date:20050507

I am a user of Teliax and voipjet.  I find voipjet to be very reliable
and good for outgoing, very low lag, etc.  Teliax is good too, but I am
finding high lag rates, to the point where there is a half-second delay.

I ended up just ordering a pots line for incoming (since I am going to
be doing advertising), as the lag is annoying.

The downside is that teliax gives me a number like 816-1000, which is a
cool did number.  The local CO does not offer such numbers.

The upside, if there is a problem it is resolved quickly, and my lag
time is much lower.

Overall, If you have good ping time to teliax, they are excellent.
Otherwise, stick with pots for important inbound.  For 800, use a normal
ld company and send to a voip inbound if service is good.  This way you
have all bases covered, although at a slightly higher cost.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Johnathan
Corgan
Sent: Saturday, May 07, 2005 12:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Who's happy with their voip service?

JD wrote:

 Inbound
 calling has been down for 2 days. 

Just FYI, mine is back up (408-903) as of about five hours ago.

I did just speak with a (Broadvoice) support tech on an entirely
unrelated matter (40 min. hold time!), mentioned mine was working, and
he seemed to think things were coming back in stages.

I've had them for two months now.  People may recall a series of emails
regarding packet loss through their PNAP link to Sprintlink (my ex-ISP
backbone.)  I ditched the Sprint BBD fixed-wireless service, got
Sonic.net DSL, and have been enjoying pretty high quality voice service
since.  The packet loss rates at PNAP still show but I think now this
shows it's an artificial measure (intentionally dropped non-VOIP
packets, all the other potential reasons hashed about in that thread.)

In spite of the service outages and long hold times for support, I still
want to give them the benefit of the doubt (and my $25 monthly.)  It
still seems like growing pains vs. incompetency.

I tried their web interface to change DIDs, as they now have them in my
home area code.  The effect was instant, I reconfigured sip.conf with
the new number and secret they provide, and something like 3 minutes
later was using the new DID.  So some things do work well.

Wish they did IAX. And ILBC. Not that important to me right now, though.

-Johnathan
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[Asterisk-Users] Good NAT Pnp Hardphone

Hello All,
I am looking for a sip phone that is capable of automatic nat.  The
Cisco ata186 for example works fine for natting with iconnecthere, but
as for asterisk, both my 7960 and polycom ip600 require you to set the
nat ip on the tftp.

Does anyone know a good phone (or ata) that can do this automatically?

For example,
I want to give a phone to my brother, who is going to europe.  His ICH
softphone is fine there.  Both the poly and cisco though require you to
setup for nat.  He would not be able to set this up though, so I want to
just give him a preconfig'ed phone and plug and go... 

Any ideas?

Greg
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[Asterisk-Users] MWI Suggestion

I had been trying to get mwi working on a pingtel phone for some time,
with no success... 

The solution was simple.  When I made my voicemail.conf, I added the
boxes to the end of the file. The problem was, at the end was in a
different context, so mwi would not light.  The solution, all I had to
do was move it up to default, and everything started working fine...

Just a thought...
Greg
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[Asterisk-Users] Re: HINT

2005-05-07 Thread Thorben Jensen

 Could you please give us some more detail as to what you did, in terms of 
 configuring the hint, and specifically what changes in the behavior of the 
 running server-phone interaction as a result?


You need to set the hint for the phone when the phone is being dialed like 
this:

exten = 201,hint,SIP/201
exten = 201,1,macro(dial-sip,201)

It's important that you write the full name of the phone SIP/201 as you 
can't use substitutions like this SIP/${EXTEN} - it took me a long time to 
figure that out.

Thorben




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[Asterisk-Users] ChanIsAvail for MGCP

2005-05-07 Thread Daniel Nyström

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
ChanIsAvail does not work with MGCP channels, as said in the wiki.
But other applications works simular, like Queue and Dial.
What's really the problem with ChanIsAvail?
Is it possible to use Queue and Dial to make a working ChanIsAvail?
I will take a better look in the source when back at work on monday,
but some tips and facts will help for sure.
- --
Daniel
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.5 (MingW32)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFCfIIJ/4dZjWjLCy0RAgHpAJ9Vs6qWuzioOwvi8M/iFVLwC18Z1QCfdziU
G9bU7CzzFFiSP6Qz5LjcUCI=
=ZgJH
-END PGP SIGNATURE-
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Re: [Asterisk-Users] Re: HINT

2005-05-07 Thread Brian Capouch

Thorben Jensen wrote:
Could you please give us some more detail as to what you did, in terms of 
configuring the hint, and specifically what changes in the behavior of the 
running server-phone interaction as a result?

You need to set the hint for the phone when the phone is being dialed like 
this:

exten = 201,hint,SIP/201
exten = 201,1,macro(dial-sip,201)
It's important that you write the full name of the phone SIP/201 as you 
can't use substitutions like this SIP/${EXTEN} - it took me a long time to 
figure that out.

Yes, but then what *happens* between the server and the phone due to 
having specified the hint.

I suspect others share my lack of comprehension as to what that then 
buys you--what then can you do that you can't do without that line in 
the dialplan.

Thx.
B.
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[Asterisk-Users] DTMF detection with Adit 600

2005-05-07 Thread Daniel Nyström

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
It seems like Asterisk are having problems detecting DTMF digits when
using an Adit 600 channel bank via MGCP.
I've tried to turn on RFC 2833 on both Adit and Asterisk, but no
digits at all are working then.
Anyone experienced simular with Adit or other channel banks?
I'm also unable to use V.90 modem through my setup (Adit600 via MGCP
- - Asterisk - E1).
Fax worked once though.. Does the Echo Cancelling make the problems
with V.90?
- --
Daniel http://www.faqs.org/rfcs/rfc2833.html
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.5 (MingW32)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFCfIMg/4dZjWjLCy0RAkIDAKCKskwPa5nHURRKBADPccqNYTrSYACbB09i
/TpbLaIZHRdd7K0iPLiwi2o=
=HtCY
-END PGP SIGNATURE-
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Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

Jay Milk wrote:
You got your groups mixed up.  Should be:
[default]
exten = _.,1,Dial(ZAP/g2/${EXTEN})
[outgoing]
exten = _.,1,Dial(ZAP/g1/${EXTEN})
Means that anything coming in to channel-group 1 (default context) will
be sent out through group 2, and vice versa.
Watch the console and be amazed when _. matches extension h, which is 
called when the far side of the call hangs up.  You get two calls to the 
same number by only dialing once!  Stop being lazy and at least use _X. 
as your pattern.
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Re: [Asterisk-Users] Music on Hold

2005-05-07 Thread Sahil Gupta

[EMAIL PROTECTED]:~# mpg123 -v
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp.
Uses code from various people. See 'README' for more!
THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!
Regards,
Sahil Gupta
VoiceValley
On Sat, 7 May 2005, Matt Riddell wrote:
Sahil Gupta wrote:
Assuming you have 0.59r (which you should), which codec is the call 
using?

ulaw to the box.
And do you definitely have 0.59r?  Paste us the output you get when you type 
mpg123 -v

--
Cheers,
Matt Riddell
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[Asterisk-Users] DTMF generated from phone or from gateway?

2005-05-07 Thread Kido NOAGBODJI

Hello all,

I was wondering if the DTMF were generated from the phone or from the
ATA? I have a cisco ATA 186.

Thanks

K.

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Re: [Asterisk-Users] CVS question

2005-05-07 Thread administrator tootai

Mark Johnson a écrit :
Is there a way to get a download of asterisk from cvs-head as of like 
3 weeks ago?  Having some weird problems and most people say that alot 
of these things have been introduced over the last few weeks.
cvs co -D 2005-02-15 asterisk
will give you the 15 february 2005 version.
--
Daniel
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Re: [Asterisk-Users] Asterisk on SMP machine with ztdummy working ?

2005-05-07 Thread Vamsi Pottangi

That is nice to hear. Congrats.
Wondering who could help me out with this unique zap channel problem of mine.

Thanks,
~Vamsi

On 5/7/05, Tim Connolly [EMAIL PROTECTED] wrote:
 I've got three dual Xeon's running Redhat Enterprise 4 with 2.6.9
 and CVS-HEAD from about a month ago. I didn't have any problems whatsoever,
 other than the problems I blame on being reluctant to RTFM. No problems with
 the SMP side whatsoever.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Vamsi Pottangi
 Sent: Friday, May 06, 2005 10:50 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Asterisk on SMP machine with ztdummy working ?
 
 Hi All,
 
 Was any Asterisk installation on SMP machine successful. Were you able
 to get ztdummt working on it. If so please let me know which linux
 favour you are using and any important steps to follow.
 I have a Dell Power edge 2800 and wanted to try asterisk on it and
 also use meetme. Which Linux flavour should I go for and the timing
 source. I don't have a zaptel interface so wanted to use ztdummy.
 Please guide me.  I tried with FC3 as mentioned in below mail but
 loading of zap module fails saying resource busy.
 
 Thanks,
 ~Vamsi
 
 -- Forwarded message --
 From: Vamsi Pottangi [EMAIL PROTECTED]
 Date: May 5, 2005 7:51 PM
 Subject: chan_zap.so: load_module fails: Fedora Core 3: SMP
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 Hi,
 
 I'm trying to install asterisk on Dell power edge 2800 running Fedora core
 3.
 I don't have have any zaptel cards, so trying to use ztdummy.
 /dev/zap is successfuly created... but I see some problems while
 starting asterisk ... chan_zap fails to load.
 Can somebody please help me in overcoming this problem.
 I was able to run asterisk on other normal PCs running Fedora core 3.
 Is this something to do with SMP ? I compile zaptel using the link
 to smp source code only.
 
 lrwxrwxrwx   1 root root  34 May  5 21:22 linux-2.6 -
 /lib/modules/2.6.9-1.667smp/source
 
 May  5 21:43:55 VERBOSE[12931]:  [chan_zap.so]May  5 21:43:55
 VERBOSE[12931]:  [chan_zap.so] = (Zapata Telephony)
 May  5 21:43:55 DEBUG[12931]: Parsing /etc/asterisk/zapata.conf
 May  5 21:43:55 WARNING[12931]: Unable to specify channel 1: Device or
 resource busy
 May  5 21:43:55 ERROR[12931]: Unable to open channel 1: Device or resource
 busy
 here = 0, tmp-channel = 1, channel = 1
 May  5 21:43:55 ERROR[12931]: Unable to register channel '1'
 May  5 21:43:55 WARNING[12931]: chan_zap.so: load_module failed, returning
 -1
 May  5 21:43:55 DEBUG[12931]: Unregistering channel type 'Zap'
 May  5 21:43:55 VERBOSE[12931]:   == Unregistered channel type 'Zap'
 May  5 21:43:55 WARNING[12931]: Loading module chan_zap.so failed!
 
 [EMAIL PROTECTED] ~]# uname -a
 Linux noname11 2.6.9-1.667smp #1 SMP Tue Nov 2 14:59:52 EST 2004 i686
 i686 i386 GNU/Linux
 [EMAIL PROTECTED] ~]#
 
 [EMAIL PROTECTED] ~]# ls -l /dev/zap/
 total 0
 crw---  1 asterisk asterisk 196, 254 May  5 21:31 channel
 crw---  1 asterisk asterisk 196,   0 May  5 21:31 ctl
 crw---  1 asterisk asterisk 196, 255 May  5 21:31 pseudo
 crw---  1 asterisk asterisk 196, 253 May  5 21:31 timer
 [EMAIL PROTECTED] ~]#
 
 Thanks,
 ~Vamsi
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Re: [Asterisk-Users] Re: chan_zap.so: load_module fails: Fedora Core 3: SMP

2005-05-07 Thread Vamsi Pottangi

Bulls Eye !!! Thanks for that Tony !
It worked.  Initially I thought that default conf file would work like
my previous installations.

Thanks,
~Vamsi

On 5/7/05, Tony Mountifield [EMAIL PROTECTED] wrote:
 In article [EMAIL PROTECTED],
 Vamsi Pottangi [EMAIL PROTECTED] wrote:
 
  I'm trying to install asterisk on Dell power edge 2800 running Fedora core 
  3.
  I don't have have any zaptel cards, so trying to use ztdummy.
  /dev/zap is successfuly created... but I see some problems while
  starting asterisk ... chan_zap fails to load.
  Can somebody please help me in overcoming this problem.
  I was able to run asterisk on other normal PCs running Fedora core 3.
  Is this something to do with SMP ? I compile zaptel using the link
  to smp source code only.
 
  lrwxrwxrwx   1 root root  34 May  5 21:22 linux-2.6 -
  /lib/modules/2.6.9-1.667smp/source
 
  May  5 21:43:55 VERBOSE[12931]:  [chan_zap.so]May  5 21:43:55
  VERBOSE[12931]:  [chan_zap.so] = (Zapata Telephony)
  May  5 21:43:55 DEBUG[12931]: Parsing /etc/asterisk/zapata.conf
  May  5 21:43:55 WARNING[12931]: Unable to specify channel 1: Device or
  resource busy
  May  5 21:43:55 ERROR[12931]: Unable to open channel 1: Device or resource 
  busy
  here = 0, tmp-channel = 1, channel = 1
  May  5 21:43:55 ERROR[12931]: Unable to register channel '1'
  May  5 21:43:55 WARNING[12931]: chan_zap.so: load_module failed, returning 
  -1
  May  5 21:43:55 DEBUG[12931]: Unregistering channel type 'Zap'
  May  5 21:43:55 VERBOSE[12931]:   == Unregistered channel type 'Zap'
  May  5 21:43:55 WARNING[12931]: Loading module chan_zap.so failed!
 
 You need to edit zapata.conf. It evidently has a channel = 1 directive
 somewhere, but if you're using ztdummy I assume you have no zaptel hardware.
 
 There should only be channel directives for hardware that exists, and
 if you DO have zaptel hardware, you don't need ztdummy.
 
 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] IAX service provider with account balance announcement

2005-05-07 Thread C W Nel

Can anyone tell me if any IAX service provider
 supply audible minutes left/account balance announcement?

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 06/05/2005
 

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Re: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread Rich Adamson

 I started out happy as a clam with my new Broadvoice account and 
 asterisk machine.  About 10 days ago things began to change.  Inbound 
 calling has been down for 2 days.  Beyond the We are currently 
 experiencing in-bound call issues with a carrier partner in some areas. 
 We are aware of the issue and our engineers are working to have it 
 resolved as soon as possible mantra their support email and people that 
 answer the phone (if you can wait long enough) aren't talking.
 
 Who's happy with their voip service using asterisk?
 Where do you get reliable DIDs? 
 The 'carrier partner' they speak of.. can you get the did directly from 
 them?
 Are all the voip providers this flakey?

Nope, just some. Been very happy with both livevoip.com and teliax.com,
but there certainly are others as well. You've seen the testimonials
already on this list.



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Re: [Asterisk-Users] Re: HINT

2005-05-07 Thread Julian J. M.

But that only works when SIP/201 receives a call, right?

What if SIP/201 is making a dialout call, does it show as busy in the
phone's keypad?

Julian J. M.

On 5/7/05, Thorben Jensen [EMAIL PROTECTED] wrote:
  Could you please give us some more detail as to what you did, in terms of
  configuring the hint, and specifically what changes in the behavior of the
  running server-phone interaction as a result?
 
 You need to set the hint for the phone when the phone is being dialed like
 this:
 
 exten = 201,hint,SIP/201
 exten = 201,1,macro(dial-sip,201)
 
 It's important that you write the full name of the phone SIP/201 as you
 can't use substitutions like this SIP/${EXTEN} - it took me a long time to
 figure that out.
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RE: [Asterisk-Users] Re: HINT

2005-05-07 Thread Armin Lediger (HotZone GmbH)

Yes it does.

Armin

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Julian J. M.
 Sent: Saturday, May 07, 2005 1:48 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Re: HINT

 But that only works when SIP/201 receives a call, right?

 What if SIP/201 is making a dialout call, does it show as 
 busy in the phone's keypad?

 Julian J. M.

 On 5/7/05, Thorben Jensen [EMAIL PROTECTED] wrote:
   Could you please give us some more detail as to what you did, in 
   terms of configuring the hint, and specifically what 
 changes in the 
   behavior of the running server-phone interaction as a result?

  You need to set the hint for the phone when the phone is 
 being dialed 
  like
  this:

  exten = 201,hint,SIP/201
  exten = 201,1,macro(dial-sip,201)

  It's important that you write the full name of the phone 
 SIP/201 as 
  you can't use substitutions like this SIP/${EXTEN} - it 
 took me a long 
  time to figure that out.
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Re: [Asterisk-Users] Who's happy with their voip service?

On May 7, 2005 12:22 am, JD wrote:
 Who's happy with their voip service using asterisk?

I am.  Nufone.  For the past 18 months.  Totally happy.

 Where do you get reliable DIDs?

I have a PRI I get my DIDs on.  I have not yet found a VOIP provider with DIDs 
available in a WIDE area with reliable inbound service.  Nufone has DIDs in a 
couple NPA/NXXes but not in ones I need.

-A.
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Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

On May 7, 2005 05:16 am, Eric Wieling aka ManxPower wrote:
 Watch the console and be amazed when _. matches extension h, which is
 called when the far side of the call hangs up.  You get two calls to the
 same number by only dialing once!  Stop being lazy and at least use _X.
 as your pattern.

I'd suggest stopping being even that lazy and making some realistic matching 
extensions.  I have 

NXX for my 7-digit local
1NXXNXX for my 10-digit LD
0NXXNXX for operator-assisted LD
011. for international

I have never understood why people are so goddamned lazy that they can't write 
out a few exten matches...  but no they want to over-optimize and 
use _. or even _X.

If you want to match phone extensions then match 2XX or 2XXX or whatever your 
pattern happens to be.  It's almost NEVER necessary to use such broad matches 
as _. and _X.

-A.
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Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

Andrew Kohlsmith wrote:
On May 7, 2005 05:16 am, Eric Wieling aka ManxPower wrote:
Watch the console and be amazed when _. matches extension h, which is
called when the far side of the call hangs up.  You get two calls to the
same number by only dialing once!  Stop being lazy and at least use _X.
as your pattern.

I'd suggest stopping being even that lazy and making some realistic matching 
extensions.  I have 

NXX for my 7-digit local
1NXXNXX for my 10-digit LD
0NXXNXX for operator-assisted LD
011. for international
I have never understood why people are so goddamned lazy that they can't write 
out a few exten matches...  but no they want to over-optimize and 
use _. or even _X.

If you want to match phone extensions then match 2XX or 2XXX or whatever your 
pattern happens to be.  It's almost NEVER necessary to use such broad matches 
as _. and _X.
In much of the world the PSTN dialplan is not that simple.  Yes, a more 
specific dialplan than my _X. exmaple is a good idea, but the USA has a 
VERY simple PSTN dialplan and is NOT like most of the world.
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Re: [Asterisk-Users] Problem with Manager Originate and SIP extension

2005-05-07 Thread Terje Elde

Terje Elde wrote:
snip
[m197]
type=friend
username=m197
secret=
qualify=200
nat=yes
host=dynamic
canreinvite=no
context=from-sip

qualify=200, when the server is in the US, and my phone is in Norway, 
might not have been the best idea.

Problem solved.
Terje
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Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

On May 7, 2005 08:29 am, Eric Wieling aka ManxPower wrote:
 In much of the world the PSTN dialplan is not that simple.  Yes, a more
 specific dialplan than my _X. exmaple is a good idea, but the USA has a
 VERY simple PSTN dialplan and is NOT like most of the world.

Perhaps I am naive but I don't think that diaplans would be that much more 
complex if people matched more accurately at all.  Granted most of my calling 
is north american, but there's some south america and germany in there as 
well, along with a sprinkle of UK.  None of them would require _X. to match 
the vast majority of the numbers in those locations.

I would, however, enjoy it if some people could give examples of screwed up 
PSTN dialing matching around the world.  I'd enjoy the challenge of creating 
tightly-matching dialplans for them.

-A.
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Re: [Asterisk-Users] IAXy Firmware Upgrade

2005-05-07 Thread Time Bandit

 I'd like to known what I have to do to upgrade
 the firmware into a IAXy device.
It does it automagically when it connect to Asterisk if a newer
version is available.
Look in /var/lib/asterisk/firmware/iax and you will see iaxy.bin.

hth
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[Asterisk-Users] Re: Re: HINT

2005-05-07 Thread Thorben Jensen

You need to put that in whether SIP/201 is recieving or making a call.

This only work for SIP/201 - you will need to do the same for every phone 
you have.

thorben


Julian J. M. [EMAIL PROTECTED] skrev i en meddelelse 
news:[EMAIL PROTECTED]
But that only works when SIP/201 receives a call, right?

What if SIP/201 is making a dialout call, does it show as busy in the
phone's keypad?

Julian J. M.

On 5/7/05, Thorben Jensen [EMAIL PROTECTED] wrote:
  Could you please give us some more detail as to what you did, in terms 
  of
  configuring the hint, and specifically what changes in the behavior of 
  the
  running server-phone interaction as a result?

 You need to set the hint for the phone when the phone is being dialed like
 this:

 exten = 201,hint,SIP/201
 exten = 201,1,macro(dial-sip,201)

 It's important that you write the full name of the phone SIP/201 as you
 can't use substitutions like this SIP/${EXTEN} - it took me a long time to
 figure that out.
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RE: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread Andre Normandin

I've had Broadvoice for over a year now, and although their outages are
really annoying, the fact that their service costs $20/month unlimited is
what keeps me with them..

I have 2 Inbound #'s through them (same account), one in GA (678-253) and
one in CT (203-935), and overall their inbound has been more reliable than
their outbound (minus the past week or so)..

I have my dialplan try BV first, and then if it cannot use BV for outbound,
it rolls to my pots line(s).. It actually works really well, except that if
BV goes completely toes up Asterisk decides that it doesn't want to do
anything either :-(

That is what I find the most annoying, quite frankly, BV is having Growing
pains (in my opinion), and I can accept that, haven't put anything critical
on my BV inbound, and 90% of the time BV outbound works fine.. The rest of
the time, the pots take care of outbound, and anyone who calls me calls on
my pots lines (except for family in GA, which is why I have the GA #).

For me personally, I just think VOIP is 'too' early in the maturity curve to
really rely on it as a provider.. It's great in-house (medium/large
companies), but for service, I think pots are the way to have rock solid
service for the time being.

I know of two of my friends that have Vonage as their only inbound numbers
(not via asterisk, via the vonage locked adapters, so it is completely
vonange), and their service also has issues at times.. Granted, I'm not sure
if it's a true vonage issue, or their internet connection, but nonetheless,
there are still issues..

If I could get Asterisk so it just work continue 'working' properly with
whatever SIP connections it can reach, I'd be a happy man..  Don't get me
wrong, I think Broadvoice needs to communicate better with their customer
base, and the latest ongoing outage is, to say the least, very frustrating,
but I am willing to cut them some slack because I think VOIP is still in
it's infancy, and broadvoice is the only BYOD provider I know that will give
unlimited for $20.00/month..

 - Andre

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Johnathan
Corgan
Sent: Saturday, May 07, 2005 12:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Who's happy with their voip service?


JD wrote:

 Inbound
 calling has been down for 2 days.

Just FYI, mine is back up (408-903) as of about five hours ago.

I did just speak with a (Broadvoice) support tech on an entirely
unrelated matter (40 min. hold time!), mentioned mine was working, and
he seemed to think things were coming back in stages.

I've had them for two months now.  People may recall a series of emails
regarding packet loss through their PNAP link to Sprintlink (my ex-ISP
backbone.)  I ditched the Sprint BBD fixed-wireless service, got
Sonic.net DSL, and have been enjoying pretty high quality voice service
since.  The packet loss rates at PNAP still show but I think now this
shows it's an artificial measure (intentionally dropped non-VOIP
packets, all the other potential reasons hashed about in that thread.)

In spite of the service outages and long hold times for support, I still
want to give them the benefit of the doubt (and my $25 monthly.)  It
still seems like growing pains vs. incompetency.

I tried their web interface to change DIDs, as they now have them in my
home area code.  The effect was instant, I reconfigured sip.conf with
the new number and secret they provide, and something like 3 minutes
later was using the new DID.  So some things do work well.

Wish they did IAX. And ILBC. Not that important to me right now, though.

-Johnathan
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[Asterisk-Users] Termination South America

2005-05-07 Thread JOAO CARLOS MOURA





We offer termination in:

Miami,USA u$s 0.019 
Buenos Aires,ARGENTINA u$s 0.019
Fortaleza,BRAZIL u$s 0.029

Check our rates in CHILE Santiago, PARAGUAY 
Asuncion, URUGUAY Montevideo, Punta del Este, BRAZIL Rio de Janeiro, San Pablo, 
Goiania, Puerto Alegre.

DID's u$s 5.50 each in all ours areas.


Joao Carlos Moura
[EMAIL PROTECTED]
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[Asterisk-Users] Inexpensive FAX and 800 Number retail service

2005-05-07 Thread lonnie

Greetings All,

I have a number of projects in the works at the moment and for one of
them, I need to locate an inexpensive and reliable service that can
provide small-office virtual services:

1. FAX to Email
2. Toll Free number with voicemail boxes for Tech Support, Billing
Inquiries, Customer Service, Abuse Reporting, etc...

I have been looking all over the Internet and there seem to be a LOT but I
am at a loss as to which are reliable and cost effective as I have see
rates ranging from $4.95/mo to $29.95/mo.

Being this, I thought that I would ask the experts, which are you guys on
the Asterisk mailing list.

Any help would be greatly appreciated,

Have a great day,
Lonnie


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[Asterisk-Users] end user gui

2005-05-07 Thread Jim Sturtevant









Ive reviewed the wiki and other resources and
havent been able to locate a tool which would allow an end user to make
changes to their service. The features and end-user might want to change
is fairly limited (call fwd, number of rings, etc). This might
require real-time. Thanks in advance.








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RE: [Asterisk-Users] Inexpensive FAX and 800 Number retail service

2005-05-07 Thread Jim Sturtevant

I've been using www.maxemail.com for quite awhile and they provide great
service.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, May 07, 2005 8:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Inexpensive FAX and 800 Number retail service

Greetings All,

I have a number of projects in the works at the moment and for one of
them, I need to locate an inexpensive and reliable service that can
provide small-office virtual services:

1. FAX to Email
2. Toll Free number with voicemail boxes for Tech Support, Billing
Inquiries, Customer Service, Abuse Reporting, etc...

I have been looking all over the Internet and there seem to be a LOT but I
am at a loss as to which are reliable and cost effective as I have see
rates ranging from $4.95/mo to $29.95/mo.

Being this, I thought that I would ask the experts, which are you guys on
the Asterisk mailing list.

Any help would be greatly appreciated,

Have a great day,
Lonnie


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RE: [Asterisk-Users] Inexpensive FAX and 800 Number retail service

2005-05-07 Thread lonnie

Thanks,

I'll look into that one as well.

 I've been using www.maxemail.com for quite awhile and they provide great
 service.



 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 [EMAIL PROTECTED]
 Sent: Saturday, May 07, 2005 8:52 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Inexpensive FAX and 800 Number retail service

 Greetings All,

 I have a number of projects in the works at the moment and for one of
 them, I need to locate an inexpensive and reliable service that can
 provide small-office virtual services:

 1. FAX to Email
 2. Toll Free number with voicemail boxes for Tech Support, Billing
 Inquiries, Customer Service, Abuse Reporting, etc...

 I have been looking all over the Internet and there seem to be a LOT but I
 am at a loss as to which are reliable and cost effective as I have see
 rates ranging from $4.95/mo to $29.95/mo.

 Being this, I thought that I would ask the experts, which are you guys on
 the Asterisk mailing list.

 Any help would be greatly appreciated,

 Have a great day,
 Lonnie


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RE: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread Chris Coulthurst

I tend to agree about the in-house being the 'stable part'.  Like
anything else on the internet, if you don't have control of all parts
(trunks and phones and dialplans), there are bound to be issues with
uptime, and how your equipment responds to 'their' downtime.  It reminds
me of the headaches I had as an ISP when a BGP4 route wouldn't switch to
the redundant carrier, because the main carrier didn't really die, it
just stopped transmitting!

It's also worth noting the design flaws with IPv4 handling priority
packets in the first place.  I think most of the little 'gotchas' in
VoIP would magically vanish if QoS was something that could be depended
upon.  All you need is one router to not know how to pass the qos token,
and now you don't really have any!  Its another example hiw an in-house
system can be stable when you hold all the cards.

By the time IPv6 gets here, it will be amazingly obsolete...

Chris Coulthurst
[EMAIL PROTECTED]
 

|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Andre Normandin
|Sent: Saturday, May 07, 2005 8:00 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] Who's happy with their voip service?
|
|I've had Broadvoice for over a year now, and although their outages are
|really annoying, the fact that their service costs $20/month unlimited
is
|what keeps me with them..
|
|I have 2 Inbound #'s through them (same account), one in GA (678-253)
and
|one in CT (203-935), and overall their inbound has been more reliable
than
|their outbound (minus the past week or so)..
|
|I have my dialplan try BV first, and then if it cannot use BV for
outbound,
|it rolls to my pots line(s).. It actually works really well, except
that if
|BV goes completely toes up Asterisk decides that it doesn't want to do
|anything either :-(
|
|That is what I find the most annoying, quite frankly, BV is having
Growing
|pains (in my opinion), and I can accept that, haven't put anything
critical
|on my BV inbound, and 90% of the time BV outbound works fine.. The rest
of
|the time, the pots take care of outbound, and anyone who calls me calls
on
|my pots lines (except for family in GA, which is why I have the GA #).
|
|For me personally, I just think VOIP is 'too' early in the maturity
curve
|to
|really rely on it as a provider.. It's great in-house (medium/large
|companies), but for service, I think pots are the way to have rock
solid
|service for the time being.
|
|I know of two of my friends that have Vonage as their only inbound
numbers
|(not via asterisk, via the vonage locked adapters, so it is completely
|vonange), and their service also has issues at times.. Granted, I'm not
|sure
|if it's a true vonage issue, or their internet connection, but
nonetheless,
|there are still issues..
|
|If I could get Asterisk so it just work continue 'working' properly
with
|whatever SIP connections it can reach, I'd be a happy man..  Don't get
me
|wrong, I think Broadvoice needs to communicate better with their
customer
|base, and the latest ongoing outage is, to say the least, very
frustrating,
|but I am willing to cut them some slack because I think VOIP is still
in
|it's infancy, and broadvoice is the only BYOD provider I know that will
|give
|unlimited for $20.00/month..
|
| - Andre
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] Behalf Of Johnathan
|Corgan
|Sent: Saturday, May 07, 2005 12:58 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Who's happy with their voip service?
|
|
|JD wrote:
|
| Inbound
| calling has been down for 2 days.
|
|Just FYI, mine is back up (408-903) as of about five hours ago.
|
|I did just speak with a (Broadvoice) support tech on an entirely
|unrelated matter (40 min. hold time!), mentioned mine was working, and
|he seemed to think things were coming back in stages.
|
|I've had them for two months now.  People may recall a series of emails
|regarding packet loss through their PNAP link to Sprintlink (my ex-ISP
|backbone.)  I ditched the Sprint BBD fixed-wireless service, got
|Sonic.net DSL, and have been enjoying pretty high quality voice service
|since.  The packet loss rates at PNAP still show but I think now this
|shows it's an artificial measure (intentionally dropped non-VOIP
|packets, all the other potential reasons hashed about in that thread.)
|
|In spite of the service outages and long hold times for support, I
still
|want to give them the benefit of the doubt (and my $25 monthly.)  It
|still seems like growing pains vs. incompetency.
|
|I tried their web interface to change DIDs, as they now have them in my
|home area code.  The effect was instant, I reconfigured sip.conf with
|the new number and secret they provide, and something like 3 minutes
|later was using the new DID.  So some things do work well.
|
|Wish they did IAX. And ILBC. Not that important to me right now,
though.
|
|-Johnathan

RE: [Asterisk-Users] Re: Re: HINT

Can you post a full dialplan example... 

Also, will this only work for certain phones and atas also? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Thorben Jensen
|Sent: Sábado, 07 de Mayo de 2005 09:45 a.m.
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Re: Re: HINT
|
|You need to put that in whether SIP/201 is recieving or making a call.
|
|This only work for SIP/201 - you will need to do the same for 
|every phone you have.
|
|thorben
|
|
|Julian J. M. [EMAIL PROTECTED] skrev i en meddelelse 
|news:[EMAIL PROTECTED]
|But that only works when SIP/201 receives a call, right?
|
|What if SIP/201 is making a dialout call, does it show as busy 
|in the phone's keypad?
|
|Julian J. M.
|
|On 5/7/05, Thorben Jensen [EMAIL PROTECTED] wrote:
|  Could you please give us some more detail as to what you did, in 
|  terms of configuring the hint, and specifically what 
|changes in the 
|  behavior of the running server-phone interaction as a result?
|
| You need to set the hint for the phone when the phone is 
|being dialed 
| like
| this:
|
| exten = 201,hint,SIP/201
| exten = 201,1,macro(dial-sip,201)
|
| It's important that you write the full name of the phone 
|SIP/201 as 
| you can't use substitutions like this SIP/${EXTEN} - it took 
|me a long 
| time to figure that out.
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|
|
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|

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Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

Andrew Kohlsmith wrote:
On May 7, 2005 08:29 am, Eric Wieling aka ManxPower wrote:
In much of the world the PSTN dialplan is not that simple.  Yes, a more
specific dialplan than my _X. exmaple is a good idea, but the USA has a
VERY simple PSTN dialplan and is NOT like most of the world.

Perhaps I am naive but I don't think that diaplans would be that much more 
complex if people matched more accurately at all.  Granted most of my calling 
is north american, but there's some south america and germany in there as 
well, along with a sprinkle of UK.  None of them would require _X. to match 
the vast majority of the numbers in those locations.

I would, however, enjoy it if some people could give examples of screwed up 
PSTN dialing matching around the world.  I'd enjoy the challenge of creating 
tightly-matching dialplans for them.
Area Code:2-5 digits
Subscriber Number:3-9 digits (see note below)
Trunk Prefix: 0
International Prefix: 00
http://www.wtng.info/wtng-49-de.html
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Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

Andrew Kohlsmith wrote:
On May 7, 2005 08:29 am, Eric Wieling aka ManxPower wrote:
In much of the world the PSTN dialplan is not that simple.  Yes, a more
specific dialplan than my _X. exmaple is a good idea, but the USA has a
VERY simple PSTN dialplan and is NOT like most of the world.

Perhaps I am naive but I don't think that diaplans would be that much more 
complex if people matched more accurately at all.  Granted most of my calling 
is north american, but there's some south america and germany in there as 
well, along with a sprinkle of UK.  None of them would require _X. to match 
the vast majority of the numbers in those locations.

I would, however, enjoy it if some people could give examples of screwed up 
PSTN dialing matching around the world.  I'd enjoy the challenge of creating 
tightly-matching dialplans for them.
Number Format - Luxembourg
Area Code:none
Subscriber Number:5-11 digits
Trunk Prefix: n/a
International Prefix: 00
http://www.wtng.info/wtng-352-lu.html

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Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

Andrew Kohlsmith wrote:
On May 7, 2005 08:29 am, Eric Wieling aka ManxPower wrote:
In much of the world the PSTN dialplan is not that simple.  Yes, a more
specific dialplan than my _X. exmaple is a good idea, but the USA has a
VERY simple PSTN dialplan and is NOT like most of the world.

Perhaps I am naive but I don't think that diaplans would be that much more 
complex if people matched more accurately at all.  Granted most of my calling 
is north american, but there's some south america and germany in there as 
well, along with a sprinkle of UK.  None of them would require _X. to match 
the vast majority of the numbers in those locations.

I would, however, enjoy it if some people could give examples of screwed up 
PSTN dialing matching around the world.  I'd enjoy the challenge of creating 
tightly-matching dialplans for them.
Number Format - Ireland (this one is great)
Area Code:1-3 digits
Subscriber Number:5-7 digits
Trunk Prefix: 0
International Prefix: 00
http://www.wtng.info/wtng-353-ie.html
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RE: [Asterisk-Users] Good NAT Pnp Hardphone

 For example,
 I want to give a phone to my brother, who is going to europe.  His ICH
 softphone is fine there.  Both the poly and cisco though require you to
 setup for nat.  He would not be able to set this up though, so I want to
 just give him a preconfig'ed phone and plug and go...

My experience with the Sipura SPA-841 is that, as long as you set up the
STUN server and say Yes to all the options under the NAT support
parameters, it works pretty much plug-n-play. I've tried it even behind
three cascaded NAT routers and it works.

--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990


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RE: [Asterisk-Users] Inexpensive FAX and 800 Number retail service

 1. FAX to Email

Check out TrustFax (http://tinyurl.com/8png8). $10/year for a toll-free fax
number and $0.10 per page in/out.

--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990

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Re: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread Michael D Schelin

Isn't amazing what has happened in the last five or six years with the 
Internet.  There is no design flaw with IPv4. It was created back when 
you were in diapers and with todays pda's having more power than the 
systems back then.  An industry protocol that is going strong 30 or more 
years is amazing in it's own right. How could they see the future of 
what we are doing today with the protocol.  I believe the engineers who 
designed IPv4 were brilliant men and did a great job designing  
something that is computer system neutral. 

Again in the the last few years VoIP has come a long way as the PSTN has 
had over 100 years to perfect theirs. If we did not have to interface 
with the PSTN don't you think we would be better off?  They didn't have 
to interface with anybody else.

Chris Coulthurst wrote:
I tend to agree about the in-house being the 'stable part'.  Like
anything else on the internet, if you don't have control of all parts
(trunks and phones and dialplans), there are bound to be issues with
uptime, and how your equipment responds to 'their' downtime.  It reminds
me of the headaches I had as an ISP when a BGP4 route wouldn't switch to
the redundant carrier, because the main carrier didn't really die, it
just stopped transmitting!
It's also worth noting the design flaws with IPv4 handling priority
packets in the first place.  I think most of the little 'gotchas' in
VoIP would magically vanish if QoS was something that could be depended
upon.  All you need is one router to not know how to pass the qos token,
and now you don't really have any!  Its another example hiw an in-house
system can be stable when you hold all the cards.
By the time IPv6 gets here, it will be amazingly obsolete...
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Andre Normandin
|Sent: Saturday, May 07, 2005 8:00 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] Who's happy with their voip service?
|
|I've had Broadvoice for over a year now, and although their outages are
|really annoying, the fact that their service costs $20/month unlimited
is
|what keeps me with them..
|
|I have 2 Inbound #'s through them (same account), one in GA (678-253)
and
|one in CT (203-935), and overall their inbound has been more reliable
than
|their outbound (minus the past week or so)..
|
|I have my dialplan try BV first, and then if it cannot use BV for
outbound,
|it rolls to my pots line(s).. It actually works really well, except
that if
|BV goes completely toes up Asterisk decides that it doesn't want to do
|anything either :-(
|
|That is what I find the most annoying, quite frankly, BV is having
Growing
|pains (in my opinion), and I can accept that, haven't put anything
critical
|on my BV inbound, and 90% of the time BV outbound works fine.. The rest
of
|the time, the pots take care of outbound, and anyone who calls me calls
on
|my pots lines (except for family in GA, which is why I have the GA #).
|
|For me personally, I just think VOIP is 'too' early in the maturity
curve
|to
|really rely on it as a provider.. It's great in-house (medium/large
|companies), but for service, I think pots are the way to have rock
solid
|service for the time being.
|
|I know of two of my friends that have Vonage as their only inbound
numbers
|(not via asterisk, via the vonage locked adapters, so it is completely
|vonange), and their service also has issues at times.. Granted, I'm not
|sure
|if it's a true vonage issue, or their internet connection, but
nonetheless,
|there are still issues..
|
|If I could get Asterisk so it just work continue 'working' properly
with
|whatever SIP connections it can reach, I'd be a happy man..  Don't get
me
|wrong, I think Broadvoice needs to communicate better with their
customer
|base, and the latest ongoing outage is, to say the least, very
frustrating,
|but I am willing to cut them some slack because I think VOIP is still
in
|it's infancy, and broadvoice is the only BYOD provider I know that will
|give
|unlimited for $20.00/month..
|
| - Andre
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] Behalf Of Johnathan
|Corgan
|Sent: Saturday, May 07, 2005 12:58 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Who's happy with their voip service?
|
|
|JD wrote:
|
| Inbound
| calling has been down for 2 days.
|
|Just FYI, mine is back up (408-903) as of about five hours ago.
|
|I did just speak with a (Broadvoice) support tech on an entirely
|unrelated matter (40 min. hold time!), mentioned mine was working, and
|he seemed to think things were coming back in stages.
|
|I've had them for two months now.  People may recall a series of emails
|regarding packet loss through their PNAP link to Sprintlink (my ex-ISP
|backbone.)  I ditched the Sprint BBD fixed-wireless service, got
|Sonic.net DSL, and have been enjoying

[Asterisk-Users] h323.conf - Asterisk not routing incoming calls based on IP - Ignoring type=user + host= + context=

2005-05-07 Thread Deon

Ok, at the bottom of my h323.conf file on my 1st server I have this:

; -
[test]
type=user
host=209.237.227.185
context=termination-test
incominglimit=10
accountcode=005
; -

Using an Asterisk at the other IP, I have this:
exten = _1NXXNXX,1,Dial(H323/[EMAIL PROTECTED],,o)

This should send a call from the test-server to the IP of the 1st server;
the server should see the call as coming from the host IP of [test] and
send it to the termination-test context, not the default incoming
context, however I see this on the 1st server:

=
  == Starting H323/ip$209.237.227.185:46373/20161 at
incoming,18005551212,1 failed so falling back to exten 's'
  == Starting H323/ip$209.237.227.185:46373/20161 at incoming,s,1 still
failed so falling back to context 'default'
May  7 14:19:18 WARNING[30649]: pbx.c:1889 ast_pbx_run: Channel
'H323/ip$209.237.227.185:46373/20161' sent into invalid extension 's' in
context 'default', but no invalid handler
=

The remote-test server sends the h323 call perfectly, but the receiving
server isn't matching and sending it to the propper context. Any idea why?
My [test] user is taken right out of the example h323.conf file
(billybob).

Best Regards,
Deon



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[Asterisk-Users] cant connect

2005-05-07 Thread Bernie

I just installed Asterisk @home 1.0 and its up and running, i added an 
extension with the web interface, but now when I try to connect with a 
sip client (x-lite) it just times out.  here is the log from x-lite 
below.  Is there any way to view a log on the asterisk side to see whats 
going on? or anybody have an idea as to why its not working?

thanks
SEND TIME: 19038812
SEND  170.20.41.165:5060
REGISTER sip:170.20.41.165 SIP/2.0
Via: SIP/2.0/UDP 
170.20.11.59:5060;rport;branch=z9hG4bKD170D188076045BE8656D59545FF9678
From: Bernie sip:[EMAIL PROTECTED];tag=2472129214
To: Bernie sip:[EMAIL PROTECTED]
Contact: Bernie sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 49490 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0

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[Asterisk-Users] two questions about the Sipura 841?

2005-05-07 Thread Joel Duffield

Ok my first question is I have seen messages about a patch for asterisk so
that I can do auto answer on these phones. I found the message in the
archives but I do not have that message as an email still, so I do not have
the attachment. Can anyone tell me where to get it? Also on this phone how
can I set the phone to release the line sooner without playing the anoying
beeping for 5 seconds, I can change how long until the beeping starts but
how do I shorten the beeps?

Thanks

Joel
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No virus found in this outgoing message.
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Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005

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RE: [Asterisk-Users] two questions about the Sipura 841?

2005-05-07 Thread Jim Sturtevant

What is the purpose of the beeping?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Duffield
Sent: Saturday, May 07, 2005 12:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] two questions about the Sipura 841?

Ok my first question is I have seen messages about a patch for asterisk so
that I can do auto answer on these phones. I found the message in the
archives but I do not have that message as an email still, so I do not have
the attachment. Can anyone tell me where to get it? Also on this phone how
can I set the phone to release the line sooner without playing the anoying
beeping for 5 seconds, I can change how long until the beeping starts but
how do I shorten the beeps?

Thanks

Joel
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005

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[Asterisk-Users] WIP-5000 and DTMF

2005-05-07 Thread Jim Meehan

My WIP-5000 phone is working well with my Asterisk box now, except for DTMF.
All DTMF key presses come across as clipped or just clicks on the remote side.

I had this problem with my Sipura ATA as well, but fixed that by playing with
the settings on the Sipura device.

I've tried dtmfmode=inband and also rfc2833, but neither seem to work.  I 
don't see any place in the settings on the WIP-5000 to change its DTMF mode.

Anyone have DTMF working with their WIP-5000 phone?

Jim Meehan
Oakland, CA

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[Asterisk-Users] WIP-5000 and DTMF

2005-05-07 Thread Jerry Geis

Jim,
My (3) WIP -5000 phones work just fine with DTMF.
I setup the user.ini file to the following and of course the same in the
sip.conf.
Hope this helps. The OpenSip is also in the browser config for the phone...
[OpenSip]
*T1 = 500
*T2 = 4000
; DTMFType - 0 RTP
; DTMFType - 1 INFO
; DTMFType - 2 rfc2833
*DTMFType = 2
Jerry
-
My WIP-5000 phone is working well with my Asterisk box now, except for DTMF.
All DTMF key presses come across as clipped or just clicks on the remote side.
I had this problem with my Sipura ATA as well, but fixed that by playing with
the settings on the Sipura device.
I've tried dtmfmode=inband and also rfc2833, but neither seem to work.  I 
don't see any place in the settings on the WIP-5000 to change its DTMF mode.

Anyone have DTMF working with their WIP-5000 phone?
Jim Meehan
Oakland, CA

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[Asterisk-Users] Polyco ip600 incoming ring time

Hello All,
Does anyone know how to reduce the incoming ringtime on the polycoms?

What I mean is, When I have an incoming call, my 7960 and pingtel ring
immediately, but the polycom seems to delay 2 seconds before ringing...

Any ideas?

Greg
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RE: [Asterisk-Users] two questions about the Sipura 841?

2005-05-07 Thread Joel Duffield

The beeping is to tell you that the remote end has hungup, im sorry I don't
know the technical term for it but it happens on your regular home phone if
the other end was to hang up and you did not hang up your receiver. the web
interface calls it the Reorder.

Thanks

Joel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jim Sturtevant
Sent: Saturday, May 07, 2005 4:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] two questions about the Sipura 841?


What is the purpose of the beeping?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Duffield
Sent: Saturday, May 07, 2005 12:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] two questions about the Sipura 841?

Ok my first question is I have seen messages about a patch for asterisk so
that I can do auto answer on these phones. I found the message in the
archives but I do not have that message as an email still, so I do not have
the attachment. Can anyone tell me where to get it? Also on this phone how
can I set the phone to release the line sooner without playing the anoying
beeping for 5 seconds, I can change how long until the beeping starts but
how do I shorten the beeps?

Thanks

Joel
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005

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Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005

--
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Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005

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[Asterisk-Users] Cisco ATA Call Waiting

2005-05-07 Thread Christopher Kenna



I currently have 2 Cisco 7960's and 2 ATA 186's connected to asterisk. The 7960's work just fine for call waiting, but the ATA's dont. I cant seem to get the ATA's to use the call waiting feature, the calls just go straight to voicemail instead of prompting with the usual tone.

Please help
Chris

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RE: [Asterisk-Users] CallerID name lookup AGI script

Jim:

I modified your script to first look up Google and then look up 411.com.
It's better for me, because 411.com has Canadian listings too. I still left
Google in because it's much faster and if it has information, I'd rather use
that. I removed the area code thing because it's no use to me. I also
removed the reversing first/last names thing that you did.

Here's the code:

#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$number = $ARGV[0];
$found = 0;

open(RESULTS, /usr/bin/curl -s -m 2 -A Mozilla/4.0
http://www.google.com/search?q=phonebook:$number |);
while (RESULTS) {
  if (m/Residential Phonebook/ || m/Business Phonebook/) {
$found = 1;
@fields = split(//);
@result = split(/-/, $fields[35]);
chop($result[0]);
$name = $result[0];
  }
  if (m/did not match any/) {
$found = 0;
  }
}

if ($found == 0) {
  open(RESULTS, /usr/bin/curl -s -m 2 -A Mozilla/4.0
http://www.411.com/10668/search/Reverse_Phone?phone=$number |);
  while (RESULTS) {
if (m/__FIRST/) {
  $found = 1;
  @fname = split(/\/);
}
elsif (m/__LAST/) {
  @lname = split(/\/);
  $name = $fname[1] .   .$lname[1];
}
  }
}

$AGI-set_variable('googlename', \$name\);

--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jim Meehan
 Sent: April 9, 2005 3:05 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] CallerID name lookup AGI script
 
 Hi all,
 
 My VoIP provider (race.com) doesn't send name info with CallerID, so I
 wrote
 an AGI script that does the following:
 
 1) If it's a toll free number (800|888|877|866), set the CallerID name to
 TollFree Caller
 2) Use curl to look up the number in Google phonebook
 3) If a business listing, set the CallerID name to business name, as is.
 4) If it's a residential listing, reverse the listing so it's last name
 first,
 then set the CallerID name to that.
 5) If there's no match in Google phonebook, look up the NPA/NXX on
 www.areacodedownload.com and set the CallerID name to @ST RATECENTER
 where
 ST is the two-letter state abbreviation, and RATECENTER is the name of
 telco rate center in that state.
 
 Thought some of you might find this AGI script useful, so I'm including it
 below.  It requires the Asterisk::AGI perl module.
 
 There are other reverse phone lookup sources that are more complete than
 Google's, but they are harder to screen scrape.  Also, I probably could
 have
 made this a little cleaner if I used the Google API rather than screen
 scraping with curl/perl.  Please feel free to take a shot at making any of
 those modifications.
 
 Here's a snippet from my extensions.conf where it gets called:
 
 exten = s,1,AGI(callerid.agi|${CALLERIDNUM})
 exten = s,2,SetCallerId,${googlename} ${CALLERIDNUM}
 exten = s,3,Dial(${PHONES},30,r)
 exten = s,4,Answer
 exten = s,5,Wait(2)
 exten = s,6,Voicemail(u3001)
 exten = s,7,Hangup
 
 
 And here's the script:
 
 #!/usr/bin/perl
 
 use Asterisk::AGI;
 
 $AGI = new Asterisk::AGI;
 
 $number = $ARGV[0];
 
 if ($number =~ m/(800|888|877|866)\d{7}/) {
   $AGI-set_variable('googlename', \TollFree Caller\);
   exit 0;
 }
 
 open(RESULTS, /usr/bin/curl -s -m 2 -A Mozilla/4.0
 http://www.google.com/search
 ?q=phonebook:$number |);
 
 while (RESULTS) {
   if (m/Residential Phonebook/) {
 $reverse = 1;
 @fields = split(//);
   }
   if (m/Business Phonebook/) {
 @fields = split(//);
   }
   if (m/did not match any/) {
 @digits = split(//, $number);
 $npa = $digits[0] . $digits[1] . $digits[2];
 $nxx = $digits[3] . $digits[4] . $digits[5];
 open(LOCATION, /usr/bin/curl -s -m 2 -A Mozilla/4.0
 http://www.areacodedown
 load.com/$npa/$nxx/ |);
 while (LOCATION) {
   if (m/State/) {
 $line = LOCATION;
 $line =~ m/\\#CACACA\\w* (\w\w)\/td/;
 $name = [EMAIL PROTECTED];
   }
   if (m/Rate Center/) {
 $line = LOCATION;
 $line =~ m/\\#CACACA\((\w|\s)*)\/td/;
 $name = $name .   . $1;
   }
 }
 $AGI-set_variable('googlename', \$name\);
 exit 0;
   }
 }
 
 @result = split(/-/, $fields[35]);
 chop($result[0]);
 if ($reverse) {
   @words = split(/ /, $result[0]);
   $last = pop(@words);
   unshift(@words, $last,);
   foreach $word (@words) {
 $name = $name . $word .  ;
   }
 }
 if ($reverse == 0) {
   $name = $result[0];
 }
 
 $AGI-set_variable('googlename', \$name\);
 
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RE: [Asterisk-Users] CallerID name lookup AGI script

BTW This does not do most business name lookups from 411.com correctly.
Maybe someone who actually knows Perl can do that :)

--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Nabeel Jafferali
 Sent: May 7, 2005 5:35 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] CallerID name lookup AGI script
 
 Jim:
 
 I modified your script to first look up Google and then look up 411.com.
 It's better for me, because 411.com has Canadian listings too. I still
 left
 Google in because it's much faster and if it has information, I'd rather
 use
 that. I removed the area code thing because it's no use to me. I also
 removed the reversing first/last names thing that you did.
 
 Here's the code:
 
 #!/usr/bin/perl
 use Asterisk::AGI;
 $AGI = new Asterisk::AGI;
 $number = $ARGV[0];
 $found = 0;
 
 open(RESULTS, /usr/bin/curl -s -m 2 -A Mozilla/4.0
 http://www.google.com/search?q=phonebook:$number |);
 while (RESULTS) {
   if (m/Residential Phonebook/ || m/Business Phonebook/) {
 $found = 1;
 @fields = split(//);
 @result = split(/-/, $fields[35]);
 chop($result[0]);
 $name = $result[0];
   }
   if (m/did not match any/) {
 $found = 0;
   }
 }
 
 if ($found == 0) {
   open(RESULTS, /usr/bin/curl -s -m 2 -A Mozilla/4.0
 http://www.411.com/10668/search/Reverse_Phone?phone=$number |);
   while (RESULTS) {
 if (m/__FIRST/) {
   $found = 1;
   @fname = split(/\/);
 }
 elsif (m/__LAST/) {
   @lname = split(/\/);
   $name = $fname[1] .   .$lname[1];
 }
   }
 }
 
 $AGI-set_variable('googlename', \$name\);
 
 --
 Nabeel Jafferali
 X2 Networks
 www.x2n.ca
 T: 1.647.722.6900
1.877.VOIP.X2N
 F: 1.866.655.6698
 FWD: 46990
 
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Jim Meehan
  Sent: April 9, 2005 3:05 PM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] CallerID name lookup AGI script
 
  Hi all,
 
  My VoIP provider (race.com) doesn't send name info with CallerID, so I
  wrote
  an AGI script that does the following:
 
  1) If it's a toll free number (800|888|877|866), set the CallerID name
 to
  TollFree Caller
  2) Use curl to look up the number in Google phonebook
  3) If a business listing, set the CallerID name to business name, as is.
  4) If it's a residential listing, reverse the listing so it's last name
  first,
  then set the CallerID name to that.
  5) If there's no match in Google phonebook, look up the NPA/NXX on
  www.areacodedownload.com and set the CallerID name to @ST RATECENTER
  where
  ST is the two-letter state abbreviation, and RATECENTER is the name
 of
  telco rate center in that state.
 
  Thought some of you might find this AGI script useful, so I'm including
 it
  below.  It requires the Asterisk::AGI perl module.
 
  There are other reverse phone lookup sources that are more complete than
  Google's, but they are harder to screen scrape.  Also, I probably could
  have
  made this a little cleaner if I used the Google API rather than screen
  scraping with curl/perl.  Please feel free to take a shot at making any
 of
  those modifications.
 
  Here's a snippet from my extensions.conf where it gets called:
 
  exten = s,1,AGI(callerid.agi|${CALLERIDNUM})
  exten = s,2,SetCallerId,${googlename} ${CALLERIDNUM}
  exten = s,3,Dial(${PHONES},30,r)
  exten = s,4,Answer
  exten = s,5,Wait(2)
  exten = s,6,Voicemail(u3001)
  exten = s,7,Hangup
 
 
  And here's the script:
 
  #!/usr/bin/perl
 
  use Asterisk::AGI;
 
  $AGI = new Asterisk::AGI;
 
  $number = $ARGV[0];
 
  if ($number =~ m/(800|888|877|866)\d{7}/) {
$AGI-set_variable('googlename', \TollFree Caller\);
exit 0;
  }
 
  open(RESULTS, /usr/bin/curl -s -m 2 -A Mozilla/4.0
  http://www.google.com/search
  ?q=phonebook:$number |);
 
  while (RESULTS) {
if (m/Residential Phonebook/) {
  $reverse = 1;
  @fields = split(//);
}
if (m/Business Phonebook/) {
  @fields = split(//);
}
if (m/did not match any/) {
  @digits = split(//, $number);
  $npa = $digits[0] . $digits[1] . $digits[2];
  $nxx = $digits[3] . $digits[4] . $digits[5];
  open(LOCATION, /usr/bin/curl -s -m 2 -A Mozilla/4.0
  http://www.areacodedown
  load.com/$npa/$nxx/ |);
  while (LOCATION) {
if (m/State/) {
  $line = LOCATION;
  $line =~ m/\\#CACACA\\w* (\w\w)\/td/;
  $name = [EMAIL PROTECTED];
}
if (m/Rate Center/) {
  $line = LOCATION;
  $line =~ m/\\#CACACA\((\w|\s)*)\/td/;
  $name = $name .   . $1;
}
  }
  $AGI-set_variable('googlename', \$name\);
  exit 0;
}
  }
 
  @result = split(/-/, $fields[35]);
  chop($result[0]);
  if ($reverse) {
@words = split(/ /, $result[0]);

Re: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread John Novack

Michael D Schelin wrote:
snip
Again in the the last few years VoIP has come a long way as the PSTN 
has had over 100 years to perfect theirs. If we did not have to 
interface with the PSTN don't you think we would be better off?  They 
didn't have to interface with anybody else.
Well, if one studies the history of telephony, one sees that really 
isn't the case

From the earliest  common battery manual exchanges, step by step,, 
panel, and crossbar offices into digital controlled analog switching  
into full digital switched circuits, service, interconnections was 
maintained, and not always under the control of one company.
There were interface problems that filled books between the various 
switching schemes,

There wouldn't be nearly as many problems if today's engineers read a 
little of the history, and didn't continue to re-invent and discard 
schemes from the past.

And if VOIP is to succeed in any large way in the foreseeable future, it 
had better figure out BETTER ways to interface with the PSTN.

JMO
John Novack
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[Asterisk-Users] Cisco ATA 186 and Asterisk

2005-05-07 Thread Christopher Iarocci

Anyone have call waiting working on the ATA-186 connected to Asterisk? 
Other VoIP phones seem to work, but I can not get the ATAs to allow call
waiting.


Christopher M Iarocci
Network Admin
JD Posillico
631-249-1872 X244
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Re: [Asterisk-Users] Cisco ATA 186 and Asterisk

2005-05-07 Thread Deon

I think it has to do with your CallFeatures.

Callfeatures: 0x

I have a screen shot of my converters config if you want it, it supports
call waiting. I had to turn it off on one of my customers converters once,
I had to change the last 2 digits or something to turn off call waiting.
But it's on by default.

What I found interesting is that ATA-186's were originally designed by
Sipura for Cisco, according to something I read. I was wondering why
linksys's latest converters were made by Sipura, seeing as how their
parent company Cisco already made SIP converters, but it makes sense now,
Cisco just went back to Sipura. 


--- Christopher Iarocci [EMAIL PROTECTED] wrote:
 Anyone have call waiting working on the ATA-186 connected to Asterisk? 
 Other VoIP phones seem to work, but I can not get the ATAs to allow call
 waiting.
 
 
 Christopher M Iarocci
 Network Admin
 JD Posillico
 631-249-1872 X244
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Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

2005-05-07 Thread Peter Svensson

On Sat, 7 May 2005, Andrew Kohlsmith wrote:

 Perhaps I am naive but I don't think that diaplans would be that much more 
 complex if people matched more accurately at all.  Granted most of my calling 
 is north american, but there's some south america and germany in there as 
 well, along with a sprinkle of UK.  None of them would require _X. to match 
 the vast majority of the numbers in those locations.

But why would you want to match more then enough to know that you want to 
send the digits to a certain destination? 

 I would, however, enjoy it if some people could give examples of screwed up 
 PSTN dialing matching around the world.  I'd enjoy the challenge of creating 
 tightly-matching dialplans for them.

In Sweden we have

1-3 digits area code (optional for calls within the same are code)
5-9 digits subscriber number
A prefix 0 is added for area codes
A prefix 00 is added for international calls
The subscriber numbers are not fixed in length within an area code. 

Peter


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Re: [Asterisk-Users] Cisco ATA 186 and Asterisk

2005-05-07 Thread Christopher Kenna

if you could set me up with your config, that would be great.

thanx
Chris
[EMAIL PROTECTED] 5/7/2005 6:52 PM
I think it has to do with your CallFeatures.Callfeatures: 0xI have a screen shot of my converters config if you want it, it supportscall waiting. I had to turn it off on one of my customers converters once,I had to change the last 2 digits or something to turn off call waiting.But it's on by default.What I found interesting is that ATA-186's were originally designed bySipura for Cisco, according to something I read. I was wondering whylinksys's latest converters were made by Sipura, seeing as how theirparent company Cisco already made SIP converters, but it makes sense now,Cisco just went back to Sipura. --- Christopher Iarocci [EMAIL PROTECTED] wrote: Anyone have call waiting working on the ATA-186 connected to Asterisk? Other VoIP phones seem to work, but I can not get the ATAs to allow call waiting. Christopher M Iarocci Network Admin JD Posillico 631-249-1872 X244 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Good NAT Pnp Hardphone

Any special settings on * or your nat firewalls? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Nabeel Jafferali
|Sent: Sábado, 07 de Mayo de 2005 01:07 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Good NAT Pnp Hardphone
|
| For example,
| I want to give a phone to my brother, who is going to 
|europe.  His ICH 
| softphone is fine there.  Both the poly and cisco though require you 
| to setup for nat.  He would not be able to set this up though, so I 
| want to just give him a preconfig'ed phone and plug and go...
|
|My experience with the Sipura SPA-841 is that, as long as you 
|set up the STUN server and say Yes to all the options under 
|the NAT support parameters, it works pretty much plug-n-play. 
|I've tried it even behind three cascaded NAT routers and it works.
|
|--
|Nabeel Jafferali
|X2 Networks
|www.x2n.ca
|T: 1.647.722.6900
|   1.877.VOIP.X2N
|F: 1.866.655.6698
|FWD: 46990
|
|
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|http://lists.digium.com/mailman/listinfo/asterisk-users
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|   http://lists.digium.com/mailman/listinfo/asterisk-users
|

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Re: [Asterisk-Users] 3 x TDM400P in one PC ??

2005-05-07 Thread C F

On 5/6/05, Ariel Batista [EMAIL PROTECTED] wrote:
 I just setup one Dell SC420 with just one SATA drive and 512mg Ram ($
 404.00) with 2 TDM04B in it that is 8 FXO ports. And a second system for
 another customer with 3 TDM 2 TDM40B 8 FXS ports and one TDM01B for 4 FXO's.
 
 Both systems are working just fine.

You don't really know if it is working fine or not. Since you just
setup, at the most I'm assuming that it's in the past month that you
set it up.The trouble if they come will come later.
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Re: [Asterisk-Users] SEND TEXT to an extension?

2005-05-07 Thread C F

On 5/6/05, Anton Krall [EMAIL PROTECTED] wrote:
 Will this only work on polycoms? Do you need to be on an active call to send
 text?
 

As far as I know polycoms are the only phones that support it, but
there might be others.
No you don't really need to be on an active call to send text messages
using the polycom phones, however the way sendtext works is that you
have to be on an active call for asterisk to send the message.


 |-Original Message-
 |From: [EMAIL PROTECTED]
 |[mailto:[EMAIL PROTECTED] On Behalf Of C F
 |Sent: Viernes, 06 de Mayo de 2005 03:51 p.m.
 |To: Asterisk Users Mailing List - Non-Commercial Discussion
 |Subject: Re: [Asterisk-Users] SEND TEXT to an extension?
 |
 |Nope, I don't think it works, but maybe you could do the following:
 |use the polycoms autoanswer feature, drop a .call file to call
 |the polycom using the autoanswer feature, then have it just
 |use the sendtext, and hangup commands. This will do what you want.
 |
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RE: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread Chris Coulthurst

To be more specific to my point -- Using the internet today,
with the demands of streaming real-time applications, which require a
level of QoS wasn't originally designed in to IPv4.  With a wide array
of mods, patches and additions, there is 'some' support for
prioritization. We would be better off with a protocol suite like v6
that not only offers a solid packet-prioritization system, but several
key network enhancements as well.  Deployment is understandably
painstaking.

Sure, I agree IPv4 is legendary.  But using it with what the
internet now demands is like trying to get to the moon with a steam
locomotive.

Chris Coulthurst
[EMAIL PROTECTED]
 

|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Michael D Schelin
|Sent: Saturday, May 07, 2005 11:17 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Who's happy with their voip service?
|
|Isn't amazing what has happened in the last five or six years with the
|Internet.  There is no design flaw with IPv4. It was created back when
|you were in diapers and with todays pda's having more power than the
|systems back then.  An industry protocol that is going strong 30 or
more
|years is amazing in it's own right. How could they see the future of
|what we are doing today with the protocol.  I believe the engineers who
|designed IPv4 were brilliant men and did a great job designing
|something that is computer system neutral.
|
|Again in the the last few years VoIP has come a long way as the PSTN
has
|had over 100 years to perfect theirs. If we did not have to interface
|with the PSTN don't you think we would be better off?  They didn't have
|to interface with anybody else.
|
|
|Chris Coulthurst wrote:
|
|I tend to agree about the in-house being the 'stable part'.  Like
|anything else on the internet, if you don't have control of all parts
|(trunks and phones and dialplans), there are bound to be issues with
|uptime, and how your equipment responds to 'their' downtime.  It
reminds
|me of the headaches I had as an ISP when a BGP4 route wouldn't switch
to
|the redundant carrier, because the main carrier didn't really die, it
|just stopped transmitting!
|
|It's also worth noting the design flaws with IPv4 handling priority
|packets in the first place.  I think most of the little 'gotchas' in
|VoIP would magically vanish if QoS was something that could be
depended
|upon.  All you need is one router to not know how to pass the qos
token,
|and now you don't really have any!  Its another example hiw an
in-house
|system can be stable when you hold all the cards.
|
|By the time IPv6 gets here, it will be amazingly obsolete...
|
|Chris Coulthurst
|[EMAIL PROTECTED]
|
|
||-Original Message-
||From: [EMAIL PROTECTED] [mailto:asterisk-users-
||[EMAIL PROTECTED] On Behalf Of Andre Normandin
||Sent: Saturday, May 07, 2005 8:00 AM
||To: Asterisk Users Mailing List - Non-Commercial Discussion
||Subject: RE: [Asterisk-Users] Who's happy with their voip service?
||
||I've had Broadvoice for over a year now, and although their outages
are
||really annoying, the fact that their service costs $20/month
unlimited
|is
||what keeps me with them..
||
||I have 2 Inbound #'s through them (same account), one in GA (678-253)
|and
||one in CT (203-935), and overall their inbound has been more reliable
|than
||their outbound (minus the past week or so)..
||
||I have my dialplan try BV first, and then if it cannot use BV for
|outbound,
||it rolls to my pots line(s).. It actually works really well, except
|that if
||BV goes completely toes up Asterisk decides that it doesn't want to
do
||anything either :-(
||
||That is what I find the most annoying, quite frankly, BV is having
|Growing
||pains (in my opinion), and I can accept that, haven't put anything
|critical
||on my BV inbound, and 90% of the time BV outbound works fine.. The
rest
|of
||the time, the pots take care of outbound, and anyone who calls me
calls
|on
||my pots lines (except for family in GA, which is why I have the GA
#).
||
||For me personally, I just think VOIP is 'too' early in the maturity
|curve
||to
||really rely on it as a provider.. It's great in-house (medium/large
||companies), but for service, I think pots are the way to have rock
|solid
||service for the time being.
||
||I know of two of my friends that have Vonage as their only inbound
|numbers
||(not via asterisk, via the vonage locked adapters, so it is
completely
||vonange), and their service also has issues at times.. Granted, I'm
not
||sure
||if it's a true vonage issue, or their internet connection, but
|nonetheless,
||there are still issues..
||
||If I could get Asterisk so it just work continue 'working' properly
|with
||whatever SIP connections it can reach, I'd be a happy man..  Don't
get
|me
||wrong, I think Broadvoice needs to communicate better with their
|customer
||base, and the latest ongoing outage is, to say the least,

[Asterisk-Users] At home Asterisk via Broadvoice?

2005-05-07 Thread John Stegenga




Hi all - sorry if 
what I'm asking is FAQ by now - I only have 2789 digest messages that I've not 
read yet...

The local phone 
company (Bell South) has gotten completely out of hand with their rates, and 
with them suing anyone who wants to compete against them... So, I'm 
thinking very hard about going ALL VOIP here at home.

Hardware I 
have:
Old 586 
chassis
Old Pentium II 
laptop
Azatel 2 port 
adapter (Board ID C02v001.01.00 / Firmware 1.5.6-RC14)

I also have 3 AMD 
Athlons running Windows 2000.

What I want to 
do

Broadvoice will give 
me 2 lines, with 2 phone numbers each - distinctive ring - for a reasonable 
fee...

I want to handle 
processing the calls incoming on each number differently - 
line 1a = home 
phone, if no answer after 5 rings, run to asterisk for call processing (voice 
mail tree?)
Line 1b = 
(distinctive ring number) goes directly to fax machine (fax detects distinctive 
ring pattern for auto answer)

Line 2a = 'business 
phone' - ring, and process to voicemail if no answer (with other post 
processing - SMS to my cell phone, etc.)
Line 2b = 'kids 
phone number' - or something...

I'm thinking 
Asterisk because broadvoice provides voice mail 'PER LINE' not per number... and 
with asterisk I can do lots more!

So, can I run 
asterisk on the laptop - and have the calls routed to the Azatel to ring the 
phones??

Any thoughts? 
Clarifications?? Has anyone done something like this before? Can 
asterisk run on windows 2000 if it does not need to use 
hardware?

Thanks a MILLION 
times in advance, guys (and gals!)!

John


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RE: [Asterisk-Users] Good NAT Pnp Hardphone

 Any special settings on * or your nat firewalls?

Nope.

--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990

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Re: [Asterisk-Users] ZapBarge a PRI DDI

2005-05-07 Thread izo

On 5/6/05, Steve Rawlings [EMAIL PROTECTED] wrote:
 I'm using a TE405p with all four spans enabled, two configured as pri_cpe
 and two as pri_net, the asterisk is sitting between our ISDN (UK BT
 EuroISDN30) and our phone system.  We have 200 DDI numbers on the ISDN's and
 I need to give one of our clients dial-in access to be able to monitor calls
 we handle on their behalf.  I've got ZapBarge working but I need to restrict
 the client access to calls on their unique DDI only.  Unfortunately, of
 course at any moment in time a call on their DDI could be on any channel so
 I can't restrict it by channel and our agents are multi-skilled so can't
 restrict by agent.  I need to be able to allow by DDI only.  Can anyone
 suggest how I  might be able to achieve this?

SIMPLE 

exten = 12345/6789,1,ZapBarge()
exten = 12345/6789,2,Goto(BYEXTENSION,1)

where 12345 would be your DID
and 6789 would be callerid of your customer.

OTHER option is to use Disa application
with pin authorization and then redirect to specific extension with ZapBarge

rgrds
m.
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Re: [Asterisk-Users] At home Asterisk via Broadvoice?

On May 7, 2005 11:04 pm, John Stegenga wrote:
 Broadvoice will give me 2 lines, with 2 phone numbers each - distinctive
 ring - for a reasonable fee...

Please do a google search for broadvoice problems site:lists.digium.com and 
reconsider your choice of VOIP provider.

That reasonable fee doesn't actually include the thing working more than 50% 
of the time.

-A.
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RE: [Asterisk-Users] Good NAT Pnp Hardphone

Is your * open on the internet? No firewalls? And on the nat firealls no
need to open any ports or do port forwarding to your natted phone? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Nabeel Jafferali
|Sent: Sábado, 07 de Mayo de 2005 10:09 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Good NAT Pnp Hardphone
|
| Any special settings on * or your nat firewalls?
|
|Nope.
|
|--
|Nabeel Jafferali
|X2 Networks
|www.x2n.ca
|T: 1.647.722.6900
|   1.877.VOIP.X2N
|F: 1.866.655.6698
|FWD: 46990
|
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Re: [Asterisk-Users] At home Asterisk via Broadvoice?

2005-05-07 Thread Luki

My thoughts:

1) I would not run asterisk on a laptop, or on Windows (if you can get
it to turn properly via emulation like Vmware).

2) A 586 *might* be enough to handle this low call volume with no transcoding.

3) I know nothing about a Azatel 2 port adapter, but you could acquire
a Sipura 2000 (or similar) which can generate distinctive ring
patterns (I am not sure the Grandstream adapters can). Besides an ATA
you would not need other hardware in the asterisk box.

4) Be sure to check up the recent postings about Broadvoice. They
provided good service to me in the last year, but the last three days
were terrible.

5) Broadvoice does send the ALERT_INFO header for distinctive ring and
you can get asterisk to check this and handle the call differently. I
think they allow two additional numbers besides your primary number,
so technically three numbers per line.

Good luck...
--Luki
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[Asterisk-Users] Getting DTMF to work with SIP?

2005-05-07 Thread beonice

Folks, from googling, I see that the dtmfmode
parameter is not valid in the [general] context. 

My problem is that my overseas DID through Libretel
seems  to want to come into the [general] context!
And, having done that, I get my welcome message, but
then the DID does not accept the DTMF when I try to
dial an extension! It plays the welcome message,
waits, and then times out (and hangs up nicely, yes).

I've actually tried setting the dtmfmode parameter in
the [general] context, obviously with no success.

Also, I note that when the incoming call uses SIP, a
physical hangup by the caller is not recognised ... it
basically goes through my timeout routine and hangs up
at some point.

The same context in extensions.conf can be reached
through my California DID which uses IAX, and in that
case I have no problem with DTMF or hangup.

Any ideas on how to solve this? I've got a question
open at Libretel, too, but was hoping someone here may
have already dealt with this?

Thanks,
Maya




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[Asterisk-Users] Setting the jitter buffer in AIX

2005-05-07 Thread Bryce W Nesbitt

Are these things possible?
1) Set the local Asterisk jitterbuffer size, but only for a particular 
connection.  I'd like to force Asterisk to use a particularly large 
buffer in certain cases.  Should I expect this to work?

[general]
jitterbuffer=no
register = username:[EMAIL PROTECTED]   ;parcelfarce
register = username:[EMAIL PROTECTED]   ;iaxtel
[parcelfarce]	;connection to parcelfarce
type=friend
auth=md5
secret=password
context=inbound-from-parcelfarce
host=parcelface.domain.net
qualify=yes
jitterbuffer=yes
maxjitterbuffer=600   

2) Set the remote jitterbuffer.  I want to tell the remote Asterisk 
that, during this call or part of a call, that a much larger jitter 
buffer is OK.  Basically I care more about quality of the delivered 
sounds, rather than latency.

3) Monitor the remote jitter buffer discards.  I want to know if my 
outgoing stream is breaking up.

Here I am building an information retrival service, so conversational 
latency is not an issue.  The remote Asterisk is at a PSTN company such 
as VoicePulse, TelAIX or LiveVOIP.
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Re: [Asterisk-Users] Good NAT Pnp Hardphone

If Asterisk is on a public IP, nat=yes in sip.conf takes care of all the 
required magic.  No port forwarding needed anywhere, no special NAT 
settings needed on the phone.

Anton Krall wrote:
Is your * open on the internet? No firewalls? And on the nat firealls no
need to open any ports or do port forwarding to your natted phone? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Nabeel Jafferali
|Sent: Sábado, 07 de Mayo de 2005 10:09 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Good NAT Pnp Hardphone
|
| Any special settings on * or your nat firewalls?
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Re: [Asterisk-Users] Am I on the right track, and consultants

2005-05-07 Thread Andrew Latham

http://www.voip-info.org/tiki-index.php?page=Asterisk+consultants+USA

On 5/6/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 Hi,
 
 I've been lurking here for about a month and I've been putting together
 our companies planned migration to a new office and a new phone system.
   Could anybody tell me if there are any pitfalls to the following setup
 I may have missed?
 
 1:  TBD PC running *
 a:  Digium TDM04b FXO connected to PSTN
 1:  lines rolling over via SBC sentrex to get multi-lines on 
 1 number
 b:  Digium TE110p to Rhino channel bank 24/FXS
 
 We have a small investment in some SBC-410 analog phones and the office
 we're moving into is already extensively wired for analog.  I'm
 thinking of the 4 incoming lines purely because we don't know if we can
 afford a PRI.
 
 In any case, I'd like to know if there are any bugs or problems I may
 run into, and on a more extensive side, we'll probably need a
 consultant to help set this stuff up.
 
 Does anybody know if Digium is going to put up the list of consultants
 referred to on this page soon:
 
 http://www.astricon.net/training/dcap.shtml
 
 We need an asterisk/telco consultant in/near san francisco asap and I
 don't know where to look for one.  Is there a list on voip-info I
 haven't found yet?
 
 cheers,
 Dan 'tack' Trudell
 
 --
 t  r  a  c  t  i  o  n
 san francisco
 
 1.415.522.3592x8511 main
 1.415.522.3594 fax
 www.tractionco.com
 
 Traction. Step One in Forward Motion.
 
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-- 
sig
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
WWW: http://lathama.com
Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
/sig
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RE: [Asterisk-Users] Good NAT Pnp Hardphone

And if asterisk is behind nat doing prot forwarding? Say you just forwarded
udp 4569 5060 5004 1-2000? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Eric Wieling aka ManxPower
|Sent: Sábado, 07 de Mayo de 2005 11:02 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Good NAT Pnp Hardphone
|
|If Asterisk is on a public IP, nat=yes in sip.conf takes care 
|of all the required magic.  No port forwarding needed 
|anywhere, no special NAT settings needed on the phone.
|
|Anton Krall wrote:
| Is your * open on the internet? No firewalls? And on the nat 
|firealls 
| no need to open any ports or do port forwarding to your natted phone?
| 
| |-Original Message-
| |From: [EMAIL PROTECTED]
| |[mailto:[EMAIL PROTECTED] On Behalf 
|Of Nabeel 
| |Jafferali
| |Sent: Sábado, 07 de Mayo de 2005 10:09 p.m.
| |To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
| |Subject: RE: [Asterisk-Users] Good NAT Pnp Hardphone
| |
| | Any special settings on * or your nat firewalls?
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RE: [Asterisk-Users] Good NAT Pnp Hardphone

 And if asterisk is behind nat doing prot forwarding? Say you just
 forwarded
 udp 4569 5060 5004 1-2000?

You'd just need to set externip correctly, assuming you have a static public
IP.

--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990

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[Asterisk-Users] Distinctive Ring

Guys.

How do you configure asterisk to recognize distingtive ringing using x100p
cards? Can this be done and how?

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[Asterisk-Users] Asterisk@Home on OnComputers Show Sunday morning

2005-05-07 Thread Kerry Garrison





From somewhere in Cyberspace to all 
points on the compass its the On Computers Radio Show. 

Sunday's show will be all tech talk with the 
team. Peter Kastner will start off the show by talking about this week's hot 
news stories then the gang will continue on with tech talk. During the second 
hour, we will be joined by Kerry Garrison of the Geek 
Gazette to talk about [EMAIL PROTECTED]. Please email any 
questions or comments to [EMAIL PROTECTED] and 
we will respond as best we can.
Kerry Garrison is the founder of Tech Data Pros, a southern 
California IT service provider and the publisher of The Geek Gazette, a how-to 
site for the more technically inclined. For over 20 years Kerry Garrison has 
been on the forefront of technology from the early days of Apples and PC's to 
the Internet, and currently is one of the leading proponents of VOIP 
systems.
The OnComputer show is 
broadcast live over the internet from http://oncomputers.info Sunday morning at 
10am PST.
Tech Data Pros(949) 502-7819http://techdatapros.comGeek Gazettehttp://geekgazette.com

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Re: [Asterisk-Users] Good NAT Pnp Hardphone