Re: [asterisk-users] ztdummy kills audio

[Chris Nestrud]

On Sun, 16 Sep 2007 01:29:11 +0300, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Sat, Sep 15, 2007 at 01:30:26PM +, Chris Nestrud wrote:
>> On Sat, 15 Sep 2007 16:03:07 +0300, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>> > On Sat, Sep 15, 2007 at 12:47:59PM +, Chris Nestrud wrote:
>> >> On Fri, 14 Sep 2007 20:46:13 -0400, John Albano <[EMAIL PROTECTED]> wrote:
>> >> > I'm seeing the problem on both etch and lenny releases.
>> >> >
>> >> > Linux ads04 2.6.18 #2 SMP Wed Sep 12 15:45:10 EDT 2007 i686 GNU/Linux
>> >> 
>> >> I have a similar problem and am also using Debian (lenny). I'm using an
>> >> SMP kernel. Could that be the issue?
>> >> 
>> >> I tested with an AGI script, and the problem is that audio isn't sent.
>> >> The script receives DTMF digits and otherwise acts as expected.
>> >
>> > Which kernel exactly?
>> Debian's linux-image-2.6.22-2-686, version 2.6.22-4.
>> 
>> > What is the output of:  uname -a
>> [EMAIL PROTECTED]:~# uname -a
>> Linux a1271.userdns.net 2.6.22-2-686 #1 SMP Fri Aug 31 00:24:01 UTC 2007
>> i686 GNU/Linux 
>> 
>> I thought this might be due to the fact that this is a Dual Core
>> Processor, so I tested using "maxcpus=1" as a parameter to the kernel. This 
>> did not
>> resolve the problem. The CPU is:
>> 
>> model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4200+ 
>
> Can you try zaptel from 1.4 SVN, or even ztdummy from 1.4 SVN?
>
>  http://svn.digium.com/svn/zaptel/branches/1.4/ztdummy.c
>
> It should now use high-resolution timers for 2.6.22 users , so this may
> work around your problem.

Thank you. I compiled and installed from the 1.4 branch and this has
solved the problem. Audio and conferencing work as expected.

--- Results after 22 passes ---
Best: 99.998 -- Worst: 99.995 -- Average: 99.996866, Difference: 99.996866 
-- 
Chris Nestrud
Email: [EMAIL PROTECTED]
http://www.panix.com/~ccn/


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Re: [asterisk-users] Paging to external speaker like in airports etc...

[C F]

Bogen http://WWW.BOGEN.COM

On 9/13/07, Deepak Naidu <[EMAIL PROTECTED]> wrote:
> Hi, I have a production asterisk-1.2.8 system with FreePBX & PRI Digium
> card.
>
> I am looking for a paging system to an external speaker.  I can page to
> internal Polycom 501 VoIP.
>
> But, what hardware or system do I need to integrate with the asterisk to
> have this acheived.
>
> --
> Deepak
>
>
>
> Linux your Life, Don't Window it [[]]
>
>{ All for the best }
>
>
>
>
> -
>  Yahoo! Answers - Get better answers from someone who knows. Tryit now.

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Re: [asterisk-users] CallWithUs Service?

[Al lists]

Actually Cbeyond does that and their quality of voice is much better analog
lines.


On 9/15/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
> Your best bet is to get your VoIP service through whoever your ISP is.
> If Global Crossing offered cheap VoIP (in comparision to some of their
> TDM offerings), I would consider it.  It's all IP in the core now
> anyways, no real reason to use TDM for the last mile.
>
> Maybe it has something to do with the number of simultaneous calls you
> can stuff down a data T1 using G729
>
> Thanks,
> Steve
>
> Al lists wrote:
> > In VOIP, your quality of your voice is as good as your network.
> > if you want clear call quality, QOS is a must.
> > Well, when the call leaves your network and enters internet, QOS is
> > not enforced.
> > As a general rule choose the closest to your network.
> > for me its Teliax, i get to their proxy after 7 hops.
> >
> >
> > On 9/14/07, *Anthony Messina* <[EMAIL PROTECTED]
> > > wrote:
> >
> > On Thursday 13 September 2007 02:32:52 pm John Meksavan wrote:
> > > I am thinking about selecting CALLWITHUS as my sip provider. Has
> > anybody
> > > ever used them? How was the call quality? DTMF Tones issues?
> >
> > it was your message that prompted me to take a look at
> > callwithus.com .
> >
> > i currently use diamondcard.us  (via iax2)
> > and have had only 2 issues in 9
> > months where some calls to verizon cell phones would get a
> > congestion signal
> > if they didn't answer instead of going to their voicemail.  i called
> > diamondcard and they fixed the trunk issue in a matter of an
> > hour.  call
> > quality is decent.
> >
> > after signing up with callwithus.com , i
> > find the call quality to be the same
> > as diamondcard, though diamondcard bills in 30sec increments at
> > 1.7 cents/min
> > in the us and callwithus bills in 1 minute increments at 1.4
> > cents/min in the
> > us.
> >
> > callwithus also has this thing where if you add a *31 to the
> > number, it will
> > choose their cheapest route.
> >
> > i'd say they are worth trying, so is diamondcard.us
> > .
> >
> > --
> > Anthony -  http://messinet.com -
> > http://messinet.com/~amessina/gallery
> > 
> > 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
> >
> > ___
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> >
> > 
> >
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Re: [asterisk-users] ztdummy kills audio

[Tzafrir Cohen]

On Sat, Sep 15, 2007 at 01:30:26PM +, Chris Nestrud wrote:
> On Sat, 15 Sep 2007 16:03:07 +0300, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> > On Sat, Sep 15, 2007 at 12:47:59PM +, Chris Nestrud wrote:
> >> On Fri, 14 Sep 2007 20:46:13 -0400, John Albano <[EMAIL PROTECTED]> wrote:
> >> > I'm seeing the problem on both etch and lenny releases.
> >> >
> >> > Linux ads04 2.6.18 #2 SMP Wed Sep 12 15:45:10 EDT 2007 i686 GNU/Linux
> >> 
> >> I have a similar problem and am also using Debian (lenny). I'm using an
> >> SMP kernel. Could that be the issue?
> >> 
> >> I tested with an AGI script, and the problem is that audio isn't sent.
> >> The script receives DTMF digits and otherwise acts as expected.
> >
> > Which kernel exactly?
> Debian's linux-image-2.6.22-2-686, version 2.6.22-4.
> 
> > What is the output of:  uname -a
> [EMAIL PROTECTED]:~# uname -a
> Linux a1271.userdns.net 2.6.22-2-686 #1 SMP Fri Aug 31 00:24:01 UTC 2007
> i686 GNU/Linux 
> 
> I thought this might be due to the fact that this is a Dual Core
> Processor, so I tested using "maxcpus=1" as a parameter to the kernel. This 
> did not
> resolve the problem. The CPU is:
> 
> model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4200+ 

Can you try zaptel from 1.4 SVN, or even ztdummy from 1.4 SVN?

 http://svn.digium.com/svn/zaptel/branches/1.4/ztdummy.c

It should now use high-resolution timers for 2.6.22 users , so this may
work around your problem.

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

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Re: [asterisk-users] DECT SIP phones

[shadowym]

You never really specified what you want this for.  If it is for enterprise
type installations then Aastra has a very robust SIP DECT solution
specifically designed for multiple roaming extensions.  When you go through
their webinar training they provide all the calculations in terms of square
footage site surveying, noise factors, expected call volume etc. 

-Original Message-
From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] 
Sent: Friday, September 14, 2007 6:33 PM
To: Tilghman Lesher
Cc: Asterisk -Users
Subject: Re: [asterisk-users] DECT SIP phones

On Fri, 2007-09-14 at 12:00 -0500,
[EMAIL PROTECTED] wrote:
> Date: Fri, 14 Sep 2007 09:32:35 -0500
> From: Tilghman Lesher <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] DECT SIP phones
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain;  charset="iso-8859-1"
> 
> On Thursday 13 September 2007 19:05:51 Stephen Bosch wrote:
> > I'm looking for a SIP DECT (cordless) phone for North American
> > installations. I've heard only of the Siemens Gigaset S450/C450
> phones.
> > Apparently these aren't sold for use in NAm, even though they're
> > supposed to be legal (in the United States, anyway).
> >
> > On top of that, I understand they have some annoying issues anyway.
> >
> > Can anyone suggest a solid alternative DECT SIP phone that is
> available
> > in North America?
> 
> I don't know how solid you would consider them, but I have repurposed
> the
> ATS X10001P phones that are sold for use with Lingo into phones that
> can
> be used with Asterisk.  At $70US, I suspect they are the least
> expensive
> SIP DECT phones available.

Wal-Mart sells the ATS X10001P for $55, and claims it has a "fax
port":
http://www.walmart.com/catalog/product.do?dest=97&product_id=6457851
&sourceid=1503142050 . Is there a way to fax with these phones
without Lingo? How does Lingo do it (over the phone's Internet connection),
if Asterisk can't?


> http://asterisk.drunkcoder.com/hacks/ats-config/

Your server seems very slow, often timing out.

 
> -- 
> Tilghman
> 
-- 

(C) Matthew Rubenstein





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Re: [asterisk-users] AGI script fails on IAX channels (from call file).

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Jonas Arndt <[EMAIL PROTECTED]> wrote:
> 
> The problem was that the IAX2 channel accepted both ulaw and gsm codec.
> Once I dropped the gsm alternative in the iax.conf the interaction with
> the AGI scripts (DTMF) worked great. So it seems that the ulaw was
> picked if the IAX call was initiated by the IAX phone and hit the AGI
> script through the extensions.conf. If the AGI script initiated the call
> gsm was for some reason picked and  wouldn't accept DTMF.
> 
> What might be a clue here is that the iaxy based phones are configured
> to just use ulaw. I tried just gsm but that didn't work. So with a phone
> doing just ulaw initiating the call, that might be the reason ulaw was
> picked and it worked through the dial plan.
> 
> Anyhow, a big thanks to everybody that tried to work on this with me. I
> have some job still ahead (figuring out how the different DTMF modes
> relates to gsm and if there might be a bug here), but I do have a
> working solution.

Jonas, glad you've found a solution, hope the suggestions were of some help.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

[shadowym]

Sorry but your not going to drag me into another one of these "you
ungrateful bastard" type arguments.  If you want to take the pepsi challenge
as to who contributes how much in what way then email me offline with your
list and I'll send you mine!

-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED] 
Sent: Friday, September 14, 2007 3:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

shadowym wrote:
> Yes thank you for reminding me it is open source.  Thank you for reminding
> me people can write their own code for it.
> 
> I'll get right on rewriting the entire sip code.  Should only take me a
few
> hours.  Including a couple hours to learn how to write c code.  How hard
can
> it be!
> 

I can't tell whether you're intending to prove the point that was being 
made, or trying to be sarcastic.  Knowing your posting history, I'll 
assume the latter.

But in case you're serious, and you really do believe the coders owe you 
something, here's another translation of the situation:

If you code, if you contribute to the coding effort by intense testing 
and/or filing bug reports, if you carry Red Bull to the programmers 
during hacking sessions, etc., then--in the vernacular of the Church of 
the Subgenius--you buy slack.

And once you have slack, you can say, "Let's do this," or "Let's do 
that," and the developers will consider it and--maybe--implement it.

When, instead, you are 100% slack-free and have been noted before 
nipping nasty mots at the hands that feed you code, the chances of 
having your tart remarks about SLA taken seriously are pretty slim.

But, and here's  the point: It's Open Source.  If the developers look 
the other way when you ask for something, if they don't answer your 
emails, if they don't drop everything when you demand something and do 
what you want,

FORK IT!  Take the code THEY they wrote and do with it what you will.

It's free.

b.

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.





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Re: [asterisk-users] Astribank and caller ID from PSTN

[Guillermo Salas M.]

Hi Tzafrir:

On Sat, 2007-09-15 at 22:05 +0300, Tzafrir Cohen wrote:
> Hi Guillermo,
> 
> On Sat, Sep 15, 2007 at 01:18:49PM -0500, Guillermo Salas M. wrote:
> > Hello,
> > 
> > I've one astribank with 8 FXO unit and 8 pstn lines connected to the
> > astribank. When I receive calls on my ipphone I get always Unknown
> > callerid.


[..]

> 
> One thing I suspect is not waiting enough.
> Try adding the following to your dialplan:
> 
> [pstn-test]
> exten => s,1,Wait(1)
> exten => s,n,NoOp(Got number ${CALLERID(all)} on ${CHANNEL})
> ; If you're just testing:
> ;exten => s,n,Playback(tt-monkeys)
> exten => s,n,Goto(from-zaptel,s,1)
> 

I've added and  used the pstn-test with the channel 4,  before I've
tested it connecting a phone and calling to the channel4 number from my
cell, the phone shows me the number of my mobile.


> And then set in zapata.conf:  context=pstn-test
> 

Done, this is the output of the log when I've one incoming call to the
channel4:

[Sep 15 14:39:13] VERBOSE[25265] logger.c: -- Executing
[EMAIL PROTECTED]:1] Wait("Zap/4-1", "1") in new stack
[Sep 15 14:39:14] VERBOSE[25265] logger.c: -- Executing
[EMAIL PROTECTED]:2] NoOp("Zap/4-1", "Got number "" <> on Zap/4-1") in new
stack
[Sep 15 14:39:14] VERBOSE[25265] logger.c: -- Executing
[EMAIL PROTECTED]:3] Goto("Zap/4-1", "from-zaptel|s|1") in new stack
[Sep 15 14:39:14] VERBOSE[25265] logger.c: -- Goto (from-zaptel,s,1)
[Sep 15 14:39:14] VERBOSE[25265] logger.c: -- Executing
[EMAIL PROTECTED]:1] NoOp("Zap/4-1", "Entering from-zaptel with DID == ")
in new stack




> 
> Other than that, there are two obvious sanity checks:
> 
> 1. Connect an analog phone with with caller ID display to the same port
> and see that caller ID is indeed detected
> 

Done, the phone is detecting the ID.

> 2. boot the same system from our live CD and see if caller ID is
> detected there.
> 


I'm going to download the live CD from www.xorcom.com . 

Thank you for your suggestions,


-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net
SIP  : [EMAIL PROTECTED]
FWD  : 558563

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


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Re: [asterisk-users] AGI script fails on IAX channels (from call file).

[Jonas Arndt]

Guys,

The problem was that the IAX2 channel accepted both ulaw and gsm codec.
Once I dropped the gsm alternative in the iax.conf the interaction with
the AGI scripts (DTMF) worked great. So it seems that the ulaw was
picked if the IAX call was initiated by the IAX phone and hit the AGI
script through the extensions.conf. If the AGI script initiated the call
gsm was for some reason picked and  wouldn't accept DTMF.

What might be a clue here is that the iaxy based phones are configured
to just use ulaw. I tried just gsm but that didn't work. So with a phone
doing just ulaw initiating the call, that might be the reason ulaw was
picked and it worked through the dial plan.

Anyhow, a big thanks to everybody that tried to work on this with me. I
have some job still ahead (figuring out how the different DTMF modes
relates to gsm and if there might be a bug here), but I do have a
working solution.

Thanks,

// Jonas

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Re: [asterisk-users] Astribank and caller ID from PSTN

[Tzafrir Cohen]

Hi Guillermo,

On Sat, Sep 15, 2007 at 01:18:49PM -0500, Guillermo Salas M. wrote:
> Hello,
> 
> I've one astribank with 8 FXO unit and 8 pstn lines connected to the
> astribank. When I receive calls on my ipphone I get always Unknown
> callerid.

> 
> It's is possible to receive the callerid from the lines on the astribank
> unit? This is my config:
> 
> [channels]
> language=es
> context=from-zaptel
> signalling=fxs_ks
> ;rxwink=300
> usecallerid=yes
> callerid=asreceived
> ;cidsignalling=bell
> ;cidstart=ring
> hidecallerid=no
> callwaiting=yes
> ;usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> ;callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> echotraining=yes
> echotraining=800
> relaxdtmf=yes
> rxgain=3.0
> txgain=3.0
> callgroup=1
> pickupgroup=1
> ;immediate=no
> callerid=asreceived
> ;amaflags=default
> busydetect=yes
> busycount=8
> ;busypattern=500,500
> answeronpolarityswitch=no
> hanguponpolarityswitch=no
> faxdetect=both
> 
> 
> ; Span 1: XBUS-00/XPD-00 "Xorcom XPD #0/0: FXO"
> ;;; line="1 XPP_FXO/0/0/0 FXSKS"
> signalling=fxs_ks
> callerid=asreceived
> group=1
> context=from-zaptel
> channel => 1
> 
> 
> When replacing callerid=phone-number I get on my ipphone phone-number as
> callerid:
> 
> ; Span 1: XBUS-00/XPD-00 "Xorcom XPD #0/0: FXO"
> ;;; line="1 XPP_FXO/0/0/0 FXSKS"
> signalling=fxs_ks
> callerid=2627839
> group=1
> context=from-zaptel
> channel => 1


One thing I suspect is not waiting enough.
Try adding the following to your dialplan:

[pstn-test]
exten => s,1,Wait(1)
exten => s,n,NoOp(Got number ${CALLERID(all)} on ${CHANNEL})
; If you're just testing:
;exten => s,n,Playback(tt-monkeys)
exten => s,n,Goto(from-zaptel,s,1)

And then set in zapata.conf:  context=pstn-test


Other than that, there are two obvious sanity checks:

1. Connect an analog phone with with caller ID display to the same port
and see that caller ID is indeed detected

2. boot the same system from our live CD and see if caller ID is
detected there.

> 
> 
> Regards,
> 
> -- 
> Guillermo Salas M.
> Telconet S.A.
> Calle 15 y Avenida 24 Esq
> Edificio Barre #2 Primer Piso
> Telefono : +593 5 262 8071
> Celular  : +593 9 985 5138
> e-mail   : [EMAIL PROTECTED]
> www  : http://www.manta.telconet.net
>http://www.telcocarrier.net
> SIP  : [EMAIL PROTECTED]
> FWD  : 558563
> 
> Linux User: 255902
> 
> Beat me, whip me, make me use Windows!
> 
> Please avoid sending me Word or PowerPoint attachments.
> See http://www.fsf.org/philosophy/no-word-attachments.html
> 
> Please avoid the Top Posting, see
> http://es.wikipedia.org/wiki/Top-posting
> 
> 
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> Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 
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-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] AGI/PHP: missing arguments

[Philipp Kempgen]

Michael Kamleitner wrote:

> I've built a simple PHP-script utilizing the AGI-interface. in
> extensions.conf I trigger the script and pass a single value as first
> argument:
> 
> exten => h,1,DeadAGI(process.php|${Enter})
> 
> On the Asterisk-console, I can actually see that the script is called
> correctly (something like "DeadAGI(process.php|1234)"). However, when I read
> stdin in the PHP script, I receive all AGI-environment variables
> (agi_request, agi_callerid etc.) correctly, but I'm missing the actual
> passed value (which should be in agi_arg_1 etc.). the last thing I get from
> stdin is the environment-variable agi_accountcode, after this it seems to
> stop.

You don't append the argument to STDIN (which is fine).
In the PHP script check the $argv array. The first argument
(after the name of the script itself) should be in $argv[1] .

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de
  My pick of the month: rfc 2822 3.6.5

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] Astribank and caller ID from PSTN

[Nasir Iqbal]

Hi,


uncomment "immediate=no"

Regards

Nasir Iqbal
ICT Innovations
http://ictinnovations.com

On Sat, 2007-09-15 at 13:18 -0500, Guillermo Salas M. wrote:
> Hello,
> 
> I've one astribank with 8 FXO unit and 8 pstn lines connected to the
> astribank. When I receive calls on my ipphone I get always Unknown
> callerid.
> 
> It's is possible to receive the callerid from the lines on the astribank
> unit? This is my config:
> 
> [channels]
> language=es
> context=from-zaptel
> signalling=fxs_ks
> ;rxwink=300
> usecallerid=yes
> callerid=asreceived
> ;cidsignalling=bell
> ;cidstart=ring
> hidecallerid=no
> callwaiting=yes
> ;usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> ;callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> echotraining=yes
> echotraining=800
> relaxdtmf=yes
> rxgain=3.0
> txgain=3.0
> callgroup=1
> pickupgroup=1
> ;immediate=no
> callerid=asreceived
> ;amaflags=default
> busydetect=yes
> busycount=8
> ;busypattern=500,500
> answeronpolarityswitch=no
> hanguponpolarityswitch=no
> faxdetect=both
> 
> 
> ; Span 1: XBUS-00/XPD-00 "Xorcom XPD #0/0: FXO"
> ;;; line="1 XPP_FXO/0/0/0 FXSKS"
> signalling=fxs_ks
> callerid=asreceived
> group=1
> context=from-zaptel
> channel => 1
> 
> 
> When replacing callerid=phone-number I get on my ipphone phone-number as
> callerid:
> 
> ; Span 1: XBUS-00/XPD-00 "Xorcom XPD #0/0: FXO"
> ;;; line="1 XPP_FXO/0/0/0 FXSKS"
> signalling=fxs_ks
> callerid=2627839
> group=1
> context=from-zaptel
> channel => 1
> 
> 
> Regards,
> 


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Re: [asterisk-users] AGI script fails on IAX channels (from call file).

[Jonas Arndt]

Tony Mountifield wrote:
> a separate context [foo], with a wildcard extension _X., which will match
> any extension of two or more digits. I then put the extension number into
> the parameter list for the AGI.
>
> So instead of generating "data: test.agi|12345" in the call file you
> generate "extension: 12345" and that then finds its way through to the
> AGI command.
>
>   
Sorry, I missed that. Well, as long as what we wanted to test is tested
> OK, this solution is technically the same as mine, concerning channel
> handling, so if you are still not getting DTMF, that is a problem.
>
> Please try adding a delay parameter to Answer, or a separate Wait line:
>
> exten => _X.,1,Answer(0.5)
>
> or:
>
> exten => _X.,1,Answer
> exten => _X.,n,Wait(0.5)
>
> It might be that something in the channel is not finished setting up
> before you call your AGI. I always have a small delay after answering.
>
>   
I did this. I waited as long as 2 seconds. Still the same problem
unfortunately. I can see the DTMF in the IAX trace. The AGI trace just
sits there...
>> So:
>>
>> == Conclusions 
>> IAX Phone => Dial Plan => AGI script
>> "Works with DTFM"
>> Call File => IAX Phone + AGI script  
>>"Fails, not DTMF communication"
>> Call File => IAX Phone + Extension in dial plan => AGI Script
>> "Fails, not DTMF communication"
>> SIP => Work always
>> == Conclusions 
>>
>> Unfortunately I am heading out for a week long Europe trip on Monday.
>> I'll try to play with this a bit more on Sunday and see if I can make
>> some progress.
>> 
>
> OK, hope you have some success.
>
> Cheers
> Tony
>   
I have a few other things to try, but that is more like a work-around or
at least a different way to attack the problem. Richard Lyman gave me
this link http://dynx.net/ASTERISK/gnudialer/agiIVR.agi and I will look
into that as well.

Generally I have had quite some issues since upgrading to 1.4 (IVR DTMF
fails to be detected sometimes on incoming calls, with SIP => IAX
providers I get dropped incoming audio). I am attacking one problem at a
time though.

// Jonas


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Re: [asterisk-users] AGI/PHP: missing arguments

[Nasir Iqbal]

Hi Michael,

Actually parameter passed to AGI script are not "Channel Variables" and
they passed to PHP/AGI directly so you cannot access them using STDIN. 

to access passed parameters simply use global variable argv like.

global $argv;

//Getting input data (Parameter Passed to Script)
$callerID  = $argv[1];


Regards

Nasir Iqbal
ICT Innovations
http://ictinnovations.com


On Sat, 2007-09-15 at 18:21 +0200, Michael Kamleitner wrote:
> hi folks,
> 
> I've built a simple PHP-script utilizing the AGI-interface. in
> extensions.conf I trigger the script and pass a single value as first
> argument:
> 
> exten => h,1,DeadAGI(process.php|${Enter})
> 
> On the Asterisk-console, I can actually see that the script is called
> correctly (something like "DeadAGI(process.php|1234)"). However, when
> I read stdin in the PHP script, I receive all AGI-environment
> variables (agi_request, agi_callerid etc.) correctly, but I'm missing
> the actual passed value (which should be in agi_arg_1 etc.). the last
> thing I get from stdin is the environment-variable agi_accountcode,
> after this it seems to stop. 
> 
> what's really strange is, that the exact same script has been working
> correctly on a different machine...
> 
> any suggestions highly appreciated, thx!
> 
> regards,
> michael
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Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

[Steve Totaro]

The is the real meat of the article is that the Microsoft tsunami is 
coming, no it wont be like the iPhone, it will be real and hard hitting 
and unlike a tsunami, it will continue to get stronger.

"Digium’s strategy is fairly straightforward. Write more code, package 
Asterisk better, educate users and resellers, and be educated by users 
and resellers. The last item is the most important, because users and 
resellers will tell Digium what they want to provide new product ideas 
and new product direction.'

I am just not sure that the way Digium is locking "resellers" into 
exclusivity deals and doling out "approved" resellers through their GUI 
and Netxusa is the proper direction for an Open Source product to go. 
That is my feedback as far as product direction.

Thanks,
Steve Totaro

shadowym wrote:
>
> Maybe his comments were taken out of context as they don’t have the 
> whole interview posted. Why is he talking about queue games, 
> Biologicall and other extremely niche crap when there are huge holes 
> in the basic offering (SLA and SCA)?
>
> *From:* Al lists [mailto:[EMAIL PROTECTED]
> *Sent:* Tuesday, September 11, 2007 8:28 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Mark Spencer: Digium is Growing Up 
> (VONMAG)
>
> I liked the queue game concept!
> although it could be cruel!
>
> On 9/11/07, *Steve Totaro* <[EMAIL PROTECTED] 
> > wrote:
>
> http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up
>
> Seems the Adtran relationship goes way back...
>
> Thanks,
> Steve Totaro
>
> _
>


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[asterisk-users] Astribank and caller ID from PSTN

[Guillermo Salas M.]

Hello,

I've one astribank with 8 FXO unit and 8 pstn lines connected to the
astribank. When I receive calls on my ipphone I get always Unknown
callerid.

It's is possible to receive the callerid from the lines on the astribank
unit? This is my config:

[channels]
language=es
context=from-zaptel
signalling=fxs_ks
;rxwink=300
usecallerid=yes
callerid=asreceived
;cidsignalling=bell
;cidstart=ring
hidecallerid=no
callwaiting=yes
;usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
;callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
relaxdtmf=yes
rxgain=3.0
txgain=3.0
callgroup=1
pickupgroup=1
;immediate=no
callerid=asreceived
;amaflags=default
busydetect=yes
busycount=8
;busypattern=500,500
answeronpolarityswitch=no
hanguponpolarityswitch=no
faxdetect=both


; Span 1: XBUS-00/XPD-00 "Xorcom XPD #0/0: FXO"
;;; line="1 XPP_FXO/0/0/0 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=1
context=from-zaptel
channel => 1


When replacing callerid=phone-number I get on my ipphone phone-number as
callerid:

; Span 1: XBUS-00/XPD-00 "Xorcom XPD #0/0: FXO"
;;; line="1 XPP_FXO/0/0/0 FXSKS"
signalling=fxs_ks
callerid=2627839
group=1
context=from-zaptel
channel => 1


Regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net
SIP  : [EMAIL PROTECTED]
FWD  : 558563

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


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Re: [asterisk-users] External FXO port.

[Guillermo Salas M.]

On Sat, 2007-09-15 at 19:25 +0530, Sanspareils Greenlans wrote:
> Sir, 
> 
> I have an audiocode MP-118 8 port external FXO gateway and i have connect 
> pstn 
> line to FXO gateway now i want to dial outside call using FXO gateway and 
> receive all outside call. but i donot know what i have add in sip.conf and 
> extension.conf to make it possible. 
> I have also attach digium TDM02b card on asterisk server and all incoming and 
> outgoing call going perfectly. but not sure how to define call receive or 
> dial through external FXO gateway. 
> 
> 
> Please give me information how we can do that. 
> 


You must have to create the config for the device on sip.conf, check the
following link, may can help you:

http://www.trixbox.org/forums/trixbox-forums/share-your-trixbox-success-stories/mp-118-and-trixbox-integration-success


Regards,


-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net
SIP  : [EMAIL PROTECTED]
FWD  : 558563

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


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Re: [asterisk-users] CallWithUs Service?

[Steve Totaro]

Your best bet is to get your VoIP service through whoever your ISP is.  
If Global Crossing offered cheap VoIP (in comparision to some of their 
TDM offerings), I would consider it.  It's all IP in the core now 
anyways, no real reason to use TDM for the last mile. 

Maybe it has something to do with the number of simultaneous calls you 
can stuff down a data T1 using G729

Thanks,
Steve

Al lists wrote:
> In VOIP, your quality of your voice is as good as your network.
> if you want clear call quality, QOS is a must.
> Well, when the call leaves your network and enters internet, QOS is 
> not enforced.
> As a general rule choose the closest to your network.
> for me its Teliax, i get to their proxy after 7 hops.
>
>
> On 9/14/07, *Anthony Messina* <[EMAIL PROTECTED] 
> > wrote:
>
> On Thursday 13 September 2007 02:32:52 pm John Meksavan wrote:
> > I am thinking about selecting CALLWITHUS as my sip provider. Has
> anybody
> > ever used them? How was the call quality? DTMF Tones issues?
>
> it was your message that prompted me to take a look at
> callwithus.com .
>
> i currently use diamondcard.us  (via iax2)
> and have had only 2 issues in 9
> months where some calls to verizon cell phones would get a
> congestion signal
> if they didn't answer instead of going to their voicemail.  i called
> diamondcard and they fixed the trunk issue in a matter of an
> hour.  call
> quality is decent.
>
> after signing up with callwithus.com , i
> find the call quality to be the same
> as diamondcard, though diamondcard bills in 30sec increments at
> 1.7 cents/min
> in the us and callwithus bills in 1 minute increments at 1.4
> cents/min in the
> us.
>
> callwithus also has this thing where if you add a *31 to the
> number, it will
> choose their cheapest route.
>
> i'd say they are worth trying, so is diamondcard.us
> .
>
> --
> Anthony -  http://messinet.com -
> http://messinet.com/~amessina/gallery
> 
> 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
>
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>
> 
>
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Re: [asterisk-users] AGI script fails on IAX channels (from call file).

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Jonas Arndt <[EMAIL PROTECTED]> wrote:
> Tony Mountifield wrote:
> > In article <[EMAIL PROTECTED]>,
> > Jonas Arndt <[EMAIL PROTECTED]> wrote:
> >   
> >> Call File
> >> === Call File ==
> >> channel: Local/[EMAIL PROTECTED]
> >> maxretries: 3
> >> retrytime: 60
> >> waittime: 60
> >> callerid: "Test" <*66>
> >> application: AGI
> >> data: test.agi|670507
> >> = End Call File 
> >> 
> >
> > Instead of calling the application directly from the call file, make a
> > special extension for it, and call that extension instead:
> >
> > [foo]
> > exten => _X.,1,Answer(0.5)
> > exten => _X.,2,AGI(test.agi|${EXTEN})
> >
> > And then have the call file like this:
> >
> > Channel: Local/[EMAIL PROTECTED]
> > Maxretries: 3
> > Retrytime: 60
> > Waittime: 60
> > CallerId: "Test" <*66>
> > Context: foo
> > Extension: 670507
> > Priority: 1
> >
> > The Answer line may or may not be necessary, but if you find it is, that
> > would explain why calling the AGI directly didn't work. In that case,
> > perhaps you could put the Answer within the AGI.
> 
> So the idea here is that the parameter, which I am calling the
> application with, will be dynamic. I am therefore struggling a bit with
> how your solution would work in the long run. It could be a good test
> though. Your Extension line in the call file (670507) will match that in
> the extensions.conf, right? What I have tried is

Yes, I figured that the 670507 part will be dynamic, which is why I used
a separate context [foo], with a wildcard extension _X., which will match
any extension of two or more digits. I then put the extension number into
the parameter list for the AGI.

So instead of generating "data: test.agi|12345" in the call file you
generate "extension: 12345" and that then finds its way through to the
AGI command.

>  Call File 
> channel: Local/[EMAIL PROTECTED]
> Context: internal
> Extension: *66
> Priority: 1
> 
> 
> Then in the extensions.con
> 
> = extensions.conf =
> exten => *66,1,Answer
> exten => *66,2,AGI(test.agi|670507)
> exten => *66,3,Hangup
> 
> 
> This is also not an elegant solution as I have hard coded the parameter.
> However, it is just for test. As you can see I am also answering the
> channel here. I have exactly the same problem here. On SIP phones it
> work great and on IAX (iaxy) phones it fails. I can't get the AGI script
> to see the DTMF even if pushing the keys generates events in an iax2
> trace.

OK, this solution is technically the same as mine, concerning channel
handling, so if you are still not getting DTMF, that is a problem.

Please try adding a delay parameter to Answer, or a separate Wait line:

exten => _X.,1,Answer(0.5)

or:

exten => _X.,1,Answer
exten => _X.,n,Wait(0.5)

It might be that something in the channel is not finished setting up
before you call your AGI. I always have a small delay after answering.

> So:
> 
> == Conclusions 
> IAX Phone => Dial Plan => AGI script
> "Works with DTFM"
> Call File => IAX Phone + AGI script  
>"Fails, not DTMF communication"
> Call File => IAX Phone + Extension in dial plan => AGI Script
> "Fails, not DTMF communication"
> SIP => Work always
> == Conclusions 
> 
> Unfortunately I am heading out for a week long Europe trip on Monday.
> I'll try to play with this a bit more on Sunday and see if I can make
> some progress.

OK, hope you have some success.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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[asterisk-users] AGI/PHP: missing arguments

[Michael Kamleitner]

hi folks,

I've built a simple PHP-script utilizing the AGI-interface. in
extensions.conf I trigger the script and pass a single value as first
argument:

exten => h,1,DeadAGI(process.php|${Enter})

On the Asterisk-console, I can actually see that the script is called
correctly (something like "DeadAGI(process.php|1234)"). However, when I read
stdin in the PHP script, I receive all AGI-environment variables
(agi_request, agi_callerid etc.) correctly, but I'm missing the actual
passed value (which should be in agi_arg_1 etc.). the last thing I get from
stdin is the environment-variable agi_accountcode, after this it seems to
stop.

what's really strange is, that the exact same script has been working
correctly on a different machine...

any suggestions highly appreciated, thx!

regards,
michael
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Re: [asterisk-users] bug in 1.2.24

[Anton Krall]

Thank you for the example Isaac. I did as you mentioned and now it seems to
be working perfectly.
 
Saludos
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Isaac Xiao
Sent: jueves, 13 de septiembre de 2007 10:33 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] bug in 1.2.24

Here is our dial plan. You need to avoid double recording as well when
you transfer the call to other extension.
exten => 7141,3,Set(CALLFILENAME=q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten => 7141,4,Set(__FROM-EXT-QUEUES=ext-queues)
exten => 7141,5,MixMonitor(${CALLFILENAME}.gsm|b)
exten => 7141,6,Playback(custom/None)
exten => 7141,7,Queue(7141|t|||7200)

Here is the CLI log. 
  -- Executing Playback("Zap/9-1", "monitoring") in new stack
-- Playing 'monitoring' (language 'md')
-- Executing Playback("Zap/9-1", "press-1-to-msg") in new stack
-- Playing 'press-1-to-msg' (language 'md')
-- Executing Goto("Zap/9-1", "ext-queues|7141|1") in new stack
-- Goto (ext-queues,7141,1)
-- Executing NoOp("Zap/9-1", "do not answer call before entering
queue") in new stack
-- Executing SetCIDName("Zap/9-1", "CN") in new stack
-- Executing Set("Zap/9-1",
"CALLFILENAME=q7141-20070914-132445-1189740177.10324") in new stack
-- Executing Set("Zap/9-1", "__FROM-EXT-QUEUES=ext-queues") in new
stack
-- Executing MixMonitor("Zap/9-1",
"q7141-20070914-132445-1189740177.10324.gsm|b") in new stack
-- Executing Playback("Zap/9-1", "custom/None") in new stack
-- Executing Queue("Zap/9-1", "7141|t|||7200") in new stack

So Yes. As long as Zap/9-1 channel (customer's channel) not hangs up, it
will be always recorded.

Isaac


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Re: [asterisk-users] AGI script fails on IAX channels (from call file).

[Jonas Arndt]

Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> Jonas Arndt <[EMAIL PROTECTED]> wrote:
>   
>> Call File
>> === Call File ==
>> channel: Local/[EMAIL PROTECTED]
>> maxretries: 3
>> retrytime: 60
>> waittime: 60
>> callerid: "Test" <*66>
>> application: AGI
>> data: test.agi|670507
>> = End Call File 
>> 
>
> Instead of calling the application directly from the call file, make a
> special extension for it, and call that extension instead:
>
> [foo]
> exten => _X.,1,Answer(0.5)
> exten => _X.,2,AGI(test.agi|${EXTEN})
>
> And then have the call file like this:
>
> Channel: Local/[EMAIL PROTECTED]
> Maxretries: 3
> Retrytime: 60
> Waittime: 60
> CallerId: "Test" <*66>
> Context: foo
> Extension: 670507
> Priority: 1
>
> The Answer line may or may not be necessary, but if you find it is, that
> would explain why calling the AGI directly didn't work. In that case,
> perhaps you could put the Answer within the AGI.
>
> Cheers
> Tony
>
>   


Hi Tony,

So the idea here is that the parameter, which I am calling the
application with, will be dynamic. I am therefore struggling a bit with
how your solution would work in the long run. It could be a good test
though. Your Extension line in the call file (670507) will match that in
the extensions.conf, right? What I have tried is

 Call File 
channel: Local/[EMAIL PROTECTED]
Context: internal
Extension: *66
Priority: 1


Then in the extensions.con

= extensions.conf =
exten => *66,1,Answer
exten => *66,2,AGI(test.agi|670507)
exten => *66,3,Hangup


This is also not an elegant solution as I have hard coded the parameter.
However, it is just for test. As you can see I am also answering the
channel here. I have exactly the same problem here. On SIP phones it
work great and on IAX (iaxy) phones it fails. I can't get the AGI script
to see the DTMF even if pushing the keys generates events in an iax2
trace. So:


== Conclusions 
IAX Phone => Dial Plan => AGI script
"Works with DTFM"
Call File => IAX Phone + AGI script  
   "Fails, not DTMF communication"
Call File => IAX Phone + Extension in dial plan => AGI Script
"Fails, not DTMF communication"
SIP => Work always
== Conclusions 

Unfortunately I am heading out for a week long Europe trip on Monday.
I'll try to play with this a bit more on Sunday and see if I can make
some progress.

Thanks for you help,

// Jonas

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Re: [asterisk-users] how to route outgoing calls on IP-level

[Seysan]

So take the challenge and go with ip routing.

I've mentioned everything that you needed in previous reply.


On 9/15/07, Kate Kretz <[EMAIL PROTECTED]> wrote:
>
> no.
>
> all packets come to the same h323 proxy.
> and actually asterisk acts as sip <--> h323 convertor.
>
> so, for instance, Bill Clinton calls asterisk as SIP, asterisk sees it's a
> Bill Clinton and sends h323 packets to the same h323 proxy as usual, but put
> certain outgoing IP address
>
> On 9/15/07, Seysan <[EMAIL PROTECTED]> wrote:
> >
> > Hi,
> >
> > I would recommend instead of Using IPs in your Billing, your Prefixes.
> > Most of the billing softwares can to billing based on Prefix, for
> > example when "Bill Clinton" from Extension 100 is calling, add 22 or 22# in
> > front of the calling number 22#12345678, then your billing can do the rest
> > based on the 22 prefix.
> > I'm doing such billings for 5 years now!
> >
> > Also for Routing based on IP Address, I have done this in the ISP. You
> > can MARK packets that are comming from specific extension because the IP
> > address on that extension is different, then you will ROUTE those MARKED
> > packets to the IP that you want.
> > Marking is done with "iptables" in the mangle module "iptables -r mangle
> > -I PREROUTING -s xx and blah blah" and for routing you can do it with
> > "ip" command and U32 match . ;)
> > it is a little bit complicated, But I recommend PREFIX Billing.
> >
> > Regards,
> >
> > Seysan
> >
> >
> > On 9/15/07, Kate Kretz < [EMAIL PROTECTED]> wrote:
> > >
> > > there's just one factor - customer, i.e. extension in terms of
> > > Asterisk.
> > >
> > > On 9/15/07, Joseph Bajin < [EMAIL PROTECTED] > wrote:
> > > >
> > > > What are the factors in deciding which interface the traffic needs
> > > > to
> > > > go out of?
> > > >
> > > > Is it based on IP address, is it based on the terminating carrier?
> > > >
> > > > --Joe
> > > >
> > > > On 9/14/07, Kate Kretz < [EMAIL PROTECTED]> wrote:
> > > > > Dear Sirs,
> > > > >
> > > > > out asterisk server has multiple network cards.
> > > > >
> > > > > I want some outgoing calls (from several extensions) to use one IP
> > > > address,
> > > > > and others to go through
> > > > > another address.
> > > > >
> > > > > is there a way to achive that using asterisk ?
> > > > >
> > > > > Cheers,
> > > > > Kate
> > > > >
> > > > > ___
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> > > > >
> > > >
> > > >
> > > > --
> > > > --Joe
> > > >
> > > > "Success is easy if you think of it like Rust:   It's inevitable if
> > > > you keep at it. Guys claim there are magic moments, but that's just
> > > > bullshit." --Fred Franzia (The famous wine guy)
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[asterisk-users] External FXO port.

[Sanspareils Greenlans]

Sir, 

I have an audiocode MP-118 8 port external FXO gateway and i have connect pstn 
line to FXO gateway now i want to dial outside call using FXO gateway and 
receive all outside call. but i donot know what i have add in sip.conf and 
extension.conf to make it possible. 

I have also attach digium TDM02b card on asterisk server and all incoming and 
outgoing call going perfectly. but not sure how to define call receive or 
dial through external FXO gateway. 


Please give me information how we can do that. 

Rajeev.

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Re: [asterisk-users] ztdummy kills audio

[Chris Nestrud]

On Sat, 15 Sep 2007 16:03:07 +0300, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Sat, Sep 15, 2007 at 12:47:59PM +, Chris Nestrud wrote:
>> On Fri, 14 Sep 2007 20:46:13 -0400, John Albano <[EMAIL PROTECTED]> wrote:
>> > I'm seeing the problem on both etch and lenny releases.
>> >
>> > Linux ads04 2.6.18 #2 SMP Wed Sep 12 15:45:10 EDT 2007 i686 GNU/Linux
>> 
>> I have a similar problem and am also using Debian (lenny). I'm using an
>> SMP kernel. Could that be the issue?
>> 
>> I tested with an AGI script, and the problem is that audio isn't sent.
>> The script receives DTMF digits and otherwise acts as expected.
>
> Which kernel exactly?
Debian's linux-image-2.6.22-2-686, version 2.6.22-4.

> What is the output of:  uname -a
[EMAIL PROTECTED]:~# uname -a
Linux a1271.userdns.net 2.6.22-2-686 #1 SMP Fri Aug 31 00:24:01 UTC 2007
i686 GNU/Linux 

I thought this might be due to the fact that this is a Dual Core
Processor, so I tested using "maxcpus=1" as a parameter to the kernel. This did 
not
resolve the problem. The CPU is:

model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4200+ 
-- 
Chris Nestrud
Email: [EMAIL PROTECTED]
http://www.panix.com/~ccn/


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Re: [asterisk-users] ztdummy kills audio

[Tzafrir Cohen]

On Sat, Sep 15, 2007 at 12:47:59PM +, Chris Nestrud wrote:
> On Fri, 14 Sep 2007 20:46:13 -0400, John Albano <[EMAIL PROTECTED]> wrote:
> > I'm seeing the problem on both etch and lenny releases.
> >
> > Linux ads04 2.6.18 #2 SMP Wed Sep 12 15:45:10 EDT 2007 i686 GNU/Linux
> 
> I have a similar problem and am also using Debian (lenny). I'm using an
> SMP kernel. Could that be the issue?
> 
> I tested with an AGI script, and the problem is that audio isn't sent.
> The script receives DTMF digits and otherwise acts as expected.

Which kernel exactly?

A packaged, or self-built?

What is the output of:  uname -a

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

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Re: [asterisk-users] ztdummy kills audio

[Chris Nestrud]

On Fri, 14 Sep 2007 20:46:13 -0400, John Albano <[EMAIL PROTECTED]> wrote:
> I'm seeing the problem on both etch and lenny releases.
>
> Linux ads04 2.6.18 #2 SMP Wed Sep 12 15:45:10 EDT 2007 i686 GNU/Linux

I have a similar problem and am also using Debian (lenny). I'm using an
SMP kernel. Could that be the issue?

I tested with an AGI script, and the problem is that audio isn't sent.
The script receives DTMF digits and otherwise acts as expected.
-- 
Chris Nestrud
Email: [EMAIL PROTECTED]
http://www.panix.com/~ccn/


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Re: [asterisk-users] Prevent multiple sip registrations

[Anselm Martin Hoffmeister]

Am Dienstag, den 11.09.2007, 19:09 +0500 schrieb Rizwan Hisham:
> The whole point of doing this is because if the user gives away his
> username/password to his friends or relative and allows them to use
> his account, that way we r gona have a lot more traffic in our
> asterisk server.
> Also we charge our users a fix amount of money every month for their
> account so if any user gives out his username and password then his
> account is more likely to do 2 to 3 times the calls as compared to aan
> account which is used by only one user. So ultimately we lose money. 

Dear Rizwan,

imagine one of your customers uses asterisk. His asterisk server
registers to your server, and he manages his own local dialplan to have
250 SIP devices using the one SIP account. (I think Asterisk can be told
to send a UserAgent ID other than the default "Asterisk whatever" - you
will not easily find out *reliably* wether someone is an Asterisk user
or not)

Are you screwed? Well, probably. You cannot outsmart some people if you
give them the liberty to play tricks on you.

If you want to go secure, buy the hardware they are going to use,
register all the SIP stuff into that hardware and make sure it cannot be
read-out easily (most SIP phones will not allow to read the password
that was previously entered, although some web-interfaces still contain
the old password in the HTML page source).

Your customers will hate you...

My personal approach would be to not bother with registrations but log
the IP addresses from which their phones register. If - over a busy
telephone day - the log shows a pattern like

123.45.67.89 - 11:15h
131.66.14.56 - 11:27h
123.45.67.89 - 11:58h
131.66.14.56 - 12:44h
123.45.67.89 - 14:05h
131.66.14.56 - 14:09h
123.45.67.89 - 14:32h

then you could still call the user and tell him to buy another account -
your contracts probably explicitely restrict usage to a single person,
right?

Even more, your contracts _could_ contain clauses like "for private
users only", and the option for immediate termination on your part if
any doubts on that arise (users tend to hate those statements as well).

Anyone having more than 400 outgoing minutes in more than 50 calls
(insert other numbers to your liking) on a day, or more than 7000
outgoing hours in more than 1000 calls in a month might attract your
special attention. You could have some log analysis to find power users.

Just an idea popping up: AFAIK you _can_ restrict asterisk SIP easily to
not more than one concurrent call for any account - and you probably
should with your business model. How about, once they trigger a certain
number of minutes threshold on their account (perhaps 2000 minutes
during the last 7*24 hours), preceding any outgoing call they make with
a short announcement like "*bling* your_telco_name Please be aware this
account is for private use only. Call customer service to get more
information *blong*"? At least this would sever re-selling of your
services - and legitimate users would in 99.99% of cases never hear that
announcement.

I know some SIP providers always send out CALLERID, not to be
suppressed, so those flat tarrifs are also less interesting for resale.
Some customers (like me) prefer being able to set that CALLERID, on the
other hand. And I surely do not abuse the tariffs I contracted for.

Whatever your system looks like in the end, that would of course be
interesting to me. On the other hand I can only advise you to not
publish the exact numbers, triggers and restrictions - for obvious
reasons.

Finally it all boils down to "you offer a flat fee, you suffer". Try to
attract customers that use less minutes than you calculated your tariff
for. Try make it attractive for the use it is intended for, and less
attractive for (irregular) power-users, re-sellers or call-center-like
businesses. Try to not irritate your users by unpopular, stupid
restrictions. If the world were just a better place, sometimes...

Just my 3 pence,

Anselm (just being returned from holidays in Kent, still in relaxed
mode)


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Re: [asterisk-users] alphabetical extension patterns

[Anselm Martin Hoffmeister]

Am Dienstag, den 11.09.2007, 17:11 +0530 schrieb Benjamin Jacob:
> Thanks Anselm. This does clears a few things for me.
> Tho, I couldnt find the patterns you mentioned in the docs(do point me
> to the location if you know of it).

I started on
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns

Patterns have to begin with "_", meaning it is a pattern. A "." stands
for "one or more characters", so I only allow three-and-more character
SIP phone numbers like [EMAIL PROTECTED], but not [EMAIL PROTECTED] This
is deliberate: I rather not have catchall-type phone numbers, I already
get enough mail spam on the few catchall-addresses I have (well, for
"historical" reasons - I once was small and stupid ;)

> About multiple domains, that is my target for sure.
> I think the "domain"(in sip.conf) thing should come into help here,
> where I associate a domain name to a context. I did try it once,
> worked fine for a couple of test domains. But it seems I can't
> associate one domain name to multple contexts. Am I correct?

You can specify one context for every domain your asterisk supports. On
one of my machines, a sip.conf might look like
8< sip.conf
[general]
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
domain=main.example.com,sip-in-examplecom
domain=private.example.org,sip-in-privateexampleorg
domain=customer.example.net,sip-in-customerexamplenet
>8
So calls coming in for [EMAIL PROTECTED] are going through the
sip.conf context "sip-in-examplecom".

In extensions.conf, I would configure like this:
8< extensions.conf
[sip-in-domains]
exten=>_...,1,Set(A=${DB(callroute/names/[EMAIL PROTECTED])})
exten=>_...,2,GotoIf($["A" = "A${A}"]?900)
exten=>_...,3,Goto(localdialplan,${A},1)
[sip-in-examplecom]
exten=>_...,1,Set(DOMAIN=example.com)
exten=>_...,2,Goto(sip-in-domains,${EXTEN},1)   
[sip-in-privateexampleorg]
exten=>_...,1,Set(DOMAIN=private.example.org)
exten=>_...,2,Goto(sip-in-domains,${EXTEN},1)   
[sip-in-customerexamplenet]
exten=>_...,1,Set(DOMAIN=customer.example.net)
exten=>_...,2,Goto(sip-in-domains,${EXTEN},1)   
>8

This would require database entries for users like
callroute/names/[EMAIL PROTECTED] => 201
callroute/names/[EMAIL PROTECTED] => 661

You can also have several domains map to the same users, e.g. you want
example.com and main.example.com to be equivalent, so you just add
another domain line to sip.conf, like
domain=example.com,sip-in-examplecom

You should be able to get around this multiple-context setup by using
the variable ${SIPDOMAIN} and only one context, but this somehow did not
work for me, so I came up with this solution. Play around, see if you
get it running. For me, it has been like this for a while, and then, I
try to avoid changing a running system. You could, for example, set all
your domains to
domain=example.net,sip-in-domains
and use
exten=>_...,1,Set(A=${DB(callroute/names/[EMAIL PROTECTED])})

which _should_ work just as well.

You probably already found out that SRV records should be set for the
domains that asterisk is going to handle, let me give an example:

[EMAIL PROTECTED]:~$ dig @localhost example.org any
; (1 server found)
;; global options:  printcmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 52979
;; flags: qr aa rd; QUERY: 1, ANSWER: 6, AUTHORITY: 0, ADDITIONAL: 1
;; WARNING: recursion requested but not available

;; QUESTION SECTION:
;example.org.IN  ANY

;; ANSWER SECTION:
example-org.  604800  IN SOA ns1.example.net. root.example.org.
 2007060504 21600 3600 1209600 21600
example.org.  604800  IN TXT "v=spf1 mx a:mxs.example.org -all"
example.org. 604800  IN   MX  10 example.org.
example.org. 604800  IN   A   81.12.999.999
example.org. 604800  IN   NS  ns1.example.net.
example.org. 604800  IN   NS  al25b.xi.yu.fiber.example.com.
example.org. 604800  IN   NAPTR 60 50 "s" "SIP+D2U" ""
_sip._udp.example.org.
;; ADDITIONAL SECTION:
ns1.example.net.604800  IN  A   81.12.999.999

;; Query time: 5 msec
;; SERVER: 127.0.0.1#53(127.0.0.1)
;; WHEN: Sat Sep 15 11:38:14 2007
;; MSG SIZE  rcvd: 269

Where
_sip._udp.example.org. 604800 IN SRV 10 10 5060 example.org.

This is a setup with all web, mail and sip running on the same machine
(IP addresses and domains changed, of course) - but you should be able
to move things around so that those services actually can be run on
different machines.

> Anything other to be done on Asterisk to support multiple domains?

Well, I think that is about enough ;-)

BR
Anselm



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Re: [asterisk-users] AGI script fails on IAX channels (from call file).

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Jonas Arndt <[EMAIL PROTECTED]> wrote:
> 
> Call File
> === Call File ==
> channel: Local/[EMAIL PROTECTED]
> maxretries: 3
> retrytime: 60
> waittime: 60
> callerid: "Test" <*66>
> application: AGI
> data: test.agi|670507
> = End Call File 

Instead of calling the application directly from the call file, make a
special extension for it, and call that extension instead:

[foo]
exten => _X.,1,Answer(0.5)
exten => _X.,2,AGI(test.agi|${EXTEN})

And then have the call file like this:

Channel: Local/[EMAIL PROTECTED]
Maxretries: 3
Retrytime: 60
Waittime: 60
CallerId: "Test" <*66>
Context: foo
Extension: 670507
Priority: 1

The Answer line may or may not be necessary, but if you find it is, that
would explain why calling the AGI directly didn't work. In that case,
perhaps you could put the Answer within the AGI.

Cheers
Tony

-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] how to route outgoing calls on IP-level

[Kate Kretz]

no.

all packets come to the same h323 proxy.
and actually asterisk acts as sip <--> h323 convertor.

so, for instance, Bill Clinton calls asterisk as SIP, asterisk sees it's a
Bill Clinton and sends h323 packets to the same h323 proxy as usual, but put
certain outgoing IP address

On 9/15/07, Seysan <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I would recommend instead of Using IPs in your Billing, your Prefixes.
> Most of the billing softwares can to billing based on Prefix, for example
> when "Bill Clinton" from Extension 100 is calling, add 22 or 22# in front of
> the calling number 22#12345678, then your billing can do the rest based on
> the 22 prefix.
> I'm doing such billings for 5 years now!
>
> Also for Routing based on IP Address, I have done this in the ISP. You can
> MARK packets that are comming from specific extension because the IP address
> on that extension is different, then you will ROUTE those MARKED packets to
> the IP that you want.
> Marking is done with "iptables" in the mangle module "iptables -r mangle
> -I PREROUTING -s xx and blah blah" and for routing you can do it with
> "ip" command and U32 match . ;)
> it is a little bit complicated, But I recommend PREFIX Billing.
>
> Regards,
>
> Seysan
>
>
> On 9/15/07, Kate Kretz <[EMAIL PROTECTED]> wrote:
> >
> > there's just one factor - customer, i.e. extension in terms of Asterisk.
> >
> > On 9/15/07, Joseph Bajin < [EMAIL PROTECTED] > wrote:
> > >
> > > What are the factors in deciding which interface the traffic needs to
> > > go out of?
> > >
> > > Is it based on IP address, is it based on the terminating carrier?
> > >
> > > --Joe
> > >
> > > On 9/14/07, Kate Kretz < [EMAIL PROTECTED]> wrote:
> > > > Dear Sirs,
> > > >
> > > > out asterisk server has multiple network cards.
> > > >
> > > > I want some outgoing calls (from several extensions) to use one IP
> > > address,
> > > > and others to go through
> > > > another address.
> > > >
> > > > is there a way to achive that using asterisk ?
> > > >
> > > > Cheers,
> > > > Kate
> > > >
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> > >
> > > --
> > > --Joe
> > >
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> > > you keep at it. Guys claim there are magic moments, but that's just
> > > bullshit." --Fred Franzia (The famous wine guy)
> > >
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