disks.  [if you can use this mode, I highly
recommend it, as the performance is shockingly good]

If you don't have AHCI, the "Intel PIIX/ICH support"
may work for you.  There are also drivers from various
other flavors of motherboard controllers, but I
haven't fooled with them.

If the support you need is already built in your kernel,
then you may not need to rebuild it.

I recommend building the relevant drivers hard into
the kernel (not loading them as modules) since you're
going to need them all the time anyway.  [I may be
clueless on that, but it works for me :) ]

Happy Holidays!

-Dorn
 
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[Asterisk-Users] trimming messages on reply

[Dorn Hetzel]

All,

Please consider trimming off the bottom of the message you are
replying to.  It usually take only a few seconds and saves
everyone reading the list from extra bloat in their mailbox :)

Happy Holidays!

-Dorn

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Re: [Asterisk-Users] final call for Departments

[Dorn Hetzel]

On Thu, Dec 30, 2004 at 11:16:01PM -0800, Alspach Family wrote:
> I don't want to sound like a TV evangelist from the 80's and 90's but if 
> you have it to give, please do.  We have operators standing by to accept 
> your donation. All you have to do is PayPal it to [EMAIL PROTECTED] 
>  (Note, this is not me.  Rob  is the guy 
> doing all the work.)

Just dropped in my $10.00, perhaps a few other folks can do likewise.

-Dorn

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Re: [Asterisk-Users] Qs about FXO/FXS cards

[Dorn Hetzel]

On Sat, Jan 01, 2005 at 07:23:58PM -0500, Jim Van Meggelen wrote:
>[...] 
> What if, for example, the TDM400 issues were a cumulative thing? If you
> had over 6dB of attenuation on the PSTN loop, coupled with greater than
> 5V potential on the neutral-ground of your elecrical receptacle,
> compounded by a cheap power supply, exascerbated by a Via-chipset, would
> you not be virtually guaranteed some strange behaviour? But if your PSTN
> was -3dB, your electrical feed derived from a power conditioner, your
> power supply manufactured by PC Power & Cooling, and a ServerWorks
> chipset-based MoBo, would your system always be faultless?
>
Can you recommend any favorite motherboards?
 
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[Asterisk-Users] reliable capacity for a single * box

[Dorn Hetzel]

I have a couple of capacity related questions for which I am hoping
to find answers (or at least hints) derived from real-world experience.

asterisk as a "trunking gateway";  bunch of sip phones in location
one need to access other non-* sip PBX device in location two over
constrained bandwidth.  I can't replace the existing SIP phones or
the other SIP device.  I'm considering using * boxes at each end of
the WAN circuit to pack all of the individual SIP sessions into an
IAX2 trunk and then unpack them back into SIP at the far end.

Assuming for the moment G.711 all the way thru, so no codec 
conversions anywhere, how many simultaneous sessions should
I reasonably expect to get through a typical single CPU P4 3.2ghz
box?

Then as an option, if I wanted to compress the SIP G.711 calls
into say IAX2-GSM trunks, how would that impact the channels per
box number?  (well, I know it's going to go down, any hints how
much?)

Any thoughts?

Regards,

-Dorn

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Re: [Asterisk-Users] Cisco Phones

[Dorn Hetzel]

On Mon, Jan 03, 2005 at 07:11:46PM -0500, Garrett Smith wrote:
> I wouldn't consider it an advertisement. There was no price, etc. I was
> simply telling those that ordered from me previously I have more. It is
> easier to send one email to 100's, then 100's of emails. If you do not want
> to save money, nor want the rest of your fellow members to have the
> opportunity to, I can stop. Fortunately, there is enough interest in these
> phones coming from the non-business discussion list to warrant a small email
> alerting everyone to a great deal. I believe great pricing from supplier has
> been brought to everyone's attention numerous times in the past...
> 
> Garrett
>

Well, here's at least one list reader who wishes you would keep the adverts 
over in the -biz list, whether they have prices or not.  It isn't polite to
make *everyone else* wade through the notes when you could have just posted
it to -biz and mailed directly to folks you already have done business with
in the past.

What you have done, in effect, is abuse our convenience to serve yours.  Oh,
and for those of who *do* subscribe to both lists, all the more annoying...

-Dorn
 
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Re: [Asterisk-Users] ISDN/SS7 book?

[Dorn Hetzel]

On Wed, Jan 05, 2005 at 03:56:39PM +0100, Roy Sigurd Karlsbakk wrote:
> hi
> 
> some time ago, I asked the list of a good book for learning ISDN and 
> SS7. I don't need to know how to write a channel driver or something; I 
> just want to know more about the possibilities and what's really sent 
> back and forth. I was told the book "ISDN and SS7: Architectures for 
> Digital Signaling Networks" by Uyless Black (ISBN 0132591936) was a 
> good choice, but this seems sold out. Does anyone know about another 
> book about the subject?
>
amazon.com links to plenty of used copies this book for under $15.00.

I just put ISDN and SS7 in the search box and it went straight to it.

-Dorn
 
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Re: [Asterisk-Users] What is acceptable network latency for voip connection?

[Dorn Hetzel]

On Sat, Jan 08, 2005 at 11:01:25PM +0800, David Liu wrote:
> Well there is nothing much you can do if you don't own all the routes.  But in
> concept you can, and this is purely just theoritical and a very unhealthy
> thing for the Internet, is to write a program running on your router that
> constantly streams traffic to your end point, this will maintain a constant
> bandwidth from your network to your far-end.  Then, your program should detect
> within a few ms that you are setting a call up and immediately reduce your
> bogus traffic and make room for your "Real" voice traffic.  Again this is
> super unhealthy for the Internet, but the idea is TDM on STDM - constantly
> occupying certain trunks (bandwidth) on the Internet.  So whenever you need
> it, you will have it.  
> 
> David

David,

This is an almost unimaginably bad idea, and what's worse, it won't even
do what you want.  No matter how much data you stream constantly to "hold
your place", there is just no such feature present in the Internet.  Any
or all of the packets you send are always subject to being dropped due to
congestion.  Whatever happened one hour or one minute or one second or 
even one packet before is, for the most part (excluding routers with
route cacheing going on, and all that does is make your 2nd and later
packets do ever so slightly better than your first packet), completely
irrelevant.  Each packet gets forwarded or dropped at each router based
pretty much on the conditions that router sees right then;  Does it have
space on the outbound queue for the chosen interface, etc.  

So all this scheme will do is cause more lost packets in a global sense.

-Dorn

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Re: [Asterisk-Users] Asterisk on a notebook?

[Dorn Hetzel]

On Thu, Jan 13, 2005 at 11:09:28AM -0500, Ken D'Ambrosio wrote:
> I'd dearly love to be able to give an Asterisk demo by just toting my 
> notebook, a PC/PCMCIA card, and a couple SIP phones.  Is there any way 
> to do this?  Or should I look for a small-profile box with PCI slots, 
> instead?

I've done this.  Not as a demo, but as a production box.  If the laptop
has built-in ethernet, you don't even need a PCMCIA card.  Just use
small boxes like Sipura SPA-3000 or 2000 for your FXO and FXS interfaces
and bring a small ethernet switch and it's all good.  I use one in
production for my "home" PBX because of the handy built-in UPS :)

Of course, you will want to be running Linux on the laptop, but then
what else would you want to be running anyway? :)

-Dorn

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[Asterisk-Users] 1xT1 PCI card for *

[Dorn Hetzel]

I have a nice used zhone channel bank I want to experiment with, but need 
a T1 interface for my * box to do so.  The TE410P looks nice, but more
money than I want to spend to experiment, and I don't need 4xT1, only
1xT1.

Are there any good 1xT1 PCI cards that are recommended?

Regards,

-Dorn

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Re: [Asterisk-Users] Re: SS7 and Asterisk solution

[Dorn Hetzel]

On Sun, Jan 16, 2005 at 01:33:54PM +0100, Roy Sigurd Karlsbakk wrote:
> 
> You can modify and/or link to GPLed code with commercial code and get 
> away with it as long as you don't distribute the stuff. That's the 
> story with G.729, with nVidia drivers etc etc etc
>
I suppose it's even possible to distribute your commercial code in source
form and ask your customer to acquire their own copy of * to link it with.
(is that actually true?)

-dorn
 
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[Asterisk-Users] asterisk and predictive dialers

[Dorn Hetzel]

Are there any predictive dialers for Asterisk which will
do answering machine detection, etc?

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[Asterisk-Users] spandsp

[dorn hetzel]

opencall.org seems to be off air since yesterday.  I am wondering if
anyone has a private cache of the most current spandsp they would be
willing to share...

regards,

-dorn
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Re: [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole

[dorn hetzel]

I have always had excellent service from Atacomm.  Not always the absolute lowest price, but they pretty much ship when they say they will.On 8/10/06, Don Tasker
 <[EMAIL PROTECTED]> wrote:Had the same issue here as well, our company was in a
dire need for some digium cards due to an internalsystem failure. So we ordered some replacements fromvoiplink for overnight delivery, we loved the pricingthey had on the website and even called to confirm
they had stock and could ship that very same day aswell as confirm the pricing.After placing the order, our card was chargedimmediately, and we thought all was well.After several failed attempts to contact them and
products not arriving the next day as promised, wewere finally able to make contact with a rep atvoiplink a few days later and found that productsstill had not YET shipped and that they were unable tofufil the order with pricing they had listed on their
website and confirmed verbally over the phone.We immediately demanded the order canceled. Since weurgently needed the products, we decided to tryfinding a more reliable vendor.They claimed to be able to ship out the products that
night so as not to lose our order, but only if we paidalmost DOUBLE the price we both agreed upon ANDaccepted payment for!!!We can only conclude that this was a "bait and switch"tactic meant to take advantage of our situation.
Either way, we wanted the order cancelled and chose togo elsewhere. We requested the amount refunded, buteven that took over a WEEK!!We ended up going with Atacomm off of a referral forthe products in hopes they could fufil our needs on
time. Being weary, I called them first and confirmedthey had them, which they told me they stocked intheir very own warehouse. We ordered the products andthe very the next day had the products in our hands
without any hassle.Additionaly, they even had tracking to me that veryday which ensured I was getting the products.Stay away from voiplink!-Don> From: Shaw Terwilliger <
[EMAIL PROTECTED]>> Date: August 10, 2006 8:28:51 AM CDT> To: Asterisk Users Mailing List - Non-CommercialDiscussion > Subject: Re: [asterisk-users] Warning - Voiplink.comdoesn't deliver - stuck in a hole> Reply-To: Asterisk Users Mailing List -Non-Commercial Discussion
>> Mr. Jones wrote:>> I have had the same experience with a Grandstreamorder from them - 7>> days and no product.
 They even told me it was shipping Monday, butcouldn't produce a>> tracking number on Tuesday.>> I ordered a T1 card from them on July 17.  No traceof it.  I've sent
> them four e-mails, the last two asking them tosimply cancel the order> and give me a refund.  The last e-mail"half-bounced" (I got bounces> from some of the addresses the original recipient
address must have> forwarded to).  I never got a single response fromthem, and they still> have my money, and I still have no card (or even atracking number).>> I'll initiate a charge-back via my credit card
company today--it's been> almost a month!>> Stay far away from voiplink.com.>> --> Shaw Terwilliger <
[EMAIL PROTECTED]>> SourceGear LLC>> ___> --Bandwidth and Colocation provided by Easynews.com--
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[Asterisk-Users] backup routing for IAX outbound

[dorn hetzel]

Is there a simple way in extensions (or elsewhere) to select among multiple outbound routes for a given call,
not so much based on cost as on current route availability.  I.e. dial out using provider X unless our connection
to them is down, or lagged, or jittery, or whatever way out of tolerance.  Perhaps skip provider X if they are out
of "qualify" bounds and use provider Y?

Any clues appreciated :)


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[Asterisk-Users] how to get separate CDR for inbound and outbound legs of a call

[dorn hetzel]

If I have a call coming in on one line and then I am using Dial to send it out to another outside line, is there a good way to get separate CDR for both call legs?
 
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