Re: [Asterisk-Users] e164.org

[Marc Storck]

configure your asterisk to use e164.org and make use of EnumLookup
then try to call +352 818 595, if your call goes to [EMAIL PROTECTED] 
then you can call me for free over the net!

Marc
At 03:33 23.05.2004, you wrote:
Dean Collins wrote:

Tony, as per you inference that e164 are up to something shady, you
should talk to one of the founders Duane, he currently has about 5 open
If it's the same duane who runs cacert he probably means well... however 
having read the site I'm still not sure whether i'd use it myself (it 
means trusting an external database to produce a least cost route.. I'm 
just not that trusting).

Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle <[EMAIL PROTECTED]>  Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2  02B8 E547 470E 4F4B 6917
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Re: [Asterisk-Users] VoicePulse SIP

[Marc Storck]

asterisk only supports IAX2, SIP and TEL, it will only use IAX2 and SIP 
entries however

so it is used to route via the Net if it cannot find a route via the 
Net or the link isn't working it will go to the next priority in your 
dialplan and do whatever you want, it doesn't re-configure your dialplan or 
route preferences let's say it's a bypass to IAX and SIP providers as 
it will tell you the username and server where users may be reached directly!!!

Marc
At 23:50 22.05.2004, you wrote:
Andres wrote:
[EMAIL PROTECTED] wrote:
Which providers give you a jitter buffer?

In Europe: VoipTalk and Magrathea.  In the US: Iconnecthere.   I am sure 
there are more.
Clearpath gives jitter buffer as well.  http://www.clearpath1.com/
John
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Re: [Asterisk-Users] sip call using name in sip.conf

[Olle E. Johansson]

Randy Bush wrote:
[foo]
type=friend
I do not beleive that will work for type=friend.  If you use separate
type=peer and type=user blocks in sip.conf it may work.  Expecially
if you also specify a port in the Dial().
Else, use the hostname (or a const).

hmmm.  then, how do i let it be dynamic if it has two
blocks in sip.conf, one for inbound and one for out?
i.e, how does it register its ip address in both?
For the user= part, there's no need to register, since we receive calls from
a user and the user authenticate for doing this.
For the peer= part, the peer registers so we can call them at the current
IP address (if host=dynamic).
/O
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Re: [Asterisk-Users] Asterisk firewall config

[Brancaleoni Matteo]

Hi.

Il dom, 2004-05-23 alle 01:52, Tony Hoyle ha scritto:
> Surely it depends on who's calling me - if they're using a SIP phone it'll 
> come in over the SIP port, and if they're using an IAX phone it'll come in 
> over the IAX port - ie there's this context in the default iax.conf:
> 
> [guest]
> type=user
> context=default
> callerid="Guest IAX User"

for letting unauthorized user to call you over IAX(2).
Like a pstn call... everyone can call you if the have your
number (or IP in Voip calls)
If you don't want that, just delete that entry :)

> btw. how many rtp streams do I need?  I only have 1 phone at the moment (max. 
> will be about 4 I think).

mmh... I dunno the values of that association, but
bear in mind that:
* are only UDP ports
* are opened only during a RTP session, in a dynamic way

so leaving open ports 1 to 2 UDP as in default rtp.conf
isn't a problem, since there's not any port open...
(unless you run any udp service on that interval :) )
and a portscan will detect these port as closed.

only during a call, * and the phone will handshake an RTP
port and use that. otherwise will be closed.

Matteo.
-- 
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl

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RE: [Asterisk-Users] Caller ID with BT CD50

[Karl Dyson]

Well, I have a USR015630B, which, according to the FAQ supports (UK)
CLI. It supports the at #cli command, but no matter what I try, it will
not pick up the caller id. Lucky I already had it and didn't buy it
soley for this purpose! My caller display unit (unfortunately a CD60 --
which I've opened to look for CD50 like boards, chips etc) picks up CLI
no problem (as you'd expect), as do my DECT phones. of course, only
when they're plugged into the line, and not when they're plugged into
the ATA186.

I will be persuading someone I know to swap the USR for a Hayes modem
later today to see if that will do it. otherwise out will come the
soldering iron, and eBay will see me bidding on CD50s!

Cheers,

Karl

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tony Hoyle
> Sent: 22 May 2004 23:55
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Caller ID with BT CD50
> 
> Karl Dyson wrote:
> 
> > Hi All,
> >
> > Having searched the archives, I can see there has been much
discussion
> > at various points regarding capture of caller id information from
good
> > old BT.
> >
> > If I understand correctly, it seems that not only do the drivers not
> > currently support it, but my X101P possibly/probably can't do it
anyway
> > due to hardware?
> 
>  From the details on http://www.ainslie.org.uk/callerid/cli_faq.htm it
> sounds
> like it wouldn't be too hard to implement, however:
> 
> "The only manufacturers that have ever supported BT Caller ID are
Pace,
> Hayes
> (Europe), and 3Com/US Robotics."
> 
> It then goes on to state all 3 of those manufactures no longer support
it.
> 
> I wonder if the low cost geographic VOIP numbers support it?
> 
> Tony
> 
> --
> Te audire no possum. Musa sapientum fixa est in aure.
> 
> Tony Hoyle <[EMAIL PROTECTED]>  Key ID: 104D/4F4B6917 2003-09-13
> Fingerprint: 063C AFB4 3026 F724 0AA2  02B8 E547 470E 4F4B 6917
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[Asterisk-Users] snom reporting busy when it shouldn't

[nicolas]

I am using asterisk cvs. 

Incoming/Outgoing calls are working.
Calling the phone when some other lines are in use on the phone is ok.
What does not work though is when the phone is ringing, nobody else can 
call the phone anymore.

That's what * is saying:

-- Got SIP response 486 "Busy Here" back from 192.168.1.250
-- SIP/snom1-4a44 is busy

I am using the 2.05e snom200 firmware.

Snom people sad must run.
nico

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Re: [Asterisk-Users] e164.org

[Duane]

Tony Hoyle wrote:
Yes, same Duane as CAcert.org...
So eg. if I've registered 3 different sip providers and an IAX provider, 
plus a couple of landlines what
is it doing?  I guess I'm missing the point somewhere.
The point is as simple or as complex as you like it to be. You can 
configure things to have only a single voip URL reference, then anyone 
that dials your pstn numbers will be directed to that voip server, or in 
the case of businesses that want to list every method that they can be 
contacted by such as email, instant messaging they can, and at some 
stage in the future if fax machines get smart enough and their is an 
enum record on the number you're faxing they could email the fax instead 
of faxing it.

The way I understand it is you pass it a phone number and it gives you a 
prefferred route to that number, which may be VOIP and may be POTS or 
from the looks of it MSN and lots of other things (including ldap???!!).
LDAP records are actually listed in the RFC and given as an example.
You then pass that result straight into a Dial command, which means it 
could potentially do absolutely anything, including call the chinese 
speaking clock at peak rate.
If there is an IAX2 or SIP or H323 NAPTR record in DNS, this increments 
the dial plan by +1, if it's a TEL (i.e. the talking clock in china) it 
increments by +51, and increments by +101 if it fails, so unless you 
tell it to dial the number at +51 it won't use any TEL fields from DNS 
so this can't happen unless you specificly allow it to.

Flip side is we're trying to do an app_dialenum to actually better 
implement enum lookups and dialling, if it hits a tel field the plan is 
to get it to read out the number and prompt the user to press 1 to skip 
and try another record or fail to pstn, or press 2 to use the tel number.

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
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RE: [Asterisk-Users] snom reporting busy when it shouldn't

[Christian Stredicke]

Did you check if the phone is in DND state? Is there anything strange on the
display?

CS

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of nicolas
> Sent: Sunday, May 23, 2004 5:43 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] snom reporting busy when it shouldn't
> 
> I am using asterisk cvs.
> 
> Incoming/Outgoing calls are working.
> Calling the phone when some other lines are in use on the phone is ok.
> What does not work though is when the phone is ringing, nobody else can
> call the phone anymore.
> 
> That's what * is saying:
> 
> -- Got SIP response 486 "Busy Here" back from 192.168.1.250
> -- SIP/snom1-4a44 is busy
> 
> I am using the 2.05e snom200 firmware.
> 
> Snom people sad must run.
> nico
> 
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Re: [Asterisk-Users] Asterisk slashdotted

[tmpm]

Yes, BRAVO!
The more read about it the merrierand the more participation.
I've had this discussion with peers recently, the overpriced land-line 
model is dying a swift death...I don't know HOW many people I've talked to 
who have said "FSCK" the phone company...and these people are ones who 
don't have a clue about linux, let alone *, they want something, anything, 
that tells the overpriced bell-ski's to "SHOVE IT", and they're willing to 
cooperate...to that end, we're making progress.
The aim is to provide toll quality, and this product allows everyone to BE 
their own phone switch, and once you show them the features, their eyes bug 
out, and their jaws dropinevitably followed by, "when can I get it"?
Doesn't matter the size of the operation, I've had bars, small businesses, 
homeowners, everyone who can appreciate the idea that THEY, not some 
slovenly bell-co can have their own dialing plan, and be their own destiny 
is falling all over themselves to get it.

WE ARE the future of telephony, lets act like it, lets get a generic, 
bulletproof model out there, and let the consumers have a go at 
it...they're chomping at the bit to do soI envision something like a 
model someone with a bit of savvy can administer from a html page like they 
set up their wireless access point...forms, fill in the blanks...am I 
wrong? I don't think so...
Right on brothers and sisters, WE ARE the 21'st century model of phone 
service, and the slashdoting is a free plug we should be thankful 
for...again as we all know, the open source model is champion...its YOU who 
make it workthank you all, theres some real brain trust here..many 
will follow our lead...lets rock...


At 19:51 5/22/2004, you wrote:
Congradulations to the Asterisk gang on getting slashdotted!
http://slashdot.org/article.pl?sid=04/05/22/1840220
Cheers,
Rich

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[Asterisk-Users] SIP with TerraCall Error

[cary]

Dear All,

I had try the new cvs version asterisk to connect to TerraCall, but fail with
the follow reply, anyone know how to solve this problem.

NOTICE[1133742896]: Failed to authenticate on INVITE to '"account number"
;tag=as1d02a70e'

Thank You.

Cary LEUNG
Administrator
CARYNET Information Center
Hong Kong

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Re: [Asterisk-Users] e164.org

[Tony Hoyle]

Duane wrote:
If there is an IAX2 or SIP or H323 NAPTR record in DNS, this increments 
the dial plan by +1, if it's a TEL (i.e. the talking clock in china) it 
increments by +51, and increments by +101 if it fails, so unless you 
tell it to dial the number at +51 it won't use any TEL fields from DNS 
so this can't happen unless you specificly allow it to.
That helps (at least until VOIP calls start being charged by the 
minute).  Maybe someone needs to implement a switch/case statement in 
extensions.conf for this kind of stuff at some point.

Flip side is we're trying to do an app_dialenum to actually better 
implement enum lookups and dialling, if it hits a tel field the plan is 
to get it to read out the number and prompt the user to press 1 to skip 
and try another record or fail to pstn, or press 2 to use the tel number.
If you could pass it a list of preferred/available protocols it'd be 
nice - eg. I don't support h323 (haven't got it to compile yet) and 
prefer SIP over IAX2 because of other problems (choppy sound).

Tony
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Re: [Asterisk-Users] Caller ID with BT CD50

[Tony Hoyle]

Karl Dyson wrote:
Well, I have a USR015630B, which, according to the FAQ supports (UK)
CLI. It supports the at #cli command, but no matter what I try, it will
not pick up the caller id. Lucky I already had it and didn't buy it
soley for this purpose! My caller display unit (unfortunately a CD60 --
which I've opened to look for CD50 like boards, chips etc) picks up CLI
no problem (as you'd expect), as do my DECT phones. of course, only
when they're plugged into the line, and not when they're plugged into
the ATA186.
I have a USR modem too but it's a brick unfortunately...  Pity as it was 
a nice modem... the right model too so it might have worked.

I wonder what it would take to get the zaptel drivers to pick up CID (or 
even a cheap conexxant modem or something like that) - these soft modems 
are pretty dumb so I'd expect there to be some scope for frobbing them 
to do new stuff.

Tony
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Re: [Asterisk-Users] e164.org

[Duane]

Tony Hoyle wrote:
That helps (at least until VOIP calls start being charged by the 
minute).  Maybe someone needs to implement a switch/case statement in 
extensions.conf for this kind of stuff at some point.
Already exists, the examples on the website if a TEL field is hit they 
just drop out and dial the PSTN number that was fed into it in the first 
place and not the number returned from DNS...

If you could pass it a list of preferred/available protocols it'd be 
nice - eg. I don't support h323 (haven't got it to compile yet) and 
prefer SIP over IAX2 because of other problems (choppy sound).
Beauty of DNS is you can pull all records into the client app and sort 
it out yourself, the DNS records can be given preferences over each 
other, but it's up to the client to follow the RFC or make their own 
rules up on how to handle things... The code we're planning to implement 
will be as RFC compliant as much as we can possibly make it, although I 
pondered before about giving IAX preference from within asterisk as a 
config option...

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
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Re: [Asterisk-Users] RxFAX generates no tiff file

[Mike Heininger]

Am 23.05.2004 um 04:33 schrieb Steve Underwood:
How do you run rxfax? You problem is probably something to do with 
that. Your's is the first report I have had of no TIFF file 
whatsoever.
[internalexten]
exten => 5000,1,Dial(SIP/mike,60,tr)
exten => 5000,2,SetLanguage(de)
exten => 5000,3,Playback(vm-nobodyavail)
exten => 6000,1,WaitMusicOnHold(30)
exten => 7000,1,rxfax(/tmp/testfax.tif)
[default]
include => internalexten
exten => h,1,Hangup
The context for the inbound call is [default] and goes to extension 
7000.

TIA,
Mike
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RE: [Asterisk-Users] Caller ID with BT CD50

[Karl Dyson]

> I have a USR modem too but it's a brick unfortunately...  Pity as it
was
> a nice modem... the right model too so it might have worked.
> 
> I wonder what it would take to get the zaptel drivers to pick up CID
(or
> even a cheap conexxant modem or something like that) - these soft
modems
> are pretty dumb so I'd expect there to be some scope for frobbing them
> to do new stuff.
> 
> Tony
> 

As I say, I lent a Hayes modem to mother in law a year ago, but have
since upgraded her to a 56k internal, but cunningly left the old kit in
her loft (think of it as redundant storage of my old computer crap ;))
so will pick it up later and try it. failing that, I've located a
possible Pace candidate

Of course, although my wife is happy with the Cisco 7905s that have
sprung up around the house, she still likes the cordless DECT units we
have, and so they're plugged into an ATA186. Problem is, they no longer
display caller id due to the ATA186 not poking it out in BT format I
guess. If I were to buy some US cordless handsets would they do the
caller id display? Or am I pushing my luck now! (I'm afraid, nice as
they look, the Cisco 7920 cordless phones are a bit out of my price
range!!)

Cheers,

Karl



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Re: RE: [Asterisk-Users] Failed to bind to 0.0.0.0:5060: Address already in use

[Stefan-Michael. Günther (in-put GbR)]

Jay Milk worte:
>Why don't you configure the IP address instead of leaving it 0.0.0.0?
>While that should work, it's always a good idea to be specific about
>your bindings.
> 
I already tested the configuration with

bindaddr=192.168.0.101

which is the correct ip of the asterisk box - same result: Address already in 
use.

Stefan


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Stefan-Michael. G=FCnther (in-put GbR)
Sent: Friday, May 21, 2004 11:58 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Failed to bind to 0.0.0.0:5060: Address
already in use


 Hi,

I have a strange error, that I haven't yet found in another posting on
the  archive.  Here's my sip.conf:

[general]
port =3D 5060 ; Port to bind to
bindaddr =3D 0.0.0.0  ; Address to bind SIP channel to
context =3D default   ; Default context for incoming calls
callerid=3DNo CallID


-- 

*
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Günther
Moltkestraße 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
*

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Re: [Asterisk-Users] T100P HDLC configuration

[Christian Hoffmeyer]

- Original Message - 
From: "Vasyl Rublyov" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 22, 2004 10:02 PM
Subject: Re: [Asterisk-Users] T100P HDLC configuration


Just would like to add, of course if it is going to help:
   I am using Linux 2.4.26 on Linux, compiled from sources and latest zaptel
sources.
   We have T1 Internet from Verizon


HDLC does not work on kernel 2.4.26

Use kernel 2.4.22

Christian Hoffmeyer
YottaDot Solutions
Huntsville, AL

(w)   256.851.8689
(c)256.655.0321
(iax)  700.859.4508

Ask me about Asterisk.

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[Asterisk-Users] *** Asterisk Sunday News: Conferences on the phone and IRL - "in real life"

[Olle E. Johansson]

Here in Sweden, it's supposed to be springtime. A wonderful time of the year,
with sunny skies and wonderful weather. Almost summer. Today, it's not.
It's winter all over again with rain and only 3 degrees celsius outside.
Better to stay inside and write a weekly Asterisk newsletter :-)
This week's topics:
---
* Looking beyond Asterisk 1.0/1.1 - what's up?
* Asterisk on SlashDot
* Asterisk on the O'Reilly Open Source Convention
* Writing community documentation - asteriskdocs.org
* Application of the week: The MeetMe conference system
* Alternatives to the standard MeetMe
*** Looking beyond Asterisk 1.0/1.1 - what's up?
-
  At Astricon in September we'll arrange a developer's meeting. This is
the first time we gather all Asterisk programmers and documentation
writers into a meeting that will be held a the third day of Astricon.
The idea is to be able to discuss future architectures and major
additions. The Digium team, with Mark Spencer in the lead, will be
an active part of othis meeting.
There are several ways to set up a meeting like this, from
totally uncontrolled chaos to organized semi-controlled chaos.
We're aiming for the latter. Read more on how you can contribute:
- http://lists.digium.com/pipermail/asterisk-users/2004-May/046858.html
- http://www.astricon.net
*** Asterisk on SlashDot

  Asterisk was mentioned on SlashDot this week, together with e164.org,
an ENUM-like registry created by very active Asterisk contributors.
In fact, we've added an e164.org-specific app to Asterisk in cvs head.
Read more:
- http://slashdot.org/article.pl?sid=04/05/22/1840220
- http://www.e164.org
- http://www.voip-info.org/tiki-index.php?page=ENUM
*** Asterisk on the O'Reilly Open Source Convention
---
  John Todd, an active member of the Asterisk community, is talking about
Asterisk on the O'Reilly Open Source Convention in Portland, Oregon,
in July:
  "This talk details how to insert Asterisk into your organization in ways
  that that won't cost any significant money, yet will impress senior
  management with only a few clever tricks to the end goal of your receiving
  funding for a full-blown beta test. The savings that can be realized from
  using Asterisk are truly astounding, and the features available at no cost
  are one of the best examples of tangible cost savings
  that open source can provide."
* http://conferences.oreillynet.com/cs/os2004/view/e_sess/5016
*** Writing community documentation - asteriskdocs.org
--
  Maybe we should produce a couple of Asterisk OpenOffice Impress
presentations that we all can use on local Open Source community
gatherings? I'll hand that idea over to the Asterisk Docs project.
On the topic of docs, they have restarted and are having regular MeetMe
phone conferences. They need your help, we need more people writing
documentation, making sure we're not adding functions without
proper documentation, explaining to new users how great
Asterisk really is behind all the code and cryptic commands.
The Asterisk Docs project is awarded my "Asterisk project of the week"
award. They have the worst web site, but is a great project to join.
(Yes, I've seen the new web layout suggestions, they're great :-)
- http://www.asteriskdocs.org
- Mailing list and IRC channel is available
*** Application of the week: The MeetMe conference system
-
  Have you discovered the not-so-simple-anymore "Simple Meetme Conference
Bridge" application? It's a killer app in Asterisk. With this app,
you get a lot of functionality:
- Multiuser conferencing over multiple channels - VoIP and PSTN
- Announcements: One talker, multiple listeners
- Administration - kicking users, muting and locking conferences
- Background music while waiting for other participants
- Statically configured conferences with pin code auth for entry
- Dynamic conferences, created on demand
  (the conference number is told at first user entry)
In CVS head, there's been a number of additions to MeetMe lately,
additions that make MeetMe much more than a "simple" conference bridge.
The dial plan applications for the  MeetMe system in CVS head are
- MeetMe - enter a conference (in some cases creates a dynamic conference)
- MeetMeAdmin - kick users, lock conferences, mute conferences
- MeetMeCount - count number of participants for a conference
There's also a new family of CLI commands for managing MeetMe:
- MeetmeList conferences
- MeetMe kick Kick a user out of a conference
- MeetMe kick  allKick all users
- MeetMe list List participants in a conference
- MeetMe lock Lock a conference - no more users
- MeetMe unlock   Unlock a conference
- MeetMe mute Mute a conference
- MeetMe unmute   Unmute a conference
Some additional notes

RE: [Asterisk-Users] Caller ID with BT CD50

[gARetH baBB]

On Sun, 23 May 2004, Karl Dyson wrote:

> Of course, although my wife is happy with the Cisco 7905s that have 
> sprung up around the house, she still likes the cordless DECT units we 
> have, and so they're plugged into an ATA186. Problem is, they no longer 
> display caller id due to the ATA186 not poking it out in BT format I 
> guess. If I were to buy some US cordless handsets would they do the 

ATA186 firmware 3.0+ supports more formats.

Certainly with 3.0 my BT DECT 3010 (rebadged Siemens) base copes fine.
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Re: [Asterisk-Users] Caller ID with BT CD50

[Tony Hoyle]

Karl Dyson wrote:
Of course, although my wife is happy with the Cisco 7905s that have
sprung up around the house, she still likes the cordless DECT units we
have, and so they're plugged into an ATA186. Problem is, they no longer
display caller id due to the ATA186 not poking it out in BT format I
guess. If I were to buy some US cordless handsets would they do the
caller id display? Or am I pushing my luck now! (I'm afraid, nice as
they look, the Cisco 7920 cordless phones are a bit out of my price
range!!)
If you had CID to start with, I'd expect it to work - eg. you wouldn't get it 
from the POTS line but if a VOIP call came in and the ATA186 retransmitted 
that in US format then a US handset would pick it up.

The CID specs look really simple... I'll definately have a go at implementing 
something like it.  Just need to find a spec sheet for the Intel chipset in 
the FX100/FX101 (to see if you somehow see the pulse that comes just before 
the ring - I always wondered why phones tended to blip a second before 
starting to ring these days.. now I know).

Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle <[EMAIL PROTECTED]>  Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2  02B8 E547 470E 4F4B 6917
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RE: [Asterisk-Users] Caller ID with BT CD50

[Karl Dyson]

> 
> ATA186 firmware 3.0+ supports more formats.
> 
> Certainly with 3.0 my BT DECT 3010 (rebadged Siemens) base copes fine.

Oooh Now which setting would I need to check?? I have a Philips Onis
DECT system, which does CLI quite happily on the BT line, and I just
checked the ATA and its running :

ata000e841adb36
Version: v3.0.0 atasip (Build 031210A)
MAC: 0.14.132.26.219.54
SerialNumber: INM07491B0E
ProductId: ATA186I2
Features: 0x
HardwareVersion: 0x0006 0x

So. I guess I must have a setting incorrect??

Thanks in advance,

Karl



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RE: [Asterisk-Users] Caller ID with BT CD50

[Stephen Davies]

On Sun, 23 May 2004, gARetH baBB wrote:

> On Sun, 23 May 2004, Karl Dyson wrote:
> 
> > Of course, although my wife is happy with the Cisco 7905s that have 
> > sprung up around the house, she still likes the cordless DECT units we 
> > have, and so they're plugged into an ATA186. Problem is, they no longer 
> > display caller id due to the ATA186 not poking it out in BT format I 
> > guess. If I were to buy some US cordless handsets would they do the 
> 
> ATA186 firmware 3.0+ supports more formats.
> 
> Certainly with 3.0 my BT DECT 3010 (rebadged Siemens) base copes fine.

I also found my Philips DECT phone in the UK had a few CID modes of
which one worked with the ATA.  Only downside was CID only displayed
after the first ring-ring.

Steve


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RE: [Asterisk-Users] Caller ID with BT CD50

[Karl Dyson]

> If you had CID to start with, I'd expect it to work - eg. you wouldn't
get
> it
> from the POTS line but if a VOIP call came in and the ATA186
retransmitted
> that in US format then a US handset would pick it up.

But once I can get the cid working, I'm hoping I can persuade the DECT
units to pick it up obviously the 7905s display CID happily. but
currently only from each other, or incoming IAX calls.
 
>I always wondered why phones tended to blip a second before
> starting to ring these days.. now I know).

Me too. suddenly dawned on me recently that old phones would blip
due to the CLI coming through moments before the "ring"



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Re: [Asterisk-Users] app_queue and app_groupcount

[Julien Levi]

Troy Settle wrote:
I just disable call waiting on all my sip phones and on all zap interfaces.
No problem.
That is fine if it can be done, though I prefer to keep as much set-up
info on the server as possible for easier admin. Do you know of a
softphone with such an option? I've been unable to find one that rejects
calls when on line is busy.
regards,
--
drbob

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RE: [Asterisk-Users] Caller ID with BT CD50

[gARetH baBB]

On Sun, 23 May 2004, Karl Dyson wrote:

> Oooh Now which setting would I need to check?? I have a Philips Onis
> DECT system, which does CLI quite happily on the BT line, and I just
> checked the ATA and its running :

http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a00801e0eb0.html#wp1113416

ETSI method (type 2).

I know my CD50 still doesn't like this method, but then as the first 
generation device it can be *very* fussy at the best of times - indeed, I 
think very short line lengths were a problem with this device, they have 
problems with ISDN where the local NTE actually generates CDS.

But as said the BT DECT 3010 is fine, and so is my CD20 etc.
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[Asterisk-Users] IAX2 REACHABLE/UNREACHABLE

[John Blackman]

All,

 

I have an issue with IAX that I can’t comprehend.  Approximately every eight minutes my
servers go unreachable.  They stay
unreachable for exactly 10ms.  I
have two servers running IAX and it happens on both servers simultaneously.  I have searched the archives and see
similar issues, but not the exact same one.  I am on the current CVS stable version of
*.

 

Also, during IAX calls, every couple of minutes there is
about a second of silence.  Silence
suppression is turned off on all phones (Grandstream
BT101s).  This issue does not occur
during SIP calls.

 

I do have a similar registration problem with SIP.  There are two phones on one site that go
reachable/unreachable.  (See extension  in the sample below.  This is a B101 that is registered to my
server, but it happens to be at the same location as the second * server.)

 

If anyone can point me in the right direction for troubleshooting,
I would appreciate it.  I am pasting
example error messages that I receive below.

 

TIA,

 

 

 

John



 



 

CLI Output:

 

May 23 12:50:05 NOTICE[7176]:
chan_iax2.c:5616 iax2_poke_noanswer: Peer 'arlington' is now
UNREACHABLE!

May 23 12:50:15 NOTICE[7176]:
chan_iax2.c:5213 socket_read: Peer 'arlington' is now
REACHABLE!

May 23 12:58:19 NOTICE[7176]:
chan_iax2.c:5616 iax2_poke_noanswer: Peer 'arlington' is now
UNREACHABLE!

May 23 12:58:29 NOTICE[7176]:
chan_iax2.c:5213 socket_read: Peer 'arlington' is now
REACHABLE!

May 23 13:06:07 NOTICE[5126]:
chan_sip.c:6901 sip_poke_noanswer: Peer '' is now
UNREACHABLE!

May 23 13:06:34 NOTICE[7176]:
chan_iax2.c:5616 iax2_poke_noanswer: Peer 'arlington' is now
UNREACHABLE!

May 23 13:06:44 NOTICE[7176]:
chan_iax2.c:5213 socket_read: Peer 'arlington' is now
REACHABLE!

May 23 13:14:49 NOTICE[7176]:
chan_iax2.c:5616 iax2_poke_noanswer: Peer 'arlington' is now
UNREACHABLE!

May 23 13:14:59 NOTICE[7176]:
chan_iax2.c:5213 socket_read: Peer 'arlington' is now
REACHABLE!

May 23 13:17:56 NOTICE[5126]:
chan_sip.c:5860 handle_response: Peer '' is now
REACHABLE!

 

Iax.conf:

 

[general]

 

;port=5036  ;I commented this out because I
discovered it is hard coded anyway.

bindaddr=10.20.30.10

amaflags=default

accountcode=default

bandwidth=low

disallow=all

allow=gsm

jitterbuffer=no

;dropcount=3

;maxjitterbuffer=500

;maxexcessbuffer=100

;trunkfreq=20

tos=lowdelay

authdebug=no

 

 

register
=> username:[EMAIL PROTECTED]  ;these are real values that I changed
for security reasons

 

 

 

[arlington]

 

type =
friend

context =
local

callerid
= 

auth =
plaintext

secret = password

;inkeys
= 

;outkey
= 

host =
dynamic

;defaultip
=

;accountcode
=

qualify =
yes

;mailbox =

trunk =
yes

Re: [Asterisk-Users] How to share Zap channels in 2 Asterisk servers

[Deepak Malhotra]

Thanks it works
- Original Message - 
From: "David J Carter" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 22, 2004 10:40 AM
Subject: RE: [Asterisk-Users] How to share Zap channels in 2 Asterisk
servers


> Call the PBX300 using IAX2 from PBX200, make sure that the call goes into
> the context that allows dial out.
>
> Example.
>
> exten => _543219XX,1,StripMSD,5
> exten => _9XX,2,Dial/[EMAIL PROTECTED]/BYEXTENSION
>
> The first line looks for an access code '54321' followed by the access
code
> for an outside line '9' and then a number.
> You next strip the access code for IAX linking and pass the rest to the
> other Asterisk PBX.
> The Asterisk PBX then runs the exten as if on the local machine.
>
> Simple huh.
>
> There is most likely a simpler method, but this works for me.
>
> Dave
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of
> [EMAIL PROTECTED]
> Sent: 22 May 2004 16:40
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] How to share Zap channels in 2 Asterisk
> servers
>
>
>
> Hello
>
> I am trying to setup Asterisk on 2 servers PBX300 and PBX200.
> PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap
> device.
> Softphone from PBX200 can talk to softphone on PBX300 but no outgoing call
> from
> PBX200.
> I can call from PBX300 outside but I am unable to configure soft Phone
> defined
> in PBX200 to dial out side using PBX300 Zap devices.
>
> I am geting error message " Rejected connect attempt from PBX200".
>
> Please help if this is possible.
>
> Thanks
>
> Deepak
>
>
>
> 
> This message was sent using IMP, the Internet Messaging Program.
>
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Re: [Asterisk-Users] Failed to bind to 0.0.0.0:5060: Address already in use

[Tilghman Lesher]

On Sunday 23 May 2004 07:09, Stefan-Michael. Günther (in-put GbR) 
wrote:
> Jay Milk worte:
> >Why don't you configure the IP address instead of leaving it
> > 0.0.0.0? While that should work, it's always a good idea to be
> > specific about your bindings.
>
> I already tested the configuration with
>
> bindaddr=192.168.0.101
>
> which is the correct ip of the asterisk box - same result: Address
> already in use.

Try the output (as root) of:  netstat -tunap

I suspect you'll find your competing process.

-- 
Tilghman
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RE: [Asterisk-Users] Caller ID with BT CD50

[Karl Dyson]

> 
>
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administr
at
> ion_guide_chapter09186a00801e0eb0.html#wp1113416
> 
> ETSI method (type 2).
> 
> I know my CD50 still doesn't like this method, but then as the first
> generation device it can be *very* fussy at the best of times -
indeed, I
> think very short line lengths were a problem with this device, they
have
> problems with ISDN where the local NTE actually generates CDS.
> 
> But as said the BT DECT 3010 is fine, and so is my CD20 etc.

Thanks for that I had only really thought of it while typing one of
my earlier replies about inbound CLI, and has automatically assumed that
the ata186 would not do CLI in a BT/UK way!

It looks like the Philips units don't like it anyway (they are ~4 yrs
old, mind). I've tried 19e62 and 19e66 so far (the main bits should be
fine as defaults I think, and so I just tried getting it to send it
before ringing, and after the 1st ring).

I'll try the BT CD 60 unit I have in a while to see if that can see the
CLI coming from the ATA.

Cheers,

Karl



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[Asterisk-Users] ZAPTEL not loading on FC2

[Taz Man]

Hello all,
I've just installed the Fedora core 2 and tried to compile the asterisk and
the zaptel drivers
Asterisk went smooth but I had troubles with the zaptel.
I did copy the .config file under the kernel source and make oldconfig and
make include/asm ; make include/version.h ; make SUBDIRS=scripts
I was able to compile the zaptel source, using the make linux26 and then ran
make install.
I see that the .ko files are in place (under
/lib/modules/2.6.5-1.358/misc/...) but when trying to load the zaptel I get:

[EMAIL PROTECTED] misc]# modprobe zaptel
FATAL: Error inserting zaptel (/lib/modules/2.6.5-1.358/misc/zaptel.ko):
Invalid module format

[EMAIL PROTECTED] misc]# modprobe wcfxo
WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358/misc/zaptel.ko):
Invalid module format
WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358/misc/zaptel.ko):
Invalid module format
FATAL: Error inserting wcfxo (/lib/modules/2.6.5-1.358/misc/wcfxo.ko):
Invalid module format
FATAL: Error running install command for wcfxo

uname -a gives
Linux server 2.6.5-1.358 #1 Sat May 8 09:04:50 EDT 2004 i686 i686 i386
GNU/Linux

any help? any Ideas?
10x, Ronen


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[Asterisk-Users] IAX2 NAT / Registration Issue

[Steven Sokol]

I have a client using IAX Phone at his office to connect to his Asterisk
located at a data center.  His IAX Phone connects through his office NAT
gateway device (unfortunately I don't know the specific brand and model).
He can make calls just fine.  However, he seems to have issues receiving
calls.

I monitored his Asterisk for a bit and noticed that his IAX Phone, which
registers every 60 seconds, always registers from a different port.  Does
this indicate that 60 seconds is too long, and that the NAT is closing the
hole between registrations?

I will try to find out what type of router/gateway he is using and post it.

Any thoughts would be appreciated.

Steve

Steven Sokol
Owner/Manager
Sokol & Associates, LLC

Phone:  816.822.1807
IaxTel: 700.613.9004
Web:http://www.sokol-associates.com


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Re: [Asterisk-Users] e164.org

[Joe Baptista]

On Sun, 23 May 2004, Tony Hoyle wrote:

> Simon Dorfman wrote:
>
> > I wonder if someone can help me understand this.  Let's say I configure my
> > asterisk box to use e164 and then I try to call a phone number in Germany.
> > I'm in the U.S.A.  So if the number I'm calling in Germany is registered in
> > e164's dns, would my call be routed directly via their voip provider?  Or
> > directly to their asterisk box?  And would it be free?
>
>  From the looks of it, they're just a directory... it looks like their not
> running asterisk themselves.
>
> They use something called EnumLookup which I guess is some kind of
> plugin/script.  If the number you're calling is in their database, it calls
> the VOIP number directly, otherwise it calls the POTS number

No they just provide the dns - it's your equipment that does the
connecting.  Exampple - if you use my number 17058721310 and conver it to
enum format it will look like this:  0.1.3.1.2.7.8.5.0.7.1.e164.org

If you do a look up of that number - you'll see the associated enum
pointer. example: dig 0.1.3.1.2.7.8.5.0.7.1.e164.org. any

; <<>> DiG 9.2.2 <<>> 0.1.3.1.2.7.8.5.0.7.1.e164.org. any
;; global options:  printcmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 44741
;; flags: qr rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 9, ADDITIONAL: 4

;; QUESTION SECTION:
;0.1.3.1.2.7.8.5.0.7.1.e164.org.IN  ANY

;; ANSWER SECTION:
0.1.3.1.2.7.8.5.0.7.1.e164.org. 600 IN  TXT "Joe Baptista"
0.1.3.1.2.7.8.5.0.7.1.e164.org. 600 IN  NAPTR   100 10 "u" "E2U+IAX2" 
"!^\\+17058721310$!iax2:[EMAIL PROTECTED]/17058721310!" .

;; AUTHORITY SECTION:
e164.org.   600 IN  NS  alberta-2.bcwireless.net.
e164.org.   600 IN  NS  ns1.au1.com.au.
e164.org.   600 IN  NS  ns1.e164.org.
e164.org.   600 IN  NS  ns2.au1.com.au.
e164.org.   600 IN  NS  ns2.au1.net.
e164.org.   600 IN  NS  ns3.bcwireless.net.
e164.org.   600 IN  NS  apollo.bcwireless.net.
e164.org.   600 IN  NS  mutual.bcwireless.net.
e164.org.   600 IN  NS  alberta.bcwireless.net.

;; ADDITIONAL SECTION:
ns2.au1.net.172800  IN  A   202.87.28.2
mutual.bcwireless.net.  3600IN  A   198.231.65.11
alberta.bcwireless.net. 3600IN  A   209.115.243.234
alberta-2.bcwireless.net. 3600  IN  A   66.244.202.59

;; Query time: 343 msec
;; SERVER: 127.0.0.1#53(127.0.0.1)
;; WHEN: Sun May 23 10:15:37 2004
;; MSG SIZE  rcvd: 434

In this case systems doing enum lookups are told to contact th iax2 server
at phone.joebaptista.com use a guest login.

So basically enum is a facility which uses the dns to route calls in such
a way that they bypass the regular PSTN system if an internet routing and
supported protocol exists.

for more information on ENUM visit

http://www.rfc-editor.org/rfcsearch.html

and search for enum and e.164.

Now the proper domain for enum is e164.arpa - and the respective
assignments by the ITU of telephone delegations under e164.arpa has been
at best very lame.  As you can well imagine telephone companies have as a
rule refrained from adopting e164 because as I have mentioned above e164
facilitates the bypassing of PSTN via the internet and that would of
course mean lost revenue.

Check out http://www.itu.int/osg/spu/enum/ for more info on the politics
and bureacracy of enum.  You can get information on enum assignments and
registries at:

http://www.ripe.net/enum/request-archives/

At you can see there are very few official enum registries - so I think
what e164.org is doing in invaluable to the community.

regards
joe baptista

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Re: [Asterisk-Users] Sip proxy registration help

[Brian Potkin]

On Sun, May 23, 2004 at 07:21:11AM +0100, Rob Franklin wrote:

> 
> Hi All,
> 
> I have just installed Asterisk and am trying to connect it to a SIP
> account that I currently have with www.voiptalk.org but without any
> success.  Although I know that voiptalk do provide asterisk accounts I
> don't want to convert the SIP account until am happy that it's gonna
> work for me.  The asterisk box is currently behind a firewall and the
> following ports are being forwarded to it - UDP 5060, 1-2, 5036,
> 4569.  When it attempts to connect I get the following messages through
> the CLI prompt:
> 
> May 23 07:29:35 WARNING[1116941120]: chan_sip.c:595 retrans_pkt: Maximum
> retries exceeded on call [EMAIL PROTECTED] for
> seqno 102 (Critical Request)
> May 23 07:29:49 NOTICE[1116941120]: chan_sip.c:3597 sip_reg_timeout:
> Registration for '[EMAIL PROTECTED]' timed out, trying again
> May 23 07:29:55 WARNING[1116941120]: chan_sip.c:595 retrans_pkt: Maximum
> retries exceeded on call [EMAIL PROTECTED] for
> seqno 103 (Critical Request)
> May 23 07:30:09 NOTICE[1116941120]: chan_sip.c:3597 sip_reg_timeout:
> Registration for '[EMAIL PROTECTED]' timed out, trying again

Coincidentally, registering asterisk with voiptalk was to be my project
for today.  With only port 5060 open for udp and forwarded to the
machine asterisk runs on I've got as far as getting 'Registered' from
'sip show registry'.  My sip.conf is

[general]

port = 5060
bindaddr = 0.0.0.0
register => 84412416:[EMAIL PROTECTED]

Brian.


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RE: [Asterisk-Users] IAX2 NAT / Registration Issue

[Todd Lieberman]

His firewall is stateless.


I've run into the same issue w/the sonic wall firewall on a client site.

TL

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven Sokol
Sent: Sunday, May 23, 2004 11:06 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] IAX2 NAT / Registration Issue


I have a client using IAX Phone at his office to connect to his Asterisk
located at a data center.  His IAX Phone connects through his office NAT
gateway device (unfortunately I don't know the specific brand and model).
He can make calls just fine.  However, he seems to have issues receiving
calls.

I monitored his Asterisk for a bit and noticed that his IAX Phone, which
registers every 60 seconds, always registers from a different port.  Does
this indicate that 60 seconds is too long, and that the NAT is closing the
hole between registrations?

I will try to find out what type of router/gateway he is using and post it.

Any thoughts would be appreciated.

Steve

Steven Sokol
Owner/Manager
Sokol & Associates, LLC

Phone:  816.822.1807
IaxTel: 700.613.9004
Web:http://www.sokol-associates.com


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[Asterisk-Users] RE: snom reporting busy when it shouldn't

[nicolas]

No, they are not in DND mode and there is nothing strange on the displays
(sorry).

nicolas


Christian Stredicke wrote:

> Did you check if the phone is in DND state? Is there anything strange on
> the display?
> 
> CS
> 
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of nicolas
>> Sent: Sunday, May 23, 2004 5:43 AM
>> To: [EMAIL PROTECTED]
>> Subject: [Asterisk-Users] snom reporting busy when it shouldn't
>> 
>> I am using asterisk cvs.
>> 
>> Incoming/Outgoing calls are working.
>> Calling the phone when some other lines are in use on the phone is ok.
>> What does not work though is when the phone is ringing, nobody else can
>> call the phone anymore.
>> 
>> That's what * is saying:
>> 
>> -- Got SIP response 486 "Busy Here" back from 192.168.1.250
>> -- SIP/snom1-4a44 is busy
>> 
>> I am using the 2.05e snom200 firmware.
>> 
>> Snom people sad must run.
>> nico
>> 
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Re: [Asterisk-Users] T100P HDLC configuration

[Michael A Rowley]

Christian,

I think this is a YMMV, situation...  From what I have read in the Ham radio lists, hdlc is supposed to work _better_ on the <2.4.26 kernels, (they actually reccommend 2.6.5.  This support has apparently been backported to 2.4.26.)  but you must use the sethdlc utility 1.15.  The problem seems to stem from the fact that there are 3 versions of sethdlc out there...  The version in zaptel source is 1.13 (I think that's right) which would correspond to a 2.4.22 kernel, and the 1.11 version is pre 2.4.21...  If you are compiling the sethdlc version that comes with the zaptel source, then 2.4.22 would be correct.I am going to try the downloaded version from the sethdlc web site.  Their directions are rather explicit on which versions to use.

Of course, I am saying all this as second hand information.  I haven't had time to "put my money where my mouth is" AIW.  
As with most things, this will have to go with the caviat YMMV.  But I have gotten replies from people who have hdlc working with 2.6.5 kernels, and one I believe is using 2.4.26...

I will post my actual experience when I get time to work on HDLC.  For now, I will have to stay with the adtran to split networking and voice services...  Too many pots, not enough burners :(

Michael

On Sunday, May 23, 2004, at 08:13 AM, Christian Hoffmeyer wrote:

HDLC does not work on kernel 2.4.26

Use kernel 2.4.22

Christian Hoffmeyer
YottaDot Solutions
Huntsville, AL

Re: [Asterisk-Users] T100P HDLC configuration

[Vasyl Rublyov]

Thank you Michael,

I used that sethdlc which is in latest zaptel, sethdlc --version does
not work, but "sethdlc hdlc0 --version" works

sethdlc --version
--version: unable to get interface information: No such device

/sbin/sethdlc hdlc0 --version
sethdlc version 1.15
Copyright (C) 2000 - 2003 Krzysztof Halasa <[EMAIL PROTECTED]>

Today, I am going to try downgrade the kernel to 2.4.19, so it will use
old HDLC API.


Michael A Rowley wrote:
Vasyl,
  
  
What sethdlc version are you using do a sethdlc --version. You
should have 1.15 with kernel 2.4.26.
  
  
  
  
I am curious to see you get this working, as I need to work on this
next. But I have to get the phone system up and running first.
  
  
Michael.
  
  
  
On Saturday, May 22, 2004, at 10:09 PM, Vasyl Rublyov wrote:
  
  
  Thank you, Michael


I tried to switch to FR mode... but it did not help. I tied DLCI as 16
and 99... the same result.


I attached one more full config from Netopia and from my Linux+Zaptel
T100P systems.



  
Michael Rowley MD
  
FP
  
  
  



-- 
Thanks and regards,
  Vasyl Rublyov

Re: [Asterisk-Users] T100P HDLC configuration

[Vasyl Rublyov]

Christian,
Opss.. than where is the problem? Is it in kernel or is in zaptel driver?
Could you please let me know when has been broken and if anyone is 
working on the fix for it?

Thank you.
Christian Hoffmeyer wrote:
- Original Message - 
From: "Vasyl Rublyov" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 22, 2004 10:02 PM
Subject: Re: [Asterisk-Users] T100P HDLC configuration

Just would like to add, of course if it is going to help:
  I am using Linux 2.4.26 on Linux, compiled from sources and latest zaptel
sources.
  We have T1 Internet from Verizon

HDLC does not work on kernel 2.4.26
Use kernel 2.4.22
Christian Hoffmeyer
YottaDot Solutions
Huntsville, AL
(w)   256.851.8689
(c)256.655.0321
(iax)  700.859.4508
Ask me about Asterisk.
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[Asterisk-Users] Problems using Adtran 750 FXO and TE405P

[Patrick J. Conroy]

I was informed that I had inadvertently sent my last posting in HTML format.
I apologize for any trouble that this caused and I am re-posting in plain
text.  Any insight into what I am doing wrong here would be appreciated.

Thanks,
Patrick

Hello,

I am trying to get an Adtran 750 w/ 1 Quad FXO and 1 Quad FXS to work with a
TE405P and I am having a few problems.  I have the FXO on channels 1-4 and
the FXS on channels 5-8.  I have a single analog phone set connected to the
first port on the FXS (channel 5) and an analog line connected to the first
port of the FXO (channel 1).  The FXS sees to be working fine.  I can call
the demo server and back and forth with SIP phones, but I cannot get
anything to connect out to the CO line.

I added these lines to zaptel.conf:
span=1,0,0,esf,b8zs
fxsks=1-4
fxols=5-8
unused=9-24

I added these lines to zapata.conf:
context=local
group=1
signalling=fxs_ks
channel=>1-4

context=local
group=2
signalling=fxo_ls
channel=>5-8


I have also tried configuring channels 1-4 as fxsls and fxsgs, but nothing
seems to work.  BTW, the Adtran is brand new and according to the document
the FXO ports are automatically provisioned as FXO loop start.  I have
attempted to connect to the admin port to verify the provisioning, but after
getting no response after 8 minutes, I decided to trust the documentation.
Any suggestions would be greatly appreciated.

Thanks,
Patrick


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

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Re: [Asterisk-Users] RxFAX generates no tiff file

[Petr Grussmann]

I have same problem connected to PBX over E1 and sync and not slip I 
have latest version spanDSP

I receiving 1/3 pages from faxis
?
who is a problems-)
I
Steve Underwood wrote:
Hi Troy,
People had a lot of problems like this with earlier versions of 
spandsp. However, the latest version is pretty solid, and people are 
using it in high volume production applications. If you are getting 
these bad results with the latest version I would be interested to see 
the audio log file, so I can investigate the reason.

Regards,
Steve
Troy Settle wrote:
Dunno about not being able to generate a tiff, I got rxfax to do 
that, but
they're badly malformed.

http://roanoke-voip01.psknet.com/fax/

--
 Troy Settle
 Pulaski Networks
 http://www.psknet.com
 866.477.5638
 

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RE: [Asterisk-Users] *** Asterisk Sunday News: Conferences on the phone and IRL - "in real life"

[Leif Madsen]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Olle E. Johansson
> Sent: Sunday, May 23, 2004 8:23 AM
> To: Users Asterisk
> Subject: [Asterisk-Users] *** Asterisk Sunday News: Conferences on the
> phone and IRL - "in real life"
>
>
> *** Writing community documentation - asteriskdocs.org
> --
>Maybe we should produce a couple of Asterisk OpenOffice Impress
> presentations that we all can use on local Open Source community
> gatherings? I'll hand that idea over to the Asterisk Docs project.
> On the topic of docs, they have restarted and are having regular MeetMe
> phone conferences. They need your help, we need more people writing
> documentation, making sure we're not adding functions without
> proper documentation, explaining to new users how great
> Asterisk really is behind all the code and cryptic commands.
> 
> The Asterisk Docs project is awarded my "Asterisk project of the week"
> award. They have the worst web site, but is a great project to join.
> (Yes, I've seen the new web layout suggestions, they're great :-)
> 
> - http://www.asteriskdocs.org
> - Mailing list and IRC channel is available

Just wanted to make a quick comment about the asterisk docs project.

Work seems to starting to progress forward, but we are desperately in need
of people to help document, especially the more advanced sections of
Asterisk.

I'll try not to go into too much depth here.  We are currently trying to
focus on the direction of book and what our ultimate goals are, plus getting
a better idea about how to write documentation.  Jared has done a fantastic
job starting to outline chapter 4 of the book (which we feel at this point
is the most important starting point for anyone starting with Asterisk).
After a hiatus of approximately 4 months (I was back in a very dense course
curriculum so I was unable to do much work on the book) we are active again.
Please join us in #asterisk-doc to discuss any sections you wish to work on,
or join the mailing list and participate.  It's not nearly the same volume
of traffic as asterisk-users, so if you are on this list, might as well just
join asterisk-docs as well! :)

On the note regarding the new website, sypher has developed us a fantastic
layout that jsmith and his wife are working on putting together.  Exomorph
is working on putting together the CMS for us, so I figure within' the next
week or so http://www.asteriskdocs.org should have a new look and feel
allowing us to update content much my dynamically.

For now, I will be attempting to spend some time on writing docs.  Maybe
some people can join us!  This Tuesday we have a tentative
conference/discussion session setup to talk about what we have been working
on for the week.

Thanks for letting me use some your bandwidth,
Leif Madsen aka blitzrage.

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Re: [Asterisk-Users] SIP with TerraCall Error

[Karl Brose]

Try using the IP address below directly and not a hostname.
The follow works for me.
[terracall]
type=friend
host=64.69.76.33
username=##x##
secret=
fromuser=##x##
fromdomain=pc.tt.xten.net
nat=yes
context=terracall-inbound

[EMAIL PROTECTED] wrote:
Dear All,
I had try the new cvs version asterisk to connect to TerraCall, but fail with
the follow reply, anyone know how to solve this problem.
NOTICE[1133742896]: Failed to authenticate on INVITE to '"account number"
;tag=as1d02a70e'
Thank You.
Cary LEUNG
Administrator
CARYNET Information Center
Hong Kong
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[Asterisk-Users] extension pattern matching

[Graham Turner]

dear all, was hoping someone could give me instruction on the syntax of
extension pattern matching for letters

the proposed 'dial plan' is one where any letter in the dialled digits
causes the pbx to assume we are dilaling a sip url and as such forward to
the appropraite sip service provider

was hoping to avoid the plan in john todd's example that assumes anything
prefixed with 3 is a sip address

gt

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[Asterisk-Users] ztdummy - how to test?

[Tony Hoyle]

I've modified ztdummy to work under 2.6 (basically ditched all the uhci stuff
and added a kernel timer instead).
How do I test my changes are doing anything useful?  zttest gives:
--- Results after 14 passes ---
Best: 99.975586 -- Worst: 99.975586
Is that good/bad/terrible...?
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle <[EMAIL PROTECTED]>  Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2  02B8 E547 470E 4F4B 6917
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RE: [Asterisk-Users] RxFAX generates no tiff file

[Sam Bingner]

You should Answer() your calls...  In the 5000 exten, you could move your
Answer to after the dial if you like... And the h exten hangs up if it
doesn't exist so that's redundant, but not bad

Sam

[internalexten]
exten => 5000,1,Answer()
exten => 5000,2,Dial(SIP/mike,60,tr)
exten => 5000,3,SetLanguage(de)
exten => 5000,4,Playback(vm-nobodyavail)

exten => 6000,1,Answer()
exten => 6000,2,WaitMusicOnHold(30)

exten => 7000,1,Answer()
exten => 7000,2,rxfax(/tmp/testfax.tif)

[default]
include => internalexten
exten => h,1,Hangup

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Heininger
Sent: Sunday, May 23, 2004 1:41 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RxFAX generates no tiff file


Am 23.05.2004 um 04:33 schrieb Steve Underwood:

> How do you run rxfax? You problem is probably something to do with
> that. Your's is the first report I have had of no TIFF file
> whatsoever.

[internalexten]
exten => 5000,1,Dial(SIP/mike,60,tr)
exten => 5000,2,SetLanguage(de)
exten => 5000,3,Playback(vm-nobodyavail)

exten => 6000,1,WaitMusicOnHold(30)
exten => 7000,1,rxfax(/tmp/testfax.tif)

[default]
include => internalexten
exten => h,1,Hangup


The context for the inbound call is [default] and goes to extension
7000.


TIA,
Mike

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smime.p7s
Description: S/MIME cryptographic signature

RE: [Asterisk-Users] extension pattern matching

[Sam Bingner]

I think may be able to do that with
_[a-z][a-z].

But I haven't tried it, you need to use 2 to make sure you don't overwrite
the system extensions.  As I understand the * regex implimentation, you
can't do _.[a-z]. to match any letters in dialplan anywhere, but that is
what you really wanted I think.

You could always code it in ;)
Sam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Graham Turner
Sent: Sunday, May 23, 2004 9:09 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] extension pattern matching


dear all, was hoping someone could give me instruction on the syntax of
extension pattern matching for letters

the proposed 'dial plan' is one where any letter in the dialled digits
causes the pbx to assume we are dilaling a sip url and as such forward to
the appropraite sip service provider

was hoping to avoid the plan in john todd's example that assumes anything
prefixed with 3 is a sip address

gt

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RE: [Asterisk-Users] ZAPTEL not loading on FC2

[Sam Bingner]

Change your symlink to not point to the linux source tree, but rather
point at /lib/modules/2.6.5-358/build, and just do a make linux26

Or apply this patch to your makefile...

Sam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Taz Man
Sent: Sunday, May 23, 2004 4:57 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] ZAPTEL not loading on FC2


Hello all,
I've just installed the Fedora core 2 and tried to compile the asterisk
and the zaptel drivers Asterisk went smooth but I had troubles with the
zaptel. I did copy the .config file under the kernel source and make
oldconfig and make include/asm ; make include/version.h ; make
SUBDIRS=scripts I was able to compile the zaptel source, using the make
linux26 and then ran make install. I see that the .ko files are in place
(under
/lib/modules/2.6.5-1.358/misc/...) but when trying to load the zaptel I
get:

[EMAIL PROTECTED] misc]# modprobe zaptel
FATAL: Error inserting zaptel (/lib/modules/2.6.5-1.358/misc/zaptel.ko):
Invalid module format

[EMAIL PROTECTED] misc]# modprobe wcfxo
WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358/misc/zaptel.ko):
Invalid module format
WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358/misc/zaptel.ko):
Invalid module format
FATAL: Error inserting wcfxo (/lib/modules/2.6.5-1.358/misc/wcfxo.ko):
Invalid module format
FATAL: Error running install command for wcfxo

uname -a gives
Linux server 2.6.5-1.358 #1 Sat May 8 09:04:50 EDT 2004 i686 i686 i386
GNU/Linux

any help? any Ideas?
10x, Ronen


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makefile-kernel26.diff
Description: Binary data


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Re: [Asterisk-Users] T100P HDLC configuration

[Vasyl Rublyov]

All,
Just now I tried Linux kernel 2.4.19, with old sethdlc utility... 
everything works, so the problem seems in the zaptel driver or HDLC 
implementation of Linux.
I really can't think the problem is in Linux kernel sources because it 
has passed for 6 releases since they released new HDLC API... so the 
problem might be in zaptel driver.

I had to hack the source code, so it not uses new API for 2.4.19 - does 
not have hdlc_close, it should be hdlc->stop(hdlc) call.

The next issue which is quite important... what should be fixed so T100P 
will work on new kernel. I am willing to volunteer and try to fix it, 
only wish to confirm that kernel >= 2.4.20 works for other T1/E1 cards 
with Cisco HDLC protocol.

Vasyl Rublyov wrote:
Christian,
Opss.. than where is the problem? Is it in kernel or is in zaptel driver?
Could you please let me know when has been broken and if anyone is 
working on the fix for it?

Thank you.
Christian Hoffmeyer wrote:
- Original Message - From: "Vasyl Rublyov" 
<[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 22, 2004 10:02 PM
Subject: Re: [Asterisk-Users] T100P HDLC configuration

Just would like to add, of course if it is going to help:
  I am using Linux 2.4.26 on Linux, compiled from sources and latest 
zaptel
sources.
  We have T1 Internet from Verizon


HDLC does not work on kernel 2.4.26
Use kernel 2.4.22
Christian Hoffmeyer
YottaDot Solutions
Huntsville, AL
(w)   256.851.8689
(c)256.655.0321
(iax)  700.859.4508
Ask me about Asterisk.
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Thanks and regards,
 Vasyl Rublyov
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RE: [Asterisk-Users] VoicePulse SIP

[Michael Graves]

Hfailed to add the illustrative bit about my installationI
DO have an X100p in my * box. I'm not using it for anything more than a
timing source since I'm not happy with it as an FXO. I've just recently
sarted playing with the Sipura SPA-3000 as an FXO.

Michael


On Sat, 22 May 2004 20:07:27 +0800, Lars Boegild Thomsen wrote:

>H - can anybody confirm this.  I have generally had little luck with IAX
>in any case so I must admit I assumed (due to info from www.voip-info.org)
>that it was due to lack of timing device.  I have actually not tried to do
>any trunking - just normal calls.
>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] Behalf Of Chris A.
>> Icide
>> Sent: 22 May 2004 13:26
>> To: [EMAIL PROTECTED]
>> Subject: Re: [Asterisk-Users] VoicePulse SIP
>>
>>
>> Lars,
>>
>> I could be quite wrong, but I think you only need a 'timing'
>> source if you
>> want to use trunking over IAX.  You can still use IAX without trunking if
>> you don't have any sort of timing device.
>>
>> -Chris
>>
>> On 06:39 PM 5/21/2004, Lars Boegild Thomsen wrote:
>>  >Dear Sirs,
>>  >
>>  >Anybody ever tried running SIP up against Voicepulse?  On their
>>  >http://connect.voicepulse.com they claim they support both SIP
>> and IAX, but
>>  >I can't seem to get SIP running.  I have as mentioned before on
>> this list -
>>  >huge problems getting any timing devices running on some of my
>> machines, so
>>  >IAX is not really an option right now.  If I try I get a "Service
>>  >Unavailable" back from gw5.voicepulse.com.  If I try IAX2 with the same
>>  >settings, the call goes through - but sound is horrible.
>>  >
>>  >Regards,
>>  >
>>  >Lars...
>>  >
>>  >--
>>  >Lars Boegild Thomsen
>>  >Technical Director
>>  >JustIT Sdn. Bhd.
>>  >Cell Phone (MY): +60 (16) 323 1999
>>  >ICQ: 6478559
>>  >Yahoo Chat: [EMAIL PROTECTED]
>>  >MSN Chat: [EMAIL PROTECTED]
>>  >http://www.justit.ws
>>  >Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY)
>>  >Fax  : +60 (3) 2057 2647 (MY)
>>  >
>>  >___
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RE: [Asterisk-Users] VoicePulse SIP

[Michael Graves]

I use VoicePulse Connect, have done for about 6 months. I have no
problems with audio quality relating to the fact that I use IAX2 as the
connection protocol. I have had issues with QoS and codecs, but these
were issues at my end. I've recently started trying iLBC instead of
GSM. 

Michael


On Sat, 22 May 2004 20:07:27 +0800, Lars Boegild Thomsen wrote:

>H - can anybody confirm this.  I have generally had little luck with IAX
>in any case so I must admit I assumed (due to info from www.voip-info.org)
>that it was due to lack of timing device.  I have actually not tried to do
>any trunking - just normal calls.
>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] Behalf Of Chris A.
>> Icide
>> Sent: 22 May 2004 13:26
>> To: [EMAIL PROTECTED]
>> Subject: Re: [Asterisk-Users] VoicePulse SIP
>>
>>
>> Lars,
>>
>> I could be quite wrong, but I think you only need a 'timing'
>> source if you
>> want to use trunking over IAX.  You can still use IAX without trunking if
>> you don't have any sort of timing device.
>>
>> -Chris
>>
>> On 06:39 PM 5/21/2004, Lars Boegild Thomsen wrote:
>>  >Dear Sirs,
>>  >
>>  >Anybody ever tried running SIP up against Voicepulse?  On their
>>  >http://connect.voicepulse.com they claim they support both SIP
>> and IAX, but
>>  >I can't seem to get SIP running.  I have as mentioned before on
>> this list -
>>  >huge problems getting any timing devices running on some of my
>> machines, so
>>  >IAX is not really an option right now.  If I try I get a "Service
>>  >Unavailable" back from gw5.voicepulse.com.  If I try IAX2 with the same
>>  >settings, the call goes through - but sound is horrible.
>>  >
>>  >Regards,
>>  >
>>  >Lars...
>>  >
>>  >--
>>  >Lars Boegild Thomsen
>>  >Technical Director
>>  >JustIT Sdn. Bhd.
>>  >Cell Phone (MY): +60 (16) 323 1999
>>  >ICQ: 6478559
>>  >Yahoo Chat: [EMAIL PROTECTED]
>>  >MSN Chat: [EMAIL PROTECTED]
>>  >http://www.justit.ws
>>  >Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY)
>>  >Fax  : +60 (3) 2057 2647 (MY)
>>  >
>>  >___
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Pixel Power Inc. [EMAIL PROTECTED]

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[Asterisk-Users] creating a single user voice mail box on asterisk?

[hank]

hello how do I go create a single boice mail box on asterisk?
thanks
hank
- -
Don't judge me because I'm blind. Judge me by what's inside. if you judge me
because I am blind, then it is you who is blind.
"time is the fire in which we burn," Tollian Soran.
"grudges aren't worth holding--One who holds them shows his self-weakness."
Contact info:
[EMAIL PROTECTED]
Email: Same as MSN.

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[Asterisk-Users] Asterisk Prepaid

[usedcanon]

I have a requirement for a setup with prepaid call credits.

I am aware of the two applications available (been researching for the past
week), app_prepaid and app_rateengine. However neither of the two sound like
exactly what I want. However I was wondering that someone who has used it
might be able to say if they could be used in my scenario.

Basically my scenario is pretty straight forward. Credit will be allocated
to the ddi, I dont need any announcements etc (maybe low credit warning
during call could be useful thoug). From the users prespective everything
will be transparent. However the call should disconnect when the credit runs
out. The CDR and the account DB need to be adjusted according to the call
made.

My guess is that app_prepaid could used with modification, I am assuming
here that this is not possible as-is with configuration.

Basically in case of the prepaid app, the card number can be replace
transparently with the callerID.

All help, guidence and comments will be extremelly appreciated.

Umar.

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[Asterisk-Users] NetJet and RAS

[Andrew Yager]

Hi,
Last weekend I was planning to buy a physical PBX system, but instead I 
have been blown away by the fact that VoIP really works, that Asterisk 
is so easy to set up and use... and free!

We're in Australia, so as I understand it, we aren't allowed to use the 
Zaptel cards. We need to set up our system to route incoming 56K data 
calls to the PPP daemon on our Linux box or to our Windows 2003 server. 
I noticed the zapras application - is there any way to achieve this 
functionality using the NetJet card?

What other solutions might exist for us? (I really would rather use 
VoIP than a silly PBX...)

Thanks in advance,
Andrew
_
Andrew Yager
Real World Technology Solutions
Real People, Real SolUtions (tm)
http://www.rwts.com.au/
_

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Re: [Asterisk-Users] IAX2 REACHABLE/UNREACHABLE

[Karl Brose]

The asterisk qualify option in sip or iax
sends a test packet to the remote host every minute
and measures the time it takes for a response to
come back. If this time frame is less than what is
configured on the qualify statement or 2000ms if
it is 'yes' than the host is flagged unreachable until
pings succeed again. In SIP the packet is an
OPTIONS request and any response is accepted.
So if you're having this problem with several of
your systems at the same time, it may point to
a problem or interruption in your network services.
Do you have any hosts or gateways on dynamic IPs
that get refreshed every 8 min?

John Blackman wrote:
All,
I have an issue with IAX that I can’t comprehend. Approximately every 
eight minutes my servers go unreachable. They stay unreachable for 
exactly 10ms. I have two servers running IAX and it happens on both 
servers simultaneously. I have searched the archives and see similar 
issues, but not the exact same one. I am on the current CVS stable 
version of *.

Also, during IAX calls, every couple of minutes there is about a 
second of silence. Silence suppression is turned off on all phones 
(Grandstream BT101s). This issue does not occur during SIP calls.

I do have a similar registration problem with SIP. There are two 
phones on one site that go reachable/unreachable. (See extension  
in the sample below. This is a B101 that is registered to my server, 
but it happens to be at the same location as the second * server.)

If anyone can point me in the right direction for troubleshooting, I 
would appreciate it. I am pasting example error messages that I 
receive below.

TIA,
John
*CLI Output:*
May 23 12:50:05 NOTICE[7176]: chan_iax2.c:5616 iax2_poke_noanswer: 
Peer 'arlington' is now UNREACHABLE!

May 23 12:50:15 NOTICE[7176]: chan_iax2.c:5213 socket_read: Peer 
'arlington' is now REACHABLE!

May 23 12:58:19 NOTICE[7176]: chan_iax2.c:5616 iax2_poke_noanswer: 
Peer 'arlington' is now UNREACHABLE!

May 23 12:58:29 NOTICE[7176]: chan_iax2.c:5213 socket_read: Peer 
'arlington' is now REACHABLE!

May 23 13:06:07 NOTICE[5126]: chan_sip.c:6901 sip_poke_noanswer: Peer 
'' is now UNREACHABLE!

May 23 13:06:34 NOTICE[7176]: chan_iax2.c:5616 iax2_poke_noanswer: 
Peer 'arlington' is now UNREACHABLE!

May 23 13:06:44 NOTICE[7176]: chan_iax2.c:5213 socket_read: Peer 
'arlington' is now REACHABLE!

May 23 13:14:49 NOTICE[7176]: chan_iax2.c:5616 iax2_poke_noanswer: 
Peer 'arlington' is now UNREACHABLE!

May 23 13:14:59 NOTICE[7176]: chan_iax2.c:5213 socket_read: Peer 
'arlington' is now REACHABLE!

May 23 13:17:56 NOTICE[5126]: chan_sip.c:5860 handle_response: Peer 
'' is now REACHABLE!

*Iax.conf**:*
[general]
;port=5036 ;I commented this out because I discovered it is hard coded 
anyway.

bindaddr=10.20.30.10
amaflags=default
accountcode=default
bandwidth=low
disallow=all
allow=gsm
jitterbuffer=no
;dropcount=3
;maxjitterbuffer=500
;maxexcessbuffer=100
;trunkfreq=20
tos=lowdelay
authdebug=no
register => username:[EMAIL PROTECTED] ;these are real values 
that I changed for security reasons

[arlington]
type = friend
context = local
callerid =
auth = plaintext
secret = password
;inkeys =
;outkey =
host = dynamic
;defaultip =
;accountcode =
qualify = yes
;mailbox =
trunk = yes
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RE: [Asterisk-Users] NetJet and RAS

[Simon Brown]

I, too, am in Australia.  I have used the X100P card and now have recently
swapped to use the TDM400P card with one FXS and one FXO.  Others in
Australia are also using Asterisk with the Zaptel cards.

Regards,

Simon Brown

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Yager
Sent: Monday, 24 May 2004 9:37
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] NetJet and RAS

Hi,

Last weekend I was planning to buy a physical PBX system, but instead I have
been blown away by the fact that VoIP really works, that Asterisk is so easy
to set up and use... and free!

We're in Australia, so as I understand it, we aren't allowed to use the
Zaptel cards. We need to set up our system to route incoming 56K data calls
to the PPP daemon on our Linux box or to our Windows 2003 server. 
I noticed the zapras application - is there any way to achieve this
functionality using the NetJet card?

What other solutions might exist for us? (I really would rather use VoIP than
a silly PBX...)

Thanks in advance,

Andrew

_
Andrew Yager
Real World Technology Solutions
Real People, Real SolUtions (tm)
http://www.rwts.com.au/
_
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Re: [Asterisk-Users] NetJet and RAS

[Andrew Yager]

Thanks! That's good to know. Please excuse my ignorance - if we have 
two telstra ISDN2 lines, which card should I get?

Thanks,
Andrew
On 24/05/2004, at 9:53 AM, Simon Brown wrote:
I, too, am in Australia.  I have used the X100P card and now have 
recently
swapped to use the TDM400P card with one FXS and one FXO.  Others in
Australia are also using Asterisk with the Zaptel cards.

Regards,
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew 
Yager
Sent: Monday, 24 May 2004 9:37
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] NetJet and RAS

Hi,
Last weekend I was planning to buy a physical PBX system, but instead 
I have
been blown away by the fact that VoIP really works, that Asterisk is 
so easy
to set up and use... and free!

We're in Australia, so as I understand it, we aren't allowed to use the
Zaptel cards. We need to set up our system to route incoming 56K data 
calls
to the PPP daemon on our Linux box or to our Windows 2003 server.
I noticed the zapras application - is there any way to achieve this
functionality using the NetJet card?

What other solutions might exist for us? (I really would rather use 
VoIP than
a silly PBX...)

Thanks in advance,
Andrew
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[Asterisk-Users] Zapata.conf setup for TE410P

[William Zhang]

Hi,

I have a TE410P with 3 E1 being enabled, some how it crashes for 2
times lately,  I suspect it might be the channel setup issue, can
anyone tell me if following part in zapata.conf is correct?

switchtype = euroisdn
signalling = pri_cpe
pridialplan=local
group = 1
context = incoming
channel => 1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124

Also, is there way to log the reason why Asterisk is crashed? Thank
you.

B
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Re: [Asterisk-Users] RxFAX generates no tiff file

[Steve Underwood]

Hi Petr,
For most people who are sure they have no frame slips, the problem 
usually turns out to be frame slips :-)

If you are *really* sure you do not have frame slips, then uncomment the 
first line in t30.c, and rebuild and reinstall spandsp. The when you 
exchange a fax you should end up with a pair of audio files in your /tmp 
directory - one for the transmit signal and one for the receive signal. 
Send those to me, and I will investigate.

Regards,
Steve
Petr Grussmann wrote:
I have same problem connected to PBX over E1 and sync and not slip I 
have latest version spanDSP

I receiving 1/3 pages from faxis
?
who is a problems-)
I
Steve Underwood wrote:
Hi Troy,
People had a lot of problems like this with earlier versions of 
spandsp. However, the latest version is pretty solid, and people are 
using it in high volume production applications. If you are getting 
these bad results with the latest version I would be interested to 
see the audio log file, so I can investigate the reason.

Regards,
Steve
Troy Settle wrote:
Dunno about not being able to generate a tiff, I got rxfax to do 
that, but
they're badly malformed.

http://roanoke-voip01.psknet.com/fax/

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RE: [Asterisk-Users] NetJet and RAS

[Simon Brown]

Look at http://www.voip-info.org/tiki-index.php?page=Asterisk, it has lots of
good information.
http://www.voip-info.org/wiki-Asterisk+Hardware in particular covers hardware
such as cards.

Simon 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Yager
Sent: Monday, 24 May 2004 9:57
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NetJet and RAS

Thanks! That's good to know. Please excuse my ignorance - if we have two
telstra ISDN2 lines, which card should I get?

Thanks,
Andrew

On 24/05/2004, at 9:53 AM, Simon Brown wrote:

> I, too, am in Australia.  I have used the X100P card and now have 
> recently swapped to use the TDM400P card with one FXS and one FXO.  
> Others in Australia are also using Asterisk with the Zaptel cards.
>
> Regards,
>
> Simon Brown
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Andrew 
> Yager
> Sent: Monday, 24 May 2004 9:37
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] NetJet and RAS
>
> Hi,
>
> Last weekend I was planning to buy a physical PBX system, but instead 
> I have been blown away by the fact that VoIP really works, that 
> Asterisk is so easy to set up and use... and free!
>
> We're in Australia, so as I understand it, we aren't allowed to use 
> the Zaptel cards. We need to set up our system to route incoming 56K 
> data calls to the PPP daemon on our Linux box or to our Windows 2003 
> server.
> I noticed the zapras application - is there any way to achieve this 
> functionality using the NetJet card?
>
> What other solutions might exist for us? (I really would rather use 
> VoIP than a silly PBX...)
>
> Thanks in advance,
>
> Andrew
>
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[Asterisk-Users] Fwd: regulating voip - aca

[Duane]

The Australian Government must be feeling left out and want to stick 
their noses in...

...
ACA moves to regulate VoIP
Rodney Gedda , Computerworld
20/05/2004 10:07:07
Mid-2005 will herald a new era in voice over IP telecommunications when 
the Australian Communications Authority (ACA) introduces specific 
regulations for service providers and enterprises stipulating how the 
technology must be delivered.

The ACA’s acting chairman Dr Bob Horton said VoIP regulation is 
“inevitable”.
“VoIP is novel and needs to be treated with flexibility for a while as 
we are faced with an innovative approach to telecommunications and want 
to make sure we don’t stifle innovation,” Horton told Computerworld.

“We need to test the boundaries of existing regulation and then fashion 
something around VoIP. Regulation is inevitable because whoever is 
carrying it has obligations for data, voice, or whatever and there needs 
to be a requirement for universal service.”

Horton said a discussion paper covering the regulation of VoIP will be 
put down as early as December this year when the ACA will allow “a 
couple of months” for all parties to ponder its implications.

“We will then draw a set of regulatory conditions from that,” he said. 
“Around February 2005 the recommendations will be put to the industry 
and the complete regulation guidelines should be finalised by July next 
year.”

Horton is also adamant that VoIP regulation will not be an impediment to 
the “due diligence process” already in place between the ACA and industry.

“There is a period of tolerance because we’d like to see 
experimentation,” he said. “And the ACA thanks the early entrants.

"There is an atmosphere of industry self-regulation so we’re giving them 
as much flexibility as possible.”
As to the type of conditions VoIP regulations might impose upon service 
providers, Horton said the three areas of quality of service, call 
location, and privacy will be considered, along with existing carrier 
guidelines such as what will happen in the event of a power blackout and 
how access will be provided for people with disabilities.

“One concern would be to give customers an understanding of what they 
are buying,” he said. “Many VoIP service providers are positioning 
[their products] as a telephone replacement so if there is any 
difference – let them know! VoIP has distinguishing characteristics, for 
example it’s transportable, so for emergency calls you wouldn’t know 
where it’s coming from.”

Furthermore, Horton said if VoIP calls are made to non-VoIP phones the 
regulations may need to stipulate a new number range so end users “can 
expect a low quality of service” which will be part of the consideration 
in December.

Regulations for enterprises running VoIP over their internal networks, 
Horton said are difficult to predict. "It’s like a private network and 
that’s not something new but if you have a transmission link you have to 
become a carrier.”

He also stressed that in addition to forthcoming regulations, rogue VoIP 
operations will not be tolerated.
“The industry would detect odd players and we would move to close them 
down,” he said. “VoIP will spur a number of new carriers as some 700 
[data] service providers are poised to offer voice.”

Individual calls are unlikely to be charged or taxed separately, Horton 
said, as regulations do not allow for this in keeping with the 
government’s stance on equal access.

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
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Re: [Asterisk-Users] NetJet and RAS

[Gonzalo Servat]

On Mon, 2004-05-24 at 09:57 +1000, Andrew Yager wrote:
> Thanks! That's good to know. Please excuse my ignorance - if we have 
> two telstra ISDN2 lines, which card should I get?

A somewhat reasonably priced ISDN card that works with Asterisk and is
sold in Australia is the AVM Fritz:

   http://www.avm.de/en/partners/distributors/AUS/index.html

Hope this helps.

Regards,
Gonzalo

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Re: [Asterisk-Users] NetJet and RAS

[Adam Hart]

Andrew Yager wrote:
Hi,
Last weekend I was planning to buy a physical PBX system, but instead I 
have been blown away by the fact that VoIP really works, that Asterisk 
is so easy to set up and use... and free!

We're in Australia, so as I understand it, we aren't allowed to use the 
Zaptel cards. 
By not allowed, you mean not Austel approved - then yes, you shouldn't 
use it but it will still work.
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Re: [Asterisk-Users] RxFAX generates no tiff file

[Darren Nickerson]

Steve,

We have no frame slips, so we probably have a frame slip problem ;-)

This may be plaguing us on our faxing. Gain and echo cancelation (ie: none)
are all approximately correct, and yet still we cannot get reliable faxing
through the POTS lines plugged into our FXO card on the Adit (whereas we can
fax well when using the POTS lines directly). Faxing T1 -> T1 via a TE405P
works well, ... it's only when we try to use the connection to the Adit (24
fxs_ks channels in a T1) that things go horribly wrong.

Is there any sure-fire way to detect frame slips? I see a counter for IRQ
misses with zttool, but that's all. In my Adit600 I see lots of measures of
errors (line errored seconds, controlled slip seconds, bursty errored
seconds etc) but they're all zero.

Am I missing an obvious way to detect./observe these events?

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
iFax Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638
+1.215.243.8335 (fax)

- Original Message - 
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, May 23, 2004 8:03 PM
Subject: Re: [Asterisk-Users] RxFAX generates no tiff file


> Hi Petr,
>
> For most people who are sure they have no frame slips, the problem
> usually turns out to be frame slips :-)
>
> If you are *really* sure you do not have frame slips, then uncomment the
> first line in t30.c, and rebuild and reinstall spandsp. The when you
> exchange a fax you should end up with a pair of audio files in your /tmp
> directory - one for the transmit signal and one for the receive signal.
> Send those to me, and I will investigate.
>
> Regards,
> Steve
>
>
> Petr Grussmann wrote:
>
> > I have same problem connected to PBX over E1 and sync and not slip I
> > have latest version spanDSP
> >
> > I receiving 1/3 pages from faxis
> >
> > ?
> > who is a problems-)
> >
> >
> > I
> > Steve Underwood wrote:
> >
> >> Hi Troy,
> >>
> >> People had a lot of problems like this with earlier versions of
> >> spandsp. However, the latest version is pretty solid, and people are
> >> using it in high volume production applications. If you are getting
> >> these bad results with the latest version I would be interested to
> >> see the audio log file, so I can investigate the reason.
> >>
> >> Regards,
> >> Steve
> >>
> >>
> >> Troy Settle wrote:
> >>
> >>> Dunno about not being able to generate a tiff, I got rxfax to do
> >>> that, but
> >>> they're badly malformed.
> >>>
> >>> http://roanoke-voip01.psknet.com/fax/
> >>
>
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[Asterisk-Users] Serious NAT problems: can't call between lines on sipura

[Bruce Komito]

I have a problem that is almost certainly nat-related, but I can't figure
out what's happening.

Since moving the Sipura behind a NAT server (Linksys), I am no longer able
to call between the two lines on the same Sipura.  When I dial one
extension from the other, it rings, but immediately after I pick up the
ringing phone, the call is uncerimoniously dumped.  I can tell the call
terminates immediately because I am watching the CDRs come out.  The *
server is on a public address with no firewall between it and the outside
world.

sip.conf: (both extensions have identical settings)
; Bruce
[5815]
type=friend
username=5815
secret=wpti5815
host=dynamic
[EMAIL PROTECTED]
context=vpbx-wpti
qualify=3000
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
nat=yes

I'm thinking this has something to do with a setting in the Sipura, but I
don't know where to start.  I have nat keep-alive turned on, but I had to
turn stun off because it was causing a long, inexplicable delay after
dialing before the call would complete.

I'm realizing NAT with VoIP is a real problem.  Anyone have a silver
bullet they wish to share?

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115



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Re: [Asterisk-Users] NetJet and RAS

[Andrew Yager]

Thanks to everyone who has replied
I'm thinking that maybe I need to take a step back and ask a more basic 
question - Is it possible to get a 56K data connection (from a normal 
PSTN line, dialled with a telephone number, causing a telephone to ring 
at your end) to work when dialing to an ISDN line?

I'm assuming the answer is yes...
Is there a better place for me to direct my (somewhat ignorant) 
questions?

Thanks,
Andrew
On 24/05/2004, at 10:04 AM, Simon Brown wrote:
Look at http://www.voip-info.org/tiki-index.php?page=Asterisk, it has 
lots of
good information.
http://www.voip-info.org/wiki-Asterisk+Hardware in particular covers 
hardware
such as cards.

Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew 
Yager
Sent: Monday, 24 May 2004 9:57
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NetJet and RAS

Thanks! That's good to know. Please excuse my ignorance - if we have 
two
telstra ISDN2 lines, which card should I get?

Thanks,
Andrew
On 24/05/2004, at 9:53 AM, Simon Brown wrote:
I, too, am in Australia.  I have used the X100P card and now have
recently swapped to use the TDM400P card with one FXS and one FXO.
Others in Australia are also using Asterisk with the Zaptel cards.
Regards,
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Yager
Sent: Monday, 24 May 2004 9:37
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] NetJet and RAS
Hi,
Last weekend I was planning to buy a physical PBX system, but instead
I have been blown away by the fact that VoIP really works, that
Asterisk is so easy to set up and use... and free!
We're in Australia, so as I understand it, we aren't allowed to use
the Zaptel cards. We need to set up our system to route incoming 56K
data calls to the PPP daemon on our Linux box or to our Windows 2003
server.
I noticed the zapras application - is there any way to achieve this
functionality using the NetJet card?
What other solutions might exist for us? (I really would rather use
VoIP than a silly PBX...)
Thanks in advance,
Andrew
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_
Andrew Yager
Real World Technology Solutions
Real People, Real SolUtions (tm)
ph: (02) 9945 2567 fax: (02) 9945 2566
mob: 0405 15 2568
http://www.rwts.com.au/
_


smime.p7s
Description: S/MIME cryptographic signature

[Asterisk-Users] HELP!!! How do I move voicemail files to a new machine?

[Paul Mahler]

I copied voicemail files to a replacement system. When vm tries to play the
file * throws an error messages:
 
Unexpected header size 16
unable to open fd on /
 
How can I copy the VM to the new machine?
 
Thanks!
 
Paul
 
 
 

Paul Mahler 
[EMAIL PROTECTED] 
 
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
 94107-1901

 Asterisk Services and Training



 

 

 


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Re: [Asterisk-Users] Serious NAT problems: can't call between lines on sipura

[Brian Cuthie]

Bruce,
I think this is related to your firewall. You may want to take a look a 
posting I did a few weeks ago.

http://lists.digium.com/pipermail/asterisk-users/2004-May/046511.html
Something on this topic probably belongs in the wiki.
-brian
Bruce Komito wrote:
I have a problem that is almost certainly nat-related, but I can't figure
out what's happening.
Since moving the Sipura behind a NAT server (Linksys), I am no longer able
to call between the two lines on the same Sipura.  When I dial one
extension from the other, it rings, but immediately after I pick up the
ringing phone, the call is uncerimoniously dumped.  I can tell the call
terminates immediately because I am watching the CDRs come out.  The *
server is on a public address with no firewall between it and the outside
world.
sip.conf: (both extensions have identical settings)
; Bruce
[5815]
type=friend
username=5815
secret=wpti5815
host=dynamic
[EMAIL PROTECTED]
context=vpbx-wpti
qualify=3000
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
nat=yes
I'm thinking this has something to do with a setting in the Sipura, but I
don't know where to start.  I have nat keep-alive turned on, but I had to
turn stun off because it was causing a long, inexplicable delay after
dialing before the call would complete.
I'm realizing NAT with VoIP is a real problem.  Anyone have a silver
bullet they wish to share?
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115

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[Asterisk-Users] Sipura SPA-3000 Beta

[Michael Graves]

Hi All,

I'm on of those brave souls who bought into the preproduction beta of
the Sipura SPA-3000 FXS/FXO adapter. I've had the unit a few days and
am exploring it's workings. I really want it mostly as a
straightforward FXO adapter, to replace an X101p. Let me be clear, I'd
love to support Digium in every way possibe, and will likely buy a
TDM40 card shortly. But, the X101p has never worked well for me.

In any case, with respect to incomming calls on the SPA-3000 I do see
one odd thing happening that perhaps someone here can provide relevent
guidance. When a call arrives at the PSTN port the SPA-3K takes three
rings to pick it up. In this way it extracts caller id et al from the
line. Once it picks up the line its internal dialplan is set to
immediatel call forward (what Sipura calls Hotline syntax) to an
extension on my * server (in fact a SNOM 200.) 

Everything generally works, but once the SPA-3K answers the line the
caller on the other end no longer gets ringing tone while * rings the
SNOM phone. They get silence. Is there something in * that requires
setup so that they here ring tone until the extension is picked up?

The basic setup of the SPA-3K as a SIP device is very like that for my
existing SPA-2000. I simple changed the line in my local outbound
dialing macro from zap/g1 to SIP/3015 and voila, outbound dialing
through the SPA. FWIW, Sipura looks to be issuing firmware quickly at
the moment, and they are participating in a forum hosted by Voxilla for
the beta users.

Michael


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

o713-861-4005
o800-905-6412
c713-201-1262

"Behind every great man is a great woman rolling her eyes."
 
** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704


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Re: [Asterisk-Users] Serious NAT problems: can't call between lines on sipura

[Brian Cuthie]

Please ignore my previous post (below), as it's not really relevant to 
your problem. I was in some kind of mindless auto-email processing mode 
and responded without fully reading your message. Too much spam, too 
little sleep. Geesh.

-brian
Brian Cuthie wrote:
Bruce,
I think this is related to your firewall. You may want to take a look 
a posting I did a few weeks ago.

http://lists.digium.com/pipermail/asterisk-users/2004-May/046511.html
Something on this topic probably belongs in the wiki.
-brian
Bruce Komito wrote:
I have a problem that is almost certainly nat-related, but I can't 
figure
out what's happening.

Since moving the Sipura behind a NAT server (Linksys), I am no longer 
able
to call between the two lines on the same Sipura.  When I dial one
extension from the other, it rings, but immediately after I pick up the
ringing phone, the call is uncerimoniously dumped.  I can tell the call
terminates immediately because I am watching the CDRs come out.  The *
server is on a public address with no firewall between it and the 
outside
world.

sip.conf: (both extensions have identical settings)
; Bruce
[5815]
type=friend
username=5815
secret=wpti5815
host=dynamic
[EMAIL PROTECTED]
context=vpbx-wpti
qualify=3000
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
nat=yes
I'm thinking this has something to do with a setting in the Sipura, 
but I
don't know where to start.  I have nat keep-alive turned on, but I 
had to
turn stun off because it was causing a long, inexplicable delay after
dialing before the call would complete.

I'm realizing NAT with VoIP is a real problem.  Anyone have a silver
bullet they wish to share?
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115

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[Asterisk-Users] setting the number of rings befor asterisk picks up?

[hank]

hello how do I set the number of rings picks up on?
I am using a single port fxo card and currently asterisk is answering after
1 or 2 rings and I want it answering after 4 5 or 6 rings
thanks
hank
- -
Don't judge me because I'm blind. Judge me by what's inside. if you judge me
because I am blind, then it is you who is blind.
"time is the fire in which we burn," Tollian Soran.
"grudges aren't worth holding--One who holds them shows his self-weakness."
Contact info:
[EMAIL PROTECTED]
Email: Same as MSN.

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[Asterisk-Users] asterisk prompts?

[hank]

hello where can I get the asterisk prompts that are included in the sample
config at?
thanks
hank
- -
Don't judge me because I'm blind. Judge me by what's inside. if you judge me
because I am blind, then it is you who is blind.
"time is the fire in which we burn," Tollian Soran.
"grudges aren't worth holding--One who holds them shows his self-weakness."
Contact info:
[EMAIL PROTECTED]
Email: Same as MSN.

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[Asterisk-Users] PRI problem???

[Timothy R. McKee]

I have just finished installing a new asterisk box at my work.  The box is
quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory.  I have a 4
port Digium T1 card for channel bank and PRI access.

I activated a PRI from a local CLEC (DMS-500 based, National protocol).
This PRI is on slot 2 of the card and is set as the primary timing source.
It is ESF/B8ZS.  

All the software is latest CVS HEAD as of 5/22/04.

My problem lies in random intermittent drops of calls.  The entire PRI seems
to disappear, dropping all current established calls.  I see occasional
printouts on an asterisk management console showing all 23 B channels
resetting with no reason given:

pbx1*CLI>
-- B-channel 1 successfully restarted on span 2
-- B-channel 2 successfully restarted on span 2
-- B-channel 3 successfully restarted on span 2
-- B-channel 4 successfully restarted on span 2
-- B-channel 5 successfully restarted on span 2
-- B-channel 6 successfully restarted on span 2
-- B-channel 7 successfully restarted on span 2
-- B-channel 8 successfully restarted on span 2
-- B-channel 9 successfully restarted on span 2
-- B-channel 10 successfully restarted on span 2
-- B-channel 11 successfully restarted on span 2
-- B-channel 12 successfully restarted on span 2
-- B-channel 13 successfully restarted on span 2
-- B-channel 14 successfully restarted on span 2
-- B-channel 15 successfully restarted on span 2
-- B-channel 16 successfully restarted on span 2
-- B-channel 17 successfully restarted on span 2
-- B-channel 18 successfully restarted on span 2
-- B-channel 19 successfully restarted on span 2
-- B-channel 20 successfully restarted on span 2
-- B-channel 21 successfully restarted on span 2
-- B-channel 22 successfully restarted on span 2
-- B-channel 23 successfully restarted on span 2
pbx1*CLI>

(This is from an asterisk console started with -r.)

Has anyone encountered similar behavior in the past?  If so, what was the
resolution?

Thanks,

Tim McKee

configs:

zaptel.conf
#
# span 1 is for channel bank (24 FXS)
span=1,2,2,esf,b8zs
# span 2 is for NuVox PRI (1-23B, 24D)
span=2,1,2,esf,b8zs
# span 3 is unused
span=3,0,0,esf,b8zs
# span 4 is unused
span=4,0,0,esf,b8zs
#
#


zapata.conf

[channels]
;
; Default language
;
language=en
;
; Default context
;
context=dialnational
;
signalling=fxo_ks
use_callerid=yes
callwaiting=no
restrictcid=no
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=yes
relaxdtmf=no
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
amaflags=billing



;
; NuVox PRI
context=sdnpri
;
switchtype = national
pridialplan = unknown
signalling = pri_cpe
callerid=asreceived
group = 2
channel => 25-47


<>

Re: [Asterisk-Users] Asterisk Prepaid

[Mike Machado]

Have you tried the calling card sample AGI listed in the Wiki? I am not
using it in production, but I tested it and it seemed straight forward. 

http://www.voip-info.org/wiki-Asterisk+tips+and+tricks

On Sun, 2004-05-23 at 16:10, usedcanon wrote:
> I have a requirement for a setup with prepaid call credits.
> 
> I am aware of the two applications available (been researching for the past
> week), app_prepaid and app_rateengine. However neither of the two sound like
> exactly what I want. However I was wondering that someone who has used it
> might be able to say if they could be used in my scenario.
> 
> Basically my scenario is pretty straight forward. Credit will be allocated
> to the ddi, I dont need any announcements etc (maybe low credit warning
> during call could be useful thoug). From the users prespective everything
> will be transparent. However the call should disconnect when the credit runs
> out. The CDR and the account DB need to be adjusted according to the call
> made.
> 
> My guess is that app_prepaid could used with modification, I am assuming
> here that this is not possible as-is with configuration.
> 
> Basically in case of the prepaid app, the card number can be replace
> transparently with the callerID.
> 
> All help, guidence and comments will be extremelly appreciated.
> 
> Umar.
> 
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Re: [Asterisk-Users] PRI problem???

[Steven Critchfield]

On Sun, 2004-05-23 at 21:52, Timothy R. McKee wrote:

> My problem lies in random intermittent drops of calls.  The entire PRI seems
> to disappear, dropping all current established calls.  I see occasional
> printouts on an asterisk management console showing all 23 B channels
> resetting with no reason given:
> 
> pbx1*CLI>
> -- B-channel 1 successfully restarted on span 2

> (This is from an asterisk console started with -r.)
> 
> Has anyone encountered similar behavior in the past?  If so, what was the
> resolution?
> 

Is there a problem with your google access?
http://tinyurl.com/2fjkb

This was answered as late as last month and covered about every 4 months
for people to lame to do their own simple google search.

-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] Aastra ADSI phone

[Michael Welter]

I've received my Aastra 390 phone.  I got the unlock procedure from the 
vendor, and the services button now shows all four entrys as "".

When I give the phone "ADSIProg()" the phone displays:
Asterisk PBX
download refused
Conflict with:

The asterisk.adsi hasn't been changed:
DESCRIPTION "Asterisk PBX"
VERSION 0x02
SECURITY 0x
FDN 0x000f
...
When I change the FDN number to anything else, the phone replies 
"Services full".

Is the phone properly unlocked?  Has anyone seen this before?
Thanks,
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
[EMAIL PROTECTED]
www.introspect.com
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Re: [Asterisk-Users] asterisk prompts?

[Mike Heininger]

Am 24.05.2004 um 04:36 schrieb hank:
hello where can I get the asterisk prompts that are included in the 
sample
config at?
they are located in the sounds folder after checkout of the cvs and in 
/var/lib/asterisk/sounds/ after installing *.

Mike
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Re: [Asterisk-Users] Asterisk Prepaid

[Stephen Davies]

On Mon, 24 May 2004, usedcanon wrote:

> I have a requirement for a setup with prepaid call credits.
> 
> I am aware of the two applications available (been researching for the past
> week), app_prepaid and app_rateengine. However neither of the two sound like
> exactly what I want. However I was wondering that someone who has used it
> might be able to say if they could be used in my scenario.
> 
> Basically my scenario is pretty straight forward. Credit will be allocated
> to the ddi, I dont need any announcements etc (maybe low credit warning
> during call could be useful thoug). From the users prespective everything
> will be transparent. However the call should disconnect when the credit runs
> out. The CDR and the account DB need to be adjusted according to the call
> made.
> 
> My guess is that app_prepaid could used with modification, I am assuming
> here that this is not possible as-is with configuration.
> 
> Basically in case of the prepaid app, the card number can be replace
> transparently with the callerID.

Hi,

I did this to app_prepaid - you can pass a parameter into Prepaid() -
its looked up in a table to find an "associated" card number - if that
is found then the card number prompt is skipped and the associated
card is used automatically.

I can send a patch if you like (will also include a minor change or
two to have app_prepaid work against CVS.

Steve


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[Asterisk-Users] STREAM FILE question

[Jer]

Dear all
I was wondering is there a way to advance/rewind in playback?(STREAM FILE) 
say 5 seconds
somehow i don't think so but I thought I' would ask

Thanks
Jer
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