Re: [Asterisk-Users] Call Quality Detail Record

[John Todd]

At 3:32 PM +0200 on 3/17/05, Calin Serbanescu wrote:
Hello,
I need some help setting up statistics per call. I need to store in a
database call quality details such as jitter, packets lost and other
informations. Is there any way to do this?
I'd really appreciate some links or any other kind of info on this.
Thanks,
Calin.
If  you're using Cisco 79xx devices with reasonably new SIP images, 
or VOIP Inc. devices, you might have some luck since both of those 
support the Tx and Rx header stats out of SIP BYE packets.

http://lists.digium.com/pipermail/asterisk-dev/2004-May/004174.html
Since that post, there are now built-in routines to extract SIP 
headers in the dialplan - see "show application SIPGetHeader".   You 
may be able to get that to work instead of making your own patches.

There is also now some stuff in the IAX2 statistics area (as of 
yesterday?) but I don't know if it's reference-able from the dialplan 
or anywhere else.  Maybe someone can fill me in on that?

You're on your own for making a database of the information, though.  My
JT
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Re: [Asterisk-Users] PRI Cause Code Help

[Peter Svensson]

On Thu, 17 Mar 2005, Eric Knudson wrote:

> hmm - that seems correct - I remember experiencing some similar
> problems(using Cisco IOS gateways, though) which were addressed by
> "voice call convert-discpi-to-prog"
> 
> http://tinyurl.com/5xn8n
> 
> That command addressed an IOS gateway's tendency to hang up before the
> in-band message could be played; similarly, it appears that * may not
> be providing the in-band message to playback to the calling party when
> an extension is out of service or something.

The two issues are only somewhat related. The RELEASE COMPLETE as an reply
to a SETUP after having sent a CALL PROCEEDING is probably not allowed by
the state transitions listen in q.931. 

The in-band announcement is more related to whether we have sent a
progress information element which states that in-band audio is available.  
I think Asterisk sends such a progress message almost as soon as possible.  
However, in this case the problem is a CALL PROCEEDING before the
RELEASE_COMPLETE answering teh SETUP. The fact that the CALL PROCEEDING
also includes a PROGRESS element is incidental.

Peter


> On Fri, 18 Mar 2005 00:06:24 +0100 (CET), Peter Svensson
> <[EMAIL PROTECTED]> wrote:
> > On Thu, 17 Mar 2005, Eric Knudson wrote:
> > 
> > > Trevor,
> > >
> > > Nah, I think the response is correct. Take a look at the chart again:
> > >
> > > http://www.lkn.ei.tum.de/lehre/kn2/anhangKap4.pdf
> > >
> > > look at the incoming setup procedure(1 of 2) (user side).
> > >
> > > If you get an incoming SETUP, then you MUST respond with one of the 
> > > following:
> > >
> > > CALL PROCEEDING, or ALERTING, or CONNECT, or RELEASE COMPLETE. Follow
> > > the rest of the logic path to completion for the rest of the call.
> > 
> > You are not allowed to first respond with a CALL PROCEEDING and then a
> > RELEASE COMPLETE. In that case DISCONNECT is allowed I think.
> > 
> > > I suspect that the carrier(since they mentioned a progress indicator)
> > > expects your equipment to cut through audio and play a message
> > > back("we're sorry, that extension is not available, blah, blah") or
> > > play a busy tone or something.
> > 
> > Peter
> > 
> >
> 

Peter
-- 
Peter Svensson  ! Pgp key available by finger, fingerprint:
<[EMAIL PROTECTED]>! 8A E9 20 98 C1 FF 43 E3  07 FD B9 0A 80 72 70 AF

Remember, Luke, your source will be with you... always...


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Re: [Asterisk-Users] Web triggered calls

[Michiel van Baak]

On 09:59, Tue 15 Mar 05, C F wrote:
> look into .call files, check the wiki for this:
> www.voip-info.org

Or look at the Manager interface.
The command you are looking for is: Originate
Good luck
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Re: [Asterisk-Users] About the weather..

[John Todd]

At 12:08 AM -0500 on 3/18/05, Kris Edwards wrote:
Ok, I've been away from the list for sometime now and feel as though I'm
going to ask something that's been asked many times before, yet I find
nothing in the wiki, so here goes:
I notice that allison has done recordings for weather forecasts, yet I
find no agi's that parse forecasts and use this recordings to piece
together the forecast.  I find plenty of weather agi's, but they all
seem to be festival related rather than using prerecorded bits and
pieces.  Anybody know of a script that makes use of these recordings?
Thanks for the help!
To my knowledge, they were recorded in anticipation of somebody 
getting ambitious and writing some AGIs around the recordings. 
Perhaps you can try  your hand at some Perl...

JT
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RE: [Asterisk-Users] Cisco 79XX Phones

[Nabeel Jafferali]

> Are there programmable softkeys in the Cisco SIP software?

Instead of lineX_name, etc., you can use, for example:

speed_label6: "Pick-up Group"
speed_line6: "*8"

Nabeel
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Re: [Asterisk-Users] Re: asterisk+radius

[Paul P. Pongco]

oops never mind, ill just read on astcc

On Fri, 2005-03-18 at 14:10, Paul P. Pongco wrote:
> Hello,
> 
> Im actually deciding if I will use asterisk+radius for AAA purposes or
> use logging directly to mysql and using  Asterisk+RealTime to store SIP
> users to mysql also. 
> Question is, what's the best way to disconnect a user, if for example,
> he runs out of credits. thanks.
> 
> On Fri, 2005-03-18 at 02:33, izo wrote:
> > set asterisk to log into database directly via there are mysql ,
> > postgresql and odbc drivers
> > available. 
> > You dont need radius at all,
> > for  billing and accounting all u need is a frontend to database
> > 
> > 
> > On Thu, 17 Mar 2005 12:29:34 -0500, Matt wrote:
> > > Oh this is sad.. I'm familiar with radius.. and was hoping to be able
> > > to use asterisk with freeradius to be able to do call accounting and
> > > billing.. so you're telling me this is now not a good idea?
> > > Am I better off (for now) parsing the csv report each month?
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Cheers,

Paul P. Pongco
Mosaic Communications Inc.



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[Asterisk-Users] Cisco 79XX Phones

[Dan Levine]

Title: Cisco 79XX Phones






Hello Everyone,


I'm trying to add a Park Button to Cisco 7960G phones connected to an Asterisk box.  Does anyone know if this is possible?  I'm concerned that our customer will not understand that they have to dial #700 to park a call.  Are there programmable softkeys in the Cisco SIP software?

Thanks so much

Dan


-

Dan Levine

CYTEXONE | Your Technology Specialists

t: 877.CYTEXONE x 810

l: 212.477.0990 x 810

e: [EMAIL PROTECTED]

http://www.cytexone.com



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Re: [Asterisk-Users] ANNOUNCEMENT: Updatesforapp_cbmysqlandMeetMe2gui(out of tree modules)

[Henry Devito]

It doesn't matter if I run it from the apps directory or the asterisk 
directory I get the same response.  This is getting frustrating.
- Original Message - 
From: "Dan Austin" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, March 18, 2005 12:17 AM
Subject: RE: [Asterisk-Users] ANNOUNCEMENT: 
Updatesforapp_cbmysqlandMeetMe2gui(out of tree modules)

I should have read a little closer-
[EMAIL PROTECTED] apps]#  patch -p1 <
/var/build_aah/asterisk_src/asterisk/apps/apps-meetme-cbmysql.txt
If you run patch from within the apps directory, you will need
to use -p2.  Or just cd .. and use the same command as above.
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[Asterisk-Users] Cisco 7940

[AS]

Can anyone share a SIPDefault.cnf file for a 7940 that is NAT'd

This is what registeres on asterisk

michael/michael  2xx.xxx.xxx.xx3D   N  255.255.255.255  50526
UNREACHABLE

Instead of 

roy/roy  2xx.xxx.xxx.xx4 D   N  255.255.255.255  5060 OK
(155 ms)

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Re: [Asterisk-Users] Why should I answer a Newbie question, therethick!

[Brian Capouch]

Vincent wrote:
Hi all,
You don't want to be fooled by - -. This guy has NO
business ethic. When He refused to realize a business deal in which I
agreed to pay for his coding help for me, will he personally pay for
the hosting of the list? More interestingly, he mentioned in the list
that he lives in  in Timbuktu, Ontario while he told me that he lives
in Asheville, North Carolina but home number is a Hendersonville, NC
phone number. I was just updated that according to the phone company
records that is not the name of the person the phone number is
associated with.
Vincent
I'm sorry to report that your bitter, petty, and unethical attempt to 
gain vengance against this list member is totally transparent.

It will have no effect on anyone's opinion of the target of your rant, 
but great effect on everyone's opinion of you.

You abused the mailing list, you took a personal gripe public in a 
vulgar way, and you proved your own cluelessness by not getting his joke 
about being from Timbuktu.

Next time you should count to, um, infinity before posting crap like 
this to the list.

B.
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RE: [Asterisk-Users] ANNOUNCEMENT: Updatesforapp_cbmysqlandMeetMe2gui (out of tree modules)

[Dan Austin]

I should have read a little closer-

> [EMAIL PROTECTED] apps]#  patch -p1 < 
> /var/build_aah/asterisk_src/asterisk/apps/apps-meetme-cbmysql.txt

If you run patch from within the apps directory, you will need
to use -p2.  Or just cd .. and use the same command as above.
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[Asterisk-Users] limit about asterisk pstn out

[FCG ZHAO Zigang]

I have a system include asterisk + ser.

when I want to limit a dial out to pstn , I will do that :

extensions.conf

exten => _9NXXNX/[EMAIL PROTECTED],Congestion
exten => _9NXXNX, 1,Dial(ZAP/g2/{EXTEN:1},30,t)
exten => _9NXXNX, 2,Hungup

but I don't confirm is it right.
I have no env to test it. 

who can help me?
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Re: [Asterisk-Users] Why should I answer a Newbie question, therethick!

[Vincent]

Hi all,

You don't want to be fooled by Race Vanderdecken. This guy has NO
business ethic. When He refused to realize a business deal in which I
agreed to pay for his coding help for me, will he personally pay for
the hosting of the list? More interestingly, he mentioned in the list
that he lives in  in Timbuktu, Ontario while he told me that he lives
in Asheville, North Carolina but home number is a Hendersonville, NC
phone number. I was just updated that according to the phone company
records that is not the name of the person the phone number is
associated with.

Regards,
Vincent


On Wed, 2 Mar 2005 11:57:12 -0500, Race Vanderdecken
<[EMAIL PROTECTED]> wrote:
> If some one would like to show me the site that explains how to setup a
> mailing list then I will create a Newbie list for asterisk and voip
> questions and answers.
> 
> I am only asking for someone to show me the site and maybe a few
> pointers on how to start it up. Only because I don't have the time or
> experience to do it quickely enough to get the newbies off the list. And
> I am a bit slow with apache and web type sutff, as you can tell by my
> website codetyrant.com.
> 
> I will personally pay for the hosting of the list.
> 
> It is not that I am tired or will ever grow tired of passing out fish
> and giving fishing lessons it is just I don't have the good fortune to
> be adept at web interfaces.
> 
> Also, suggestions for the domain name would be welcomed.
> 
> Race "The Tyrant" Vanderdecken
> 
> "In the Land of the Blind, the One-eyed man is Elvis...", copyright RPV
> 1997.
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Steven
> Critchfield
> Sent: Wednesday, March 02, 2005 11:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Why should I answer a Newbie
> question,therethick!
> 
> On Wed, 2005-03-02 at 11:01 -0500, Race Vanderdecken wrote:
> > This list is owned by no one.
> 
> Actually it is owned by Digium. It has many contributers though.
> 
> > When someone drives off the road into a ditch in a snow storm they
> last
> > thing they need is someone telling them they should have invested in
> > snow chains and defensive driving lessons before leaving the house.
> >
> > Newbies need help getting out of the ditch so traffic can continue to
> > flow and the rubber neckers can be abated. If you are not willing to
> > pull off to the side of the road and help the fool by pushing their
> car
> > out of the ditch you have no right to give him the finger as you drive
> > past.
> 
> As a person who spent 9 hours in traffic last winter just to drive 15
> miles due to idiots who should have just stayed home, I think your
> analogy breaks down.
> 
> At some point, you either need to learn to drive or you pay someone else
> to transport you or your stuff. Same applies to computer work, either
> you can do the work yourself or you pay someone else to do it. Even your
> snow driver analogy works here, you either get yourself out of the ditch
> or you pay someone to do it for you.
> 
> The payment is not always monetary. Sometimes the payment is just a
> showing of sufficient effort. Back to your snow driver analogy, if the
> driver in the ditch is just waiting in the car for you to come over and
> push them out without even attempting anything on their own, you would
> be less inclined to bother. You would be even less inclined to continue
> exerting your own effort if the driver was not cooperating or wasn't
> even interested in getting out to help push.
> --
> Steven Critchfield <[EMAIL PROTECTED]>
> 
> ___
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[Asterisk-Users] gsm cannot be found in any file form... but it's there

[Scheda]

Hey, I recorded this intro, and changed it to a gsm file in the shell,
and I'm getting an error saying that it isn't in the directory at all
when it's sitting right there. I don't know why that is.

If you want to hear it, it's http://scheda.underfireradio.com/astintro.mp3

I don't know what the matter is, I've tried renaming it, copy and
pasting it in there, deleting it and placing it back... I'm kinda out
of ideas.
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Re: [Asterisk-Users] Re: asterisk+radius

[Paul P. Pongco]

Hello,

Im actually deciding if I will use asterisk+radius for AAA purposes or
use logging directly to mysql and using  Asterisk+RealTime to store SIP
users to mysql also. 
Question is, what's the best way to disconnect a user, if for example,
he runs out of credits. thanks.

On Fri, 2005-03-18 at 02:33, izo wrote:
> set asterisk to log into database directly via there are mysql ,
> postgresql and odbc drivers
> available. 
> You dont need radius at all,
> for  billing and accounting all u need is a frontend to database
> 
> 
> On Thu, 17 Mar 2005 12:29:34 -0500, Matt wrote:
> > Oh this is sad.. I'm familiar with radius.. and was hoping to be able
> > to use asterisk with freeradius to be able to do call accounting and
> > billing.. so you're telling me this is now not a good idea?
> > Am I better off (for now) parsing the csv report each month?
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Cheers,

Paul P. Pongco
Mosaic Communications Inc.



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RE: [Asterisk-Users] ANNOUNCEMENT: Updatesforapp_cbmysqlandMeetMe2gui (out of tree modules)

[Dan Austin]

I tried this with a fresh tree, so I know it works.  Based
on your last message, I think you may have missed step 1.
My needs are for both CBMysql and MeetMe2, so I bundled
the patch file.  If you are only interested in one, then
delete the sections of the patch that are not needed.

1.  Download the source for MeetMe2.c from
http://www.areski.net/asterisk-meetme/about.php?s=0 
and put it in asterisk/apps

2.  Download the source for CBMysql from
 http://www.mithotech.com/asterisk/

3.  Extract the contents of the tar file and move/rename
the conference.c file to asterisk/apps/app_cbmysql.c

4.  Download the apps-meetme-cbmysql.txt and 
meetme2-gui.tar.gz from www.fitawi.com/Asterisk
While there, you may as well get the database table
description and the sample cbmysql.conf

5.  Copy cbmysql.conf to /etc/asterisk, modify to suit.

6.  Create/update your database to match the tables in
 the description file.
7.  cd to asterisk/apps and run 
patch -p1 < path-to/apps-meetme-cbmysql.txt

8.  make; make install

9.  Extract the meetme2-gui files to your webserver
document tree.

10.  Modify the ./lib/Defines.php to match your db_name,
db_user and db_password 

Dan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry
Devito
Sent: Thursday, March 17, 2005 7:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ANNOUNCEMENT:
Updatesforapp_cbmysqlandMeetMe2gui (out of tree modules)

Still a no go.  I can't figure out what I am doing wrong.  This is the
error 
I get

[EMAIL PROTECTED] apps]#  patch -p1 < 
/var/build_aah/asterisk_src/asterisk/apps/apps-meetme-cbmysql.txt
can't find file to patch at input line 4
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--
|diff -u asterisk-1.0.5-orig/apps/app_cbmysql.c 
asterisk-1.0.5/apps/app_cbmysql.c
|--- asterisk-1.0.5-orig/apps/app_cbmysql.c 2005-02-28 
23:19:19.0 -0800
|+++ asterisk-1.0.5/apps/app_cbmysql.c  2005-03-03 18:39:34.0
-0800
--

It asks for the file to patch if i put in /path/app_cbmysql.c

I get this error

File to patch: /var/build_aah/asterisk_src/asterisk/apps/app_cbmysql.c
patching file /var/build_aah/asterisk_src/asterisk/apps/app_cbmysql.c
can't find file to patch at input line 331
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--
|diff -u asterisk-1.0.5-orig/apps/app_meetme2.c 
asterisk-1.0.5/apps/app_meetme2.c
|--- asterisk-1.0.5-orig/apps/app_meetme2.c 2004-04-07 
04:37:18.0 -0700
|+++ asterisk-1.0.5/apps/app_meetme2.c  2005-03-03 10:31:36.0
-0800




- Original Message - 
From: "Dan Austin" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Wednesday, March 16, 2005 6:56 PM
Subject: RE: [Asterisk-Users] ANNOUNCEMENT: Updates 
forapp_cbmysqlandMeetMe2gui (out of tree modules)


Bugger.  I knew I'd screw up the patch instructions.
Try this-
#> cd to /var/build_aah/asterisk_src/asterisk
#> patch -p1 < path-to/apps-meetme-cbmysql.txt

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry
Devito
Sent: Wednesday, March 16, 2005 7:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ANNOUNCEMENT: Updates for
app_cbmysqlandMeetMe2gui (out of tree modules)

Dan,  Thanks for the time helping me out.  I figured everything out
except
for the patch.
>7.  cd to asterisk/apps and run patch -p0 <
>path-to/apps-meetme-cbmysql.txt

When I do this step it errors out and asks for the file to patch..  When
I
look at the apps-meetme-cbmysql.txt It shows the file name to be
app_cbmysql.c so I changed the name of the file cbmysql.c to
app_cbmysql.c
but it still doesn't work.

Also in the apps-meetme-cbmysql.txt it shows the path to be
asterisk-1.0.5-orig I am trying to install this on [EMAIL PROTECTED]  So
the
source is in /var/build_aah/asterisk_src/asterisk.  Maybe I'm looking in
the
wrong direction. any help would be appreciated.  I can even give you
root
access to my box.  When all is said and done I will write up a wiki page
for
installation.

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[Asterisk-Users] leaky reload

[Thomas Andrews]

If I comment out the following line in zapata.conf I would expect
asterisk to "forget" the cli information for that channel when I reload:

callerid="Uniden Dead" <(256) 428-6125>

... but it doesn't; I have to restart asterisk for it to take effect.
The funny thing is that the reverse is *not* true - ie if I uncomment
the line and "reload" then it learns about the caller id "Uniden Dead".

Why is this a one-way process ?

Thanks,
Thomas
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RE: [Asterisk-Users] Getting caller-name - cid_rewrite 1.0.0

[Steven Frazier]

I just dropped in the update, nice work, Jay!  I love the address feature
and the additional Google lookup, nice touch!

Steve


-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Friday, March 18, 2005 12:02 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Getting caller-name - cid_rewrite 1.0.0


Hi folks, I think my little agi script is ready for the big one-oh-oh.

Available at http://muware.com/asterisk is cid_rewrite-1.0.0.  
This agi-script does the following:

- Standardize incoming caller-id numbers to adhere to US 
dialing code; NANPA numbers are reformatted to 1+10, 
international numbers become 011 (this is 
customizable with a little PHP knowledge).
- Look up the associated caller-id name in a mysql table.
- If not found in the DB, it attempts a reverse-lookup on 
411.com (and extracts the City name even if the number was not found).
- If not found on 411.com, it attemps a reverse-lookup on 
google.com; if not found, it falls back to pulling the 
rate-center from telcodata.us.
- If available, the address associated with the phone-number is 
also extracted and inserted into the DB.

This has been working for a little over a month for me.  Have 
fun getting it to work on your end.

Disclaimer -- pulling information from 411.com doesn't agree 
with their terms of service.  Use at your own risk, and don't 
use it commercially. Lookups on google.com don't seem to have 
that restriction.

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Re: [Asterisk-Users] Three way calling with X-Lite / MeetMe

[Ronald Wiplinger]

Vyom A wrote:
In X-Lite, the "Conference" button is disabled, but that can probably
be done in X-Pro (from the  XPRO_users_guide.pdf)

There is a crack available:
http://www.regnow.com/softsell/nph-softsell.cgi?item=9054-12
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Re: [Asterisk-Users] Channel name (and substring)

[Vyom A]

I too have a doubt regarding the variables:
currently, to forward a call to a SIP proxy, I use
 
exten => _0.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
 
where 'sipproxy' is defined as a peer in sip.conf. Using this, I can onlyuse the 'name' part of the URI as anything after @ is removed.
But how can I retain the 'domain' part and still forward the calls to theproxy?. 
i.e. I have to send the whole string ${EXTEN:[EMAIL PROTECTED] tothe peer sipproxy. Can this be done?
		Do you Yahoo!? 
Make Yahoo! your home page 
 
 
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Re: [Asterisk-Users] Three way calling with X-Lite / MeetMe

[Vyom A]

In X-Lite, the "Conference" button is disabled, but that can probablybe done in X-Pro (from the  XPRO_users_guide.pdf)
		Do you Yahoo!? 
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Re: [Asterisk-Users] Newbie can't dial out to pstn

[Greg]

I have just run ztcfg and got these errors:
# ztcfg
Notice: Configuration file is /etc/zaptel.conf
line 209: Cannot get number of tones chanel 1
line 209: Cannot init tones chanel 1
line 209: Cannot get number of tones chanel 2
line 209: Cannot init tones chanel 2
line 209: Cannot get number of tones chanel 3
line 209: Cannot init tones chanel 3
line 209: Cannot get number of tones chanel 4
line 209: Cannot init tones chanel 4
What would these mean. I searched the archives and couldn't find these 
errors.

Greg
On 18/03/2005, at 1:24 PM, Greg wrote:
I was just copy an example from somewhere. I made the change but the 
mobile still doesn't ring. The line the card is attached to works 
fine. here is the new output

Executing Goto("SIP/2002-4385", "mobile|0400039953|1") in new stack
-- Goto (mobile,0400039953,1)
-- Executing Goto("SIP/2002-4385", "localcall|0400039953|1") in 
new stack
-- Goto (localcall,0400039953,1)
-- Executing Dial("SIP/2002-4385", "ZAP/1/0400039953|60|r") in new 
stack
-- Called 1/0400039953
-- Zap/1-1 answered SIP/2002-4385
-- Hungup 'Zap/1-1'
  == Spawn extension (localcall, 0400039953, 1) exited non-zero on 
'SIP/2002-4385'

is this line -- Zap/1-1 answered SIP/2002-4385 displayed when the card 
tries to make the call or when the card thinks it has established the 
call?

Regards,
Greg
By the way, I'm on the Gold Coast.
On 18/03/2005, at 12:32 PM, Shane Dalgleish wrote:
Greg,
Any reason why you are putting the country code on the front for a 
mobile
call through pstn?
(Unless you have something like an Ericsson F220M Fixed Cellular 
Terminal
connected to it?)

And you said the tdm400p never tries to pick up the phone?
Have you connected a normal phone on the line and had a listen?
Where is Aus are you? :o)
Cheers
Shane
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Sent: Friday, 18 March 2005 1:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Newbie can't dial out to pstn
Hi,
I have just put in a tdm400p with 4 fxo modules and am trying
to dial out from x-lite to dial my mobile phone just to test.
The output in the asterisk console is like this
Executing Goto("SIP/2002-239b", "mobile|61400039953|1") in new stack
 -- Goto (mobile,61400039953,1)
 -- Executing Goto("SIP/2002-239b",
"localcall|61400039953|1") in new stack
 -- Goto (localcall,61400039953,1)
 -- Executing Dial("SIP/2002-239b",
"ZAP/1/61400039953|60|r") in new stack
 -- Called 1/61400039953
 -- Zap/1-1 answered SIP/2002-239b
 -- Hungup 'Zap/1-1'
   == Spawn extension (localcall, 61400039953, 1) exited
non-zero on 'SIP/2002-239b'
It never tries to pick up the phone and dial out. I'm not
sure if the config is correct, but I can easily dial between
x-lite clients, just not get the pstn.
Can anyone see any glaring mistakes?
Any help is grealty appreciated.
Regards,
Greg
My extensions.conf part is this:
exten => _04,1,GoTo(mobile,61${EXTEN:1},1)
[localcall] ; local calls by PSTN ?is a fixed charge, voip is
per minute exten => _X.,1,Dial(ZAP/1/${EXTEN},60,r) exten =>
_X.,2,Congestion exten => _X.,3,Hangup exten =>
_X.,103,Hangup exten => _X.,104,Hangup exten => _X.,105,Hangup
[mobile] ; Maybe be cheaper to route mobile calls differently
to STD in some cases exten => _X.,1,Goto(localcall,${EXTEN},1)
zaptel.conf
fxsks=1-4
loadzone=au
defaultzone=au
channels=1-4
zapata.conf
[channels]
 
busydetect=1
busycount=7
 
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
 
usecallerid=yes
 
echocancel=yes
echocancelwhenbridged=yes
 
rxgain=0.0
txgain=0.0
 
group=1
pickupgroup=1-4
 
immediate=no
 
context=incomingcall
 
signalling=fxs_ks
callerid=asreceived
channel=1-4
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[Asterisk-Users] About the weather..

[Kris Edwards]

Ok, I've been away from the list for sometime now and feel as though I'm
going to ask something that's been asked many times before, yet I find
nothing in the wiki, so here goes:

I notice that allison has done recordings for weather forecasts, yet I
find no agi's that parse forecasts and use this recordings to piece
together the forecast.  I find plenty of weather agi's, but they all
seem to be festival related rather than using prerecorded bits and
pieces.  Anybody know of a script that makes use of these recordings?

Thanks for the help!
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Re: [Asterisk-Users] Getting caller-name - cid_rewrite 1.0.0

[Duane]

On Fri, March 18, 2005 16:01, Jay Milk said:
> Hi folks, I think my little agi script is ready for the big one-oh-oh.

Feel free to check the e164.org zone for TXT records...

-- 
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

"In the long run the pessimist may be proved right,
but the optimist has a better time on the trip."

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[Asterisk-Users] Getting caller-name - cid_rewrite 1.0.0

[Jay Milk]

Hi folks, I think my little agi script is ready for the big one-oh-oh.

Available at http://muware.com/asterisk is cid_rewrite-1.0.0.  This
agi-script does the following:

- Standardize incoming caller-id numbers to adhere to US dialing code;
NANPA numbers are reformatted to 1+10, international numbers become
011 (this is customizable with a little PHP
knowledge).
- Look up the associated caller-id name in a mysql table.
- If not found in the DB, it attempts a reverse-lookup on 411.com (and
extracts the City name even if the number was not found).
- If not found on 411.com, it attemps a reverse-lookup on google.com; if
not found, it falls back to pulling the rate-center from telcodata.us.
- If available, the address associated with the phone-number is also
extracted and inserted into the DB.

This has been working for a little over a month for me.  Have fun
getting it to work on your end.

Disclaimer -- pulling information from 411.com doesn't agree with their
terms of service.  Use at your own risk, and don't use it commercially.
Lookups on google.com don't seem to have that restriction.

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[Asterisk-Users] Realtime - how to reload ?

[Ronald Wiplinger]

I had the impression  that the command:
*CLI> realtime load sippeers name 621
(The new configuration was displayed after that command)
would re-load the config of phone 621
(I changed the context and tried above command, however, it kept the old 
info!)

bye
Ronald
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Re: [Asterisk-Users] Sip billing {expanded} for Pre-Paid BillingSystem needed.

[Henry Devito]

That phone number is a Hendersonville, NC phone number.
According to the phone company records that is not the name of the person 
the phone number is associated with.

- Original Message - 
From: "Vincent" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Thursday, March 17, 2005 10:02 PM
Subject: Re: [Asterisk-Users] Sip billing {expanded} for Pre-Paid 
BillingSystem needed.


Hi all,
This guy (Race Vanderdecken) told me that he is Asheville, North
Carolina, EST timezone. However, below he said he is in Timbuktu,
Ontario. If you are going to make any business deal with him, please
make sure you think twice and be very careful.
A month ago I disclosed my idea/design/system/requirements to him and
made a business deal with him such that he promised to integrate
Asterisk with IBS billing for me within 40 hours.  He took the
idea/design and advertised it in the mail-list in his unique own way
to attract other people's attention for more business but has been
reluctant to realize his promise to me. Deadline has been revised and
revised with different execuses. I just hung up with him at his home
number 1 828 891 5102 and was told that many people/customers showed
up with more money for the code, too much work and the job was
underpriced.
Regards,
Vincent
On Tue, 22 Feb 2005 11:06:48 -0500, Race Vanderdecken
<[EMAIL PROTECTED]> wrote:
My two proposals are:
1. You can contact webvoip.com they are billing guys who do what you
need.
3. You can wait for IBS to become integrated into Asterisk.
Your question is a little vague as to what you need.
The more chatty among us readers will surprise you with help if you can
give more details, such as.
I am in Timbuktu, Ontario, and I have 300 customers who I service with
SIP phones and I need to install/create/buy ad pre-paid SIP billing
system.
Or I need a pre-paid billing system that will allow me to handle 10,000
customers using SIP. I have installed Asterisk and I am using IAX2 to
PSTN.
There are several good systems out there that I do not have knowledge of
at this time.
You should also post your questions, revised and detailed, to the
Asterisk-Biz mail list. The sales guys from reputable firms are waiting
for you now.
Good luck Fredrik,
Race "They Tyrant" Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Insider KT
Sent: Tuesday, February 22, 2005 9:50 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sip billing
Hi. I am trying to setup a pre-paid sip billing, but I can only find
calling
card application.
Is there somewhere I can find one ?
Fredrik
_
MSN Messenger http://www.msn.no/computing/messenger Den raskeste veien
mellom deg og dine venner
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Re: [Asterisk-Users] Re: Snom190 intercom

[Shaun Dwyer]

Hi Josh, List,
I've managed to get the intercom working with the patch as available from:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom
(labelled as: For those who want a patch that doesn't affect VXML_URL, 
get it here .),
then as per the instructions from the wiki, SetVar(_INTERCOM=true) 
before dial.

This has the effect of doing what Josh said needs to be done in that 
intercom=true
is put on the end of the request URI instead of appended to the To: 
header as per what
SetVar(_VXML_URL=intercom=true) would do.

So now this is confirmed as working properly with the Snom190.
You must also enable the intercom either thru a subscribed config that 
tells the phone:
intercom_enabled&: on

or by setting it in the advanced options.
I am running firmware snom190-SIP 3.57v.
If anyone wants more detail on how I went about getting this working, 
please do email me :)

Cheers,
-Shaun
Shaun Dwyer wrote:
Hi Josh,
Thanks for the info..
how did you get intercom=true into the URI, and onto the end of the 
INVITE line?

btw, I got an intresting response from Sven of Snom...
[Sven Fischer (support) wrote:]
Hi,
as far as I understood intercom will only work if you are not using 
any password for registering at the registrar at the moment.

But we will add a line based auto answer functionality which should 
enable intercom for our phones more easily.

regards,
Sven Fischer


Cheers,
-Shaun
Josh Dady wrote:
As you can see from the SIP trace below (from the called phone), 
intercom=true is being appended to the To: header as per requirements.

The "intercom=true" needs to be appended to the request URI, not to 
the header as a whole -- your To: header should be:

To: 
Mind you, I didn't get the phone to respond to the intercom=true 
until I added it on the request line as well, so the INVITE line of 
your request would be:

INVITE sip:1011 at 
192.168.10.150:2051;line=9avrmhew;intercom=true SIP/2.0

I'm on a Snom 220 with SIP 3.56t, and I'm stuck on the very next step 
of the process -- answering the phone's challenge to the INVITE 
request.  The wiki indicates that the Snom needs to challenge with 
realm=snom, but even if I add snom into our internal DNS so that I 
can set the registrar to snom (that being the only way I can see to 
change what the phone uses as realm), it still rejects the digest 
response.  Anyone have this working with recent loads of SIP that can 
shed any light on this?

I've email'd snom a few days ago but have yet to get a response.

According to their web page, they have a new office as of April 1, 
and I got a response to a support request (on this very issue) today 
saying that they'd likely not be able to respond until people are 
settled into the new offices, so you'll likely have to be patient 
with them.

--
Joshua P. Dady

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RE: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server]

[Peter Childs]

We switched out our Cisco AS5300 with a new TE410P card, works a treat.

We couldn't get the D channels up when we hooked up to a PBX ISDN-PRI when
the zap was pri_net, but hooking up to a carrier ISDN-PRI as pri_cpe works a
treat.

Now we just have to figure out how to get the echocancelling tail a little
longer without stuffing the line full of static (?!!)  [=yes works =256 very
noisy...]From testing with our Cisco we needed more than 16ms tail, 32ms
tail on the cisco cleaned up incoming that originated from analog PSTN
fine...

 Cheers,
   Peter

-Original Message-
From: Eric Bishop [mailto:[EMAIL PROTECTED] 
Sent: Saturday, 12 March 2005 11:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Peter Childs
Subject: Re: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq
DL380 G4 server]

How did you go?


On Tue, 8 Mar 2005 11:28:59 +1030, Peter Childs <[EMAIL PROTECTED]>
wrote:
> 
> Digium shipped me a replacement card, but they sent the wrong one, so 
> they fedex'd another and its just arrived.
> 
> Should be testing in the next two days (the box is in another 
> state...)
> 
> The last I heard from Eric Bishop (on the 1st march) was that he had 
> received an updated card from digium, but it didn't function in his 
> DL380...
> 
> I can let you know the outcome of the test if you'd like.

[...]

> -Original Message-
> From: Mark F. Vickers [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, 8 March 2005 11:20 AM
> To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
> Subject: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq 
> DL380 G4 server]
> 
> Was there any resolution on this I also have a TE410P in an box with 
> an Intel E7501 chipset?
> 
> -Vickers

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[Asterisk-Users] seeking GSM 850/1900 gateway

[Stewart Nelson]

Hi,

I'm looking for a reliable, reasonably-priced, single-channel
interface between * and US GSM.

The VOIP GSM Gateways listed at
http://www.voip-info.org/wiki-VOIP+GSM+Gateways
(VoiceBlue, QUTEX) are multichannel systems, very expensive
($2500 or more).

Next step down, there are various Fixed Cellular Terminal
(FCT) or Fixed Wireless Terminal (FWT) devices.  These
typically have an FXS interface.
http://www.voip-info.org/wiki-Asterisk+Connecting+to+the+Cellular+Network
Unfortunately, the mainstream devices (Nokia, Ericsson, Siemens) seem to
omit the US 850 MHz band. The few that have this band, e.g. Telular, are
quite expensive (> $500) and I can't seem to find any good reviews.

At the bottom are "docking stations" for cellular handsets, e.g. CellSocket.
They are cheap enough (~ $100 + an old phone), and there is lots of
commentary about them, but alas, it's mostly negative.

Anyone have good luck connecting to US GSM?

Thanks,

Stewart

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Re: [Asterisk-Users] Sip billing {expanded} for Pre-Paid Billing System needed.

[Vincent]

Hi all,

This guy (Race Vanderdecken) told me that he is Asheville, North
Carolina, EST timezone. However, below he said he is in Timbuktu,
Ontario. If you are going to make any business deal with him, please
make sure you think twice and be very careful.

A month ago I disclosed my idea/design/system/requirements to him and
made a business deal with him such that he promised to integrate
Asterisk with IBS billing for me within 40 hours.  He took the
idea/design and advertised it in the mail-list in his unique own way
to attract other people's attention for more business but has been
reluctant to realize his promise to me. Deadline has been revised and
revised with different execuses. I just hung up with him at his home
number 1 828 891 5102 and was told that many people/customers showed
up with more money for the code, too much work and the job was
underpriced.

Regards,
Vincent

On Tue, 22 Feb 2005 11:06:48 -0500, Race Vanderdecken
<[EMAIL PROTECTED]> wrote:
> My two proposals are:
> 
> 1. You can contact webvoip.com they are billing guys who do what you
> need.
> 
> 3. You can wait for IBS to become integrated into Asterisk.
> 
> Your question is a little vague as to what you need.
> The more chatty among us readers will surprise you with help if you can
> give more details, such as.
> 
> I am in Timbuktu, Ontario, and I have 300 customers who I service with
> SIP phones and I need to install/create/buy ad pre-paid SIP billing
> system.
> 
> Or I need a pre-paid billing system that will allow me to handle 10,000
> customers using SIP. I have installed Asterisk and I am using IAX2 to
> PSTN.
> 
> There are several good systems out there that I do not have knowledge of
> at this time.
> 
> You should also post your questions, revised and detailed, to the
> Asterisk-Biz mail list. The sales guys from reputable firms are waiting
> for you now.
> 
> Good luck Fredrik,
> 
> Race "They Tyrant" Vanderdecken
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Insider KT
> Sent: Tuesday, February 22, 2005 9:50 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Sip billing
> 
> Hi. I am trying to setup a pre-paid sip billing, but I can only find
> calling
> card application.
> Is there somewhere I can find one ?
> 
> Fredrik
> 
> _
> MSN Messenger http://www.msn.no/computing/messenger Den raskeste veien
> mellom deg og dine venner
> 
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[Asterisk-Users] Configuring Asterisk with BroadVoice

[Kerry Garrison]

As promised, we have completed the instructions for getting BroadVoice to
work.The instructions are geared towards [EMAIL PROTECTED] / AMP Interface but
instructions are given for non-AMP users. The complete writeup can be found
at:
http://geekgazette.com/index.php?option=com_content&task=view&id=20&Itemid=2
6
 
Kerry Garrison
http://www.geekgazette.com


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[Asterisk-Users] Caller Presentation on E1 PRI

[Kong]

I am using a E1 PRI in malaysia with 30 channels.
every time i make a call, it passes my calleridnum through as the calling 
presentation.
i tried to use sercallerpres and also callingpres to override so that it 
will use the default calling presentation number, but there is no luck, is 
there any bug on this 2 functions? or did i use it wrong. any working 
example is appreciated.

thank you.
---
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Re: [Asterisk-Users] Use of asterisk to make use of IP phone speakerphones as a baby monitor....

[Kristian Kielhofner]

Francois Menard (Mailing List Account) wrote:
Is this possible?
f.
You bet!!!
Use a Polycom IP500 and use the Auto Answer config from my website:
www.kriscompanies.com
--
Kristian Kielhofner
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RE: [Asterisk-Users] Use of asterisk to make use of IP phone speakerphones as a baby monitor....

[Nick Stein]

Yes

Nick Stein

 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francois
Menard (Mailing List Account)
Sent: Thursday, March 17, 2005 7:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Use of asterisk to make use of IP phone
speakerphones as a baby monitor

Is this possible?

f.

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[Asterisk-Users] Extension ringing but no ringing sound asterisk

[Paul Dracevich]

When I call from extension A on Box
and to Extension A on Box B I get no ringing sound.

 

Regards

Paul Dracevich

Wireless Technology Consultant

Wayby Group

 

Mobile +64 29 638 9675

Phone +64 9 623 2143

Fax +64 9 623 1380

email [EMAIL PROTECTED]

website www.vnet.cc

 

"the freedom to communicate is the right of every
individual in the 21st century" "Intellectual Property protection is
the key to the Knowledge Economy" This email was sent to you via YOUtopia
... it's all about YOU.

 

The information contained in this email and any attachments
is confidential and may be legally privileged. If you are not the intended
recipient then you must not use, disseminate, distribute or copy any
information contained in this email or any attachments. If you have received
this email in error, please contact us immediately and delete this email. Thank
you.

 






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Re: [Asterisk-Users] Codec negotiation

[Rod Bacon]

Further to this, does anyone know if there is a simple way to set the party 
priority in codec negotiation? (NOT the codec priority)

In other words, I want the calling (client) preferences to be considered 
FIRST.

Currently, my logs show
Accepting AUTHENTICATED call from 203.89.xxx.xx:
  > requested format = ulaw,
  > requested prefs = (),
  > actual format = ilbc,
  > host prefs = (ilbc|g729|gsm|speex|g726|alaw|ulaw),
  > priority = mine
As can be seen, the request was for ulaw, which IS in the list, but the 
request is ignored because "priority = mine".

Can "priority = caller"? Or is this caused by the fact that "requested 
prefs" is empty?


- Original Message - 
From: "Mark Eissler" <[EMAIL PROTECTED]>
Sent: Wednesday, January 26, 2005 8:51 AM
Subject: [Asterisk-Users] Codec negotiation


Can't you just create a different context for inbound and outbound
calls? Then specify your codec preference order in there. I don't think
you can specify the bandwidth= parameter within a context other than
the global one though.
-mark
On Jan 25, 2005, at 6:13 PM, <[EMAIL PROTECTED]> wrote:
I don't want that... because
- for outbound calls I want priority to be g729 first
- for inbound calls I want no priority at all (e.g. the calling
asterisk
to decide which codec we will use)
The last doesn't happen..
This by the way DID happen correctly with previous versions of asterisk
(1.0.3 for example) the current CVS-HEAD version doesn't
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohammed
Salim
Sent: dinsdag 25 januari 2005 22:10
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Codec negotiation
The order matters in asterisk so if you want GSM to take priority over
G729,
simply put that ahead of the G729... so your settings should be:
Allow=all
Allow=gsm
Allow=g729
Allow=ulaw
Allow=alaw
Try that and see if it works.
Regards,
Mohammed Salim
EZZI Telecom, Inc.

--
Mark Eissler, [EMAIL PROTECTED]
Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com


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Re: [Asterisk-Users] Newbie can't dial out to pstn

[Greg]

I was just copy an example from somewhere. I made the change but the 
mobile still doesn't ring. The line the card is attached to works fine. 
here is the new output

Executing Goto("SIP/2002-4385", "mobile|0400039953|1") in new stack
-- Goto (mobile,0400039953,1)
-- Executing Goto("SIP/2002-4385", "localcall|0400039953|1") in new 
stack
-- Goto (localcall,0400039953,1)
-- Executing Dial("SIP/2002-4385", "ZAP/1/0400039953|60|r") in new 
stack
-- Called 1/0400039953
-- Zap/1-1 answered SIP/2002-4385
-- Hungup 'Zap/1-1'
  == Spawn extension (localcall, 0400039953, 1) exited non-zero on 
'SIP/2002-4385'

is this line -- Zap/1-1 answered SIP/2002-4385 displayed when the card 
tries to make the call or when the card thinks it has established the 
call?

Regards,
Greg
By the way, I'm on the Gold Coast.
On 18/03/2005, at 12:32 PM, Shane Dalgleish wrote:
Greg,
Any reason why you are putting the country code on the front for a 
mobile
call through pstn?
(Unless you have something like an Ericsson F220M Fixed Cellular 
Terminal
connected to it?)

And you said the tdm400p never tries to pick up the phone?
Have you connected a normal phone on the line and had a listen?
Where is Aus are you? :o)
Cheers
Shane
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Sent: Friday, 18 March 2005 1:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Newbie can't dial out to pstn
Hi,
I have just put in a tdm400p with 4 fxo modules and am trying
to dial out from x-lite to dial my mobile phone just to test.
The output in the asterisk console is like this
Executing Goto("SIP/2002-239b", "mobile|61400039953|1") in new stack
 -- Goto (mobile,61400039953,1)
 -- Executing Goto("SIP/2002-239b",
"localcall|61400039953|1") in new stack
 -- Goto (localcall,61400039953,1)
 -- Executing Dial("SIP/2002-239b",
"ZAP/1/61400039953|60|r") in new stack
 -- Called 1/61400039953
 -- Zap/1-1 answered SIP/2002-239b
 -- Hungup 'Zap/1-1'
   == Spawn extension (localcall, 61400039953, 1) exited
non-zero on 'SIP/2002-239b'
It never tries to pick up the phone and dial out. I'm not
sure if the config is correct, but I can easily dial between
x-lite clients, just not get the pstn.
Can anyone see any glaring mistakes?
Any help is grealty appreciated.
Regards,
Greg
My extensions.conf part is this:
exten => _04,1,GoTo(mobile,61${EXTEN:1},1)
[localcall] ; local calls by PSTN ?is a fixed charge, voip is
per minute exten => _X.,1,Dial(ZAP/1/${EXTEN},60,r) exten =>
_X.,2,Congestion exten => _X.,3,Hangup exten =>
_X.,103,Hangup exten => _X.,104,Hangup exten => _X.,105,Hangup
[mobile] ; Maybe be cheaper to route mobile calls differently
to STD in some cases exten => _X.,1,Goto(localcall,${EXTEN},1)
zaptel.conf
fxsks=1-4
loadzone=au
defaultzone=au
channels=1-4
zapata.conf
[channels]
 
busydetect=1
busycount=7
 
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
 
usecallerid=yes
 
echocancel=yes
echocancelwhenbridged=yes
 
rxgain=0.0
txgain=0.0
 
group=1
pickupgroup=1-4
 
immediate=no
 
context=incomingcall
 
signalling=fxs_ks
callerid=asreceived
channel=1-4
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RE: [Asterisk-Users] IAX Registration being lost

[Tony Davidson]

Well I think the problems solved.

After finding that the registration was lost overnight and the ADSL
connection hadn't dropped (i.e. not to do with the Dynamic IP Address) I
investigated everything else.  Seems that for some reason the port
forwarding on the firewall for port 4569 had been disabled.  Re-enabled it
and now works fine (so far :).

I still don't understand why it would work okay for an initial registration
without the port forward but then lose it later on

tony


Tony Davidson CNA CA (IT) DCE
Director, Zero Effort Networking Pty Ltd
Ph: 0411 478 004, Fax: (02) 8569 2012
http://www.zeroeffortnetworking.com.au
 


My mailbox is spam-free with ChoiceMail, the leader in personal and corporate 
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Re: [Asterisk-Users] Backing up configurations and *@home list?

[[EMAIL PROTECTED]]

no mailing list just the Discussion Forums on
http://sourceforge.net/projects/asteriskathome/
a backup of [EMAIL PROTECTED] config files and sql db is
done every night. you can download it on the admin
page


--- Don Murray <[EMAIL PROTECTED]> wrote:
> 
> Hello again,
> 
> I have [EMAIL PROTECTED] set up and working.  There are a
> couple of questions I 
> have about system administration that I couldn't
> find on the wiki 
> (although I may not be using the right buzzwords).
> 
> (1) is there an [EMAIL PROTECTED] specific mailing list?
>  I believe someone 
> on this list mentioned there was but I cannot find
> it at sourceforge, 
> the [EMAIL PROTECTED] web page, or the asterisk.org web
> page.
> 
> (2) is there a utility for backing up asterisk
> configurations and 
> current status?  I'd like to be able to do regular
> back-ups and if a big 
> problem happens, have a method to easily re-install
> the system on a 
> different machine.  Is there an *-friendly way of
> doing this or should I 
> just do directory dumps periodically and copy them
> back into place after 
> a fresh install?
> 
> Thanks
> 
> Don
> 
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Re: [Asterisk-Users] ANNOUNCEMENT: Updates forapp_cbmysqlandMeetMe2gui (out of tree modules)

[Henry Devito]

Still a no go.  I can't figure out what I am doing wrong.  This is the error 
I get

[EMAIL PROTECTED] apps]#  patch -p1 < 
/var/build_aah/asterisk_src/asterisk/apps/apps-meetme-cbmysql.txt
can't find file to patch at input line 4
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--
|diff -u asterisk-1.0.5-orig/apps/app_cbmysql.c 
asterisk-1.0.5/apps/app_cbmysql.c
|--- asterisk-1.0.5-orig/apps/app_cbmysql.c 2005-02-28 
23:19:19.0 -0800
|+++ asterisk-1.0.5/apps/app_cbmysql.c  2005-03-03 18:39:34.0 -0800
--

It asks for the file to patch if i put in /path/app_cbmysql.c
I get this error
File to patch: /var/build_aah/asterisk_src/asterisk/apps/app_cbmysql.c
patching file /var/build_aah/asterisk_src/asterisk/apps/app_cbmysql.c
can't find file to patch at input line 331
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--
|diff -u asterisk-1.0.5-orig/apps/app_meetme2.c 
asterisk-1.0.5/apps/app_meetme2.c
|--- asterisk-1.0.5-orig/apps/app_meetme2.c 2004-04-07 
04:37:18.0 -0700
|+++ asterisk-1.0.5/apps/app_meetme2.c  2005-03-03 10:31:36.0 -0800


- Original Message - 
From: "Dan Austin" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Wednesday, March 16, 2005 6:56 PM
Subject: RE: [Asterisk-Users] ANNOUNCEMENT: Updates 
forapp_cbmysqlandMeetMe2gui (out of tree modules)

Bugger.  I knew I'd screw up the patch instructions.
Try this-
#> cd to /var/build_aah/asterisk_src/asterisk
#> patch -p1 < path-to/apps-meetme-cbmysql.txt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry
Devito
Sent: Wednesday, March 16, 2005 7:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ANNOUNCEMENT: Updates for
app_cbmysqlandMeetMe2gui (out of tree modules)
Dan,  Thanks for the time helping me out.  I figured everything out
except
for the patch.
7.  cd to asterisk/apps and run patch -p0 <
path-to/apps-meetme-cbmysql.txt
When I do this step it errors out and asks for the file to patch..  When
I
look at the apps-meetme-cbmysql.txt It shows the file name to be
app_cbmysql.c so I changed the name of the file cbmysql.c to
app_cbmysql.c
but it still doesn't work.
Also in the apps-meetme-cbmysql.txt it shows the path to be
asterisk-1.0.5-orig I am trying to install this on [EMAIL PROTECTED]  So
the
source is in /var/build_aah/asterisk_src/asterisk.  Maybe I'm looking in
the
wrong direction. any help would be appreciated.  I can even give you
root
access to my box.  When all is said and done I will write up a wiki page
for
installation.
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[Asterisk-Users] Use of asterisk to make use of IP phone speakerphones as a baby monitor....

[Francois Menard (Mailing List Account)]

Is this possible?
f.
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Re: [Asterisk-Users] Broadvoice solution finally

[Luki]

> I just wanted BV users to know that a solution is on its way.
OK, great, I guess. I'm curious to see what the problem was as there
were many people reporting problems. As I stated before, I personally
don't have any problems dialing out through their LAX or DCA server,
and didn't have to make any changes for months.

--Luki
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Re: [Asterisk-Users] Newbie can't dial out to pstn

[Shaun Ewing]

On Fri, 18 Mar 2005 12:00:58 +1000, Greg
<[EMAIL PROTECTED]> wrote:

> Can anyone see any glaring mistakes?

Yes.

> My extensions.conf part is this:
> 
> exten => _04,1,GoTo(mobile,61${EXTEN:1},1)

In Australia we don't prefix calls to mobiles with 61.

You want something like:

exten => _04,1,Goto(mobile,${EXTEN},1)

If you're using a VoIP provider that requires 61, as well as routing
calls through Zap where no 61 is required, you'll have to put in some
logic to fix that up.

-Shaun
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[Asterisk-Users] Newbie can't dial out to pstn

[Greg]

Hi,
I have just put in a tdm400p with 4 fxo modules and am trying to dial 
out from x-lite to dial my mobile phone just to test.

The output in the asterisk console is like this
Executing Goto("SIP/2002-239b", "mobile|61400039953|1") in new stack
-- Goto (mobile,61400039953,1)
-- Executing Goto("SIP/2002-239b", "localcall|61400039953|1") in 
new stack
-- Goto (localcall,61400039953,1)
-- Executing Dial("SIP/2002-239b", "ZAP/1/61400039953|60|r") in new 
stack
-- Called 1/61400039953
-- Zap/1-1 answered SIP/2002-239b
-- Hungup 'Zap/1-1'
  == Spawn extension (localcall, 61400039953, 1) exited non-zero on 
'SIP/2002-239b'

It never tries to pick up the phone and dial out. I'm not sure if the 
config is correct, but I can easily dial between x-lite clients, just 
not get the pstn.

Can anyone see any glaring mistakes?
Any help is grealty appreciated.
Regards,
Greg
My extensions.conf part is this:
exten => _04,1,GoTo(mobile,61${EXTEN:1},1)
[localcall] ; local calls by PSTN ?is a fixed charge, voip is per minute
exten => _X.,1,Dial(ZAP/1/${EXTEN},60,r)
exten => _X.,2,Congestion
exten => _X.,3,Hangup
exten => _X.,103,Hangup
exten => _X.,104,Hangup
exten => _X.,105,Hangup
[mobile] ; Maybe be cheaper to route mobile calls differently to STD in 
some cases
exten => _X.,1,Goto(localcall,${EXTEN},1)

zaptel.conf
fxsks=1-4
loadzone=au
defaultzone=au
channels=1-4
zapata.conf
[channels]
 
busydetect=1
busycount=7
 
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
 
usecallerid=yes
 
echocancel=yes
echocancelwhenbridged=yes
 
rxgain=0.0
txgain=0.0
 
group=1
pickupgroup=1-4
 
immediate=no
 
context=incomingcall
 
signalling=fxs_ks
callerid=asreceived
channel=1-4
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[Asterisk-Users] Re: Snom190 intercom

[Shaun Dwyer]

Hi Josh,
Thanks for the info..
how did you get intercom=true into the URI, and onto the end of the 
INVITE line?

btw, I got an intresting response from Sven of Snom...
[Sven Fischer (support) wrote:]
Hi,
as far as I understood intercom will only work if you are not using any 
password for registering at the registrar at the moment.

But we will add a line based auto answer functionality which should enable 
intercom for our phones more easily.

regards,
Sven Fischer


Cheers,
-Shaun
Josh Dady wrote:
As you can see from the SIP trace below (from the called phone), 
intercom=true is being appended to the To: header as per requirements.

The "intercom=true" needs to be appended to the request URI, not to 
the header as a whole -- your To: header should be:

To: 
Mind you, I didn't get the phone to respond to the intercom=true until 
I added it on the request line as well, so the INVITE line of your 
request would be:

INVITE sip:1011 at 192.168.10.150:2051;line=9avrmhew;intercom=true 
SIP/2.0

I'm on a Snom 220 with SIP 3.56t, and I'm stuck on the very next step 
of the process -- answering the phone's challenge to the INVITE 
request.  The wiki indicates that the Snom needs to challenge with 
realm=snom, but even if I add snom into our internal DNS so that I can 
set the registrar to snom (that being the only way I can see to change 
what the phone uses as realm), it still rejects the digest response.  
Anyone have this working with recent loads of SIP that can shed any 
light on this?

I've email'd snom a few days ago but have yet to get a response.

According to their web page, they have a new office as of April 1, and 
I got a response to a support request (on this very issue) today 
saying that they'd likely not be able to respond until people are 
settled into the new offices, so you'll likely have to be patient with 
them.

--
Joshua P. Dady

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Re: [Asterisk-Users] How to make Span Port Selection in "Round Robin"fashion? - [SP]

[Barton Fisher]

You Da Man! - Thanks


- Original Message - 
From: "Kevin P. Fleming" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, March 17, 2005 5:01 PM
Subject: Re: [Asterisk-Users] How to make Span Port Selection in "Round
Robin"fashion? - [SP]


> Barton Fisher wrote:
> > I have span in a group (ZAP/g1) - How can I make this group sequentially
> > select ports on the span, instead always selecting port 1?
>
> Amazingly, a quick search on the wiki turned up this page:
>
> http://www.voip-info.org/wiki-Asterisk+ZAP+channels
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Re: [Asterisk-Users] How to make Span Port Selection in "Round Robin" fashion?

[Kevin P. Fleming]

Barton Fisher wrote:
I have span in a group (ZAP/g1) - How can I make this group sequentially
select ports on the span, instead always selecting port 1?
Amazingly, a quick search on the wiki turned up this page:
http://www.voip-info.org/wiki-Asterisk+ZAP+channels
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[Asterisk-Users] How to make Span Port Selection in "Round Robin" fashion?

[Barton Fisher]

I have span in a group (ZAP/g1) - How can I make this group sequentially
select ports on the span, instead always selecting port 1?

TIA

Bart



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RE: [Asterisk-Users] OT: Best DB

[Chris Albertson]

What is the best truck?  A recent survey finds that
there are far more Ford Rangr pickup trucks on the road
then there are Frightliner 18 wheelers

In another survey we find that Chevy outnumbers Porche.

Closer to home in the computer world, more people use
MS Windows than Solaris.

I think Budwieser outsells every other beer.

In most organizations followers outnumber the leaders

The poor will always outnumber the rich.

Still interrested in that database poll?

What's the "best" DB.  First you must define "best".
After you do that the answer is easy.


--- David Brodbeck <[EMAIL PROTECTED]> wrote:
> > -Original Message-
> > From: Steven Critchfield [mailto:[EMAIL PROTECTED]
> 
> > > Top Deployed Databases poll shows following databases in use: 
> > > 
> > > SQL Server with 78%, Oracle - 55%, MySQL - 33% and PostgreSQL -
> 8%. 
> > 
> > I see they created this with Mysql,
> > 78 + 55 + 44 + 8 = 185%
> > I'm sure if you add in the others we would get to something 
> > around 300%
> > deployment.
> 
> Presumably some sites had more than one type of database in use.
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  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK



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[Asterisk-Users] X-Lite and Asterisk

[Scheda]

So I'm trying to set up X-Lite for use with Asterisk, and I  can't get
it to work. I used a PDF I found on Voip-Info to set it up, and it's
still not working, I'm using Nufone as the provider... I'll include my
sip.cocnf and extentions.conf here.

extentions.conf

[CODE][outgoing]
exten => _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}



[inbound]
exten => Asteriskboxnumberhere,1,Answer
exten => Asteriskboxnumberhere,2,Background(agent-alreadyon)
exten => Asteriskboxnumberhere,3,Hangup()

[scheda]
exten => [_123456789],1,NoOp("call for "${EXTEN})
exten => [_123456789],2,Dial(SIP/${EXTEN},60,tr)
exten => [_123456789],3,Congestion[/CODE]


sip.conf

[scheda]

[CODE]type=friend
secret=mypasshere
auth=md5
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="Scheda" <2000>
disallow=all
allow=gsm
allow=ulaw
allow=alaw[/CODE]

Yes, my iax.conf is set up correctly, I can call into the box and get
the Playback() message of my choosing.
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[Asterisk-Users] Backing up configurations and *@home list?

[Jeff Glassman]

Date: Thu, 17 Mar 2005 14:06:48 -0800
From: Don Murray <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Backing up configurations and [EMAIL PROTECTED] list?
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed


Hello again,

I have [EMAIL PROTECTED] set up and working.  There are a couple of questions I 
have about system administration that I couldn't find on the wiki 
(although I may not be using the right buzzwords).

(1) is there an [EMAIL PROTECTED] specific mailing list?  I believe someone 
on this list mentioned there was but I cannot find it at sourceforge, 
the [EMAIL PROTECTED] web page, or the asterisk.org web page.


 (2) is there a utility for backing up asterisk configurations and 
current status?  I'd like to be able to do regular back-ups and if a big

problem happens, have a method to easily re-install the system on a 
different machine.  Is there an *-friendly way of doing this or should I

just do directory dumps periodically and copy them back into place after

a fresh install?

The built in AMP interface takes care of backup and restore

Thanks

Don

[EMAIL PROTECTED] info

https://sourceforge.net/forum/?group_id=123387

http://www.techdatapros.com/asterisk/


Backup and restore info

The built in AMP interface takes care of backup and restore


Jeff


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Re: [Asterisk-Users] Got 200 OK on REGISTER that isn't a register

[Kevin P. Fleming]

Ronald Wiplinger wrote:
What does that mean? Where can I get more info about it?
Mar 18 07:19:51 WARNING[9563]: chan_sip.c:7549 handle_response: Got 200 
OK on REGISTER that isn't a register
It means Asterisk received a 200 OK response to a REGISTER request, but 
it no longer has any evidence that it sent out that REGISTER request.

This can happen if the 200 OK arrives very, very late, or if it's been 
corrupted, or if the registrar doesn't respond with the proper headers 
to be able to match to the original request.
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[Asterisk-Users] Got 200 OK on REGISTER that isn't a register

[Ronald Wiplinger]

What does that mean? Where can I get more info about it?
Mar 18 07:19:51 WARNING[9563]: chan_sip.c:7549 handle_response: Got 200 
OK on REGISTER that isn't a register

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[Asterisk-Users] Broadvoice solution finally

[Kerry Garrison]

We have worked out 
all of the settings needed to get BroadVoice working. We have just completed 
testing on two seperate systems to make sure it is working. I am writing up 
details instructions right now and will have something available shortly. I just 
wanted BV users to know that a solution is on its way.
 
 
Kerry 
Garrison
http://www.geekgazette.com
 
 
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Re: [Asterisk-Users] Undocumented "exten" syntax?

[Scott Laird]

On Mar 17, 2005, at 1:36 PM, Scott Nelson wrote:
I groked the answer and it is more than pseudo code, it actually 
works.  I'm using them in my extensions.conf now!

Priority n means 1+last defined priority.  Putting a name inside 
parens defines that name.

So, in the example you sent, checkavail is set to 2, dial is 5, and I 
am not sure what s+2 means if not 7, and that doesn't make sense since 
you will leave 6 undefined.
It works, but only in CVS-HEAD, not in 1.0.x.  It's one of the reasons 
to look forward to the next major release.

Scott
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Re: [Asterisk-Users] Test post

[C F]

On Thu, 17 Mar 2005 12:05:27 -0800, Kerry Garrison
<[EMAIL PROTECTED]> wrote:
> Please ignore, intelligent messages will come later

Doesn't look like "intelligent messages" blah blah blah. You could
have waited to see the messages coming in, you didn't have to test it.
It's an idiotic thing to do.


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Re: [Asterisk-Users] PRI Cause Code Help

[Eric Knudson]

hmm - that seems correct - I remember experiencing some similar
problems(using Cisco IOS gateways, though) which were addressed by
"voice call convert-discpi-to-prog"

http://tinyurl.com/5xn8n

That command addressed an IOS gateway's tendency to hang up before the
in-band message could be played; similarly, it appears that * may not
be providing the in-band message to playback to the calling party when
an extension is out of service or something.

Eric

On Fri, 18 Mar 2005 00:06:24 +0100 (CET), Peter Svensson
<[EMAIL PROTECTED]> wrote:
> On Thu, 17 Mar 2005, Eric Knudson wrote:
> 
> > Trevor,
> >
> > Nah, I think the response is correct. Take a look at the chart again:
> >
> > http://www.lkn.ei.tum.de/lehre/kn2/anhangKap4.pdf
> >
> > look at the incoming setup procedure(1 of 2) (user side).
> >
> > If you get an incoming SETUP, then you MUST respond with one of the 
> > following:
> >
> > CALL PROCEEDING, or ALERTING, or CONNECT, or RELEASE COMPLETE. Follow
> > the rest of the logic path to completion for the rest of the call.
> 
> You are not allowed to first respond with a CALL PROCEEDING and then a
> RELEASE COMPLETE. In that case DISCONNECT is allowed I think.
> 
> > I suspect that the carrier(since they mentioned a progress indicator)
> > expects your equipment to cut through audio and play a message
> > back("we're sorry, that extension is not available, blah, blah") or
> > play a busy tone or something.
> 
> Peter
> 
>
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Re: [Asterisk-Users] What cable to connect TE110P to telco PRI ?

[Jerry]

Straight through patch cord works well. Assuming you are going from the 
card to the telco NIU. Otherwise you may need to make a crossover 
cable. Pins 1,2,4,5 are the relevant ones.

Most BRI these days are single pr so no.
On Mar 17, 2005, at 4:55 PM, Robert Rozman wrote:
Hi,
call me stupid, but cable is not delivered with te110p. What cable can 
I use
to connect to telco NT PRI line ? Is it same as for BRI interface ?

Thanks,
regards,
Rob.
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Re: [Asterisk-Users] Database families and keys

[Kevin P. Fleming]

Ed Greenberg wrote:
/SIP/Registry/216  : 192.168.1.80:5061:3600:216:sip:[EMAIL PROTECTED]:5061
If I want to delete one of these keys, I do 'help database del' and I get:
Usage: database del  
  Deletes an entry in the Asterisk database for a given
So what's a "family"? How do I do a database del  on, let's say, 211?
The family is always the first component.
CLI> database del SIP Registry/211
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[Asterisk-Users] Database families and keys

[Ed Greenberg]

When I do a database show, I get something like this:
/SIP/Registry/202  : 192.168.1.89:5060:3600:202:sip:[EMAIL PROTECTED]:5060
/SIP/Registry/203  : 192.168.1.89:5060:3600:203:sip:[EMAIL PROTECTED]:5060
/SIP/Registry/211  : 24.184.18.140:5060:60:211:sip:[EMAIL PROTECTED]:5060
/SIP/Registry/212  : 192.168.1.86:5060:60:212:sip:[EMAIL PROTECTED]:5060
/SIP/Registry/213  : 192.168.1.86:5060:3600:213:sip:[EMAIL PROTECTED]:5060
/SIP/Registry/214  : 192.168.1.81:5060:3600:214:sip:[EMAIL PROTECTED]:5060
/SIP/Registry/215  : 192.168.1.80:5060:3600:215:sip:[EMAIL PROTECTED]:5060
/SIP/Registry/216  : 192.168.1.80:5061:3600:216:sip:[EMAIL PROTECTED]:5061
If I want to delete one of these keys, I do 'help database del' and I get:
Usage: database del  
  Deletes an entry in the Asterisk database for a given
So what's a "family"? How do I do a database del  on, let's say, 211?

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Re: [Asterisk-Users] STUN Server

[Madhawa]

Hi!
Asterisk supports NAT!
http://www.voip-info.org/wiki-Asterisk+Avoid+SIP+NAT+Traversal
http://www.voip-info.org/wiki-Asterisk+sip+nat+solutions

/Madhawa


On Thu, 17 Mar 2005 18:17:14 -0500, Matt <[EMAIL PROTECTED]> wrote:
> Hi,
> Does asterisk have in itself an STUN server built in?  Or do I need to
> set one up seperately?  And if that is the case, what is recommended
> for use with asterisk (to allow VOIP users behind nats to connect to
> my VOIP servers)
> 
> Matt
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[Asterisk-Users] My appologies

[Henry Devito]

 

 

 

My apologies to the list and to everyone that had bounced
messages sent to them from my email.  Mediacom had an issue with my email
account it was bouncing messages saying it was full when it was not full. 
I apologize again if this inconvenienced anyone.

 

Henry






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RE: [Asterisk-Users] IAX Registration being lost

[Anton Krall]

Speaking of this. Why is it that sometimes the port is shown as something
differente than 4569 on some hosts? For ex.

Host  UsernamePerceived Refresh
State
210.80.176.12:221108990608214  1.2.3.4:4569 60
Registered  

And why that host changes port each time it reboots?
This is happening to one IAX box I connect to.. And it's a pain cause I have
to put the port on my dial and registry entries in order to register on it
or dial to it.

Why is that?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler
Sent: Jueves, 17 de Marzo de 2005 03:08 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] IAX Registration being lost

Tony,

Do you have port 4569 on your external firewall IP port-forwarded to your
internal IP on the * box?

You should create a port forward of the external eth1:4569 -->
192.168.100.183:4569

Assuming that you exxternal IP were something like 1.2.3.4, you should see
this when you run iax2 show registry.

Host  UsernamePerceived Refresh
State
210.80.176.12:45698990608214  1.2.3.4:4569 60
Registered 

Thanks,
Wiley



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony Davidson
Sent: Thursday, March 17, 2005 1:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] IAX Registration being lost

Well, this is getting more interesting.  I started looking at this this
morning and realised that Asterisk had lost registration, yet my ADSL
connection has been up for almost 2 days - and it was working fine
yesterday.  Therefore it doesn't appear to be related to the IP address
changing.

I'm thinking it's more that the registration is lost for any reason (such as
an ADSL reconnect or the registration needing to be refreshed) and it won't
come back.  Get this message as before:

Host  UsernamePerceived Refresh  State
210.80.176.12:45698990608214   60  Request
Sent

I tried a ping and a traceroute and both working fine.  An ifconfig just
shows the internal address (192.168.100.183).


Tony Davidson CNA CA (IT) DCE
Director, Zero Effort Networking Pty Ltd
Ph: 0411 478 004, Fax: (02) 8569 2012
http://www.zeroeffortnetworking.com.au
 

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Wiley 
> Siler
> Sent: Thursday, 17 March 2005 9:19 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] IAX Registration being lost
> 
> What doesn't make sense about that is that if you are setup like 
> this...
> 
> DSL Router ---> Your Firewall/Router ---> Asterisk Box
> 
> Then the issue of being dynamic will not matter to the * box. 
>  IP storing is mute since the end point and start point are not 
> changing.
> All that is changing is the IP on the outside of your Firewall/Router 
> and thus a momentary loss of connectivity.
> AAH would not care about that in relation to what it has stored.  It 
> will just attempt the registration and pass data to the gateway 
> (inside interface of your
> FW/Router) just like before.  As far as it is concerned, nothing has 
> changed except now the attempt to communicate outward dies on the 
> first hop until the new IP is assigned to the external interface of 
> your FW.
> 
> Try this.  Start some IAX debug in the CLI the next time it happens.
> Tracert your IAX target and see if you can get to it.
> Ifconfig the interface to see what is setup.
> 
> Report back.
> 
> Thanks,
> Wiley
> 
> 


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[Asterisk-Users] STUN Server

[Matt]

Hi,
Does asterisk have in itself an STUN server built in?  Or do I need to
set one up seperately?  And if that is the case, what is recommended
for use with asterisk (to allow VOIP users behind nats to connect to
my VOIP servers)

Matt
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[Asterisk-Users] My appologies

[Henry Devito]

My apologies to the list and to everyone that had bounced
messages sent to them from my email.  Mediacom had an issue with my email
account it was bouncing messages saying it was full when it was not full.  I apologize
again if this inconvenienced anyone.

 

Henry






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Re: [Asterisk-Users] PRI Cause Code Help

[Peter Svensson]

On Thu, 17 Mar 2005, Eric Knudson wrote:

> Trevor,
> 
> Nah, I think the response is correct. Take a look at the chart again:
> 
> http://www.lkn.ei.tum.de/lehre/kn2/anhangKap4.pdf
> 
> look at the incoming setup procedure(1 of 2) (user side).
> 
> If you get an incoming SETUP, then you MUST respond with one of the following:
> 
> CALL PROCEEDING, or ALERTING, or CONNECT, or RELEASE COMPLETE. Follow
> the rest of the logic path to completion for the rest of the call.

You are not allowed to first respond with a CALL PROCEEDING and then a
RELEASE COMPLETE. In that case DISCONNECT is allowed I think.

> I suspect that the carrier(since they mentioned a progress indicator)
> expects your equipment to cut through audio and play a message
> back("we're sorry, that extension is not available, blah, blah") or
> play a busy tone or something.

Peter


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Re: [Asterisk-Users] PRI Cause Code Help

[Peter Svensson]

On Thu, 17 Mar 2005, Trevor Peirce wrote:

> Trevor Peirce wrote:
> 
> > Anyhow, they are seeing the RELEASE COMPLETE message with cause code 
> > 1, however the tech told me they expect a PROGRESS indicator with a 
> > value between 1 and 10.
> 
> Okay after printing off a dozen pages and taking up tons of floor space 
> I think I may have figured this out... but this being my first day in 
> PRI-language I'd appreciate some feed back if I'm on the right track.
> 
> First, I am seeing
> 
> < SETUP
>  > CALL PROCEEDING
> < FACILITY (w/ CID Name)
>  > RELEASE COMPLETE (Cause: 1)
> 
>  From what I understand from Q.931's figure A.2, is that the CALL 
> PROCEEDING should *not* be sent to do this... and if it is we've got to 
> addto the dialog slightly

Call proceeding is allowed even if the number will later turn out to be 
unallocated, I think. 

RELEASE COMPLETE is valid in a few specified circumstances enumerated in 
5.3.2 in q.931. These are situations where the indicated B-channel is 
unavailable or unacceptable. 

Normal causes such as busy, no such number or congestion (apart from a 
shortage of B-channels on the link) should follow the general call 
rejection pattern with a DISCONNECT described in 5.3.3.

Futhermore the SDL diagrams in Figure A.2(b) and A.2(d) indicate that 
RELEASE COMPLETE is only allowed as a response to SETUP and that later 
clearing is done through DISCONNECT. I.e. clearing after a CALL PROCEEDING 
should be done with DISCONNECT.

> 
> < SETUP
>  > CALL PROCEEDING
> < FACILITY (w/ CID Name)
>  > DISCONNECT (Cause: 1)
> < RELEASE
>  > RELEASE COMPLETE (Cause: 1)
> 
> Someone please let me know if I'm talking nonsense or if I am going 
> somewhere with this...

Have a look at q931_hangup() in q931.c. Small modifications should fix 
your problem.

Peter


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[Asterisk-Users] What cable to connect TE110P to telco PRI ?

[Robert Rozman]

Hi,
call me stupid, but cable is not delivered with te110p. What cable can I use
to connect to telco NT PRI line ? Is it same as for BRI interface ?
Thanks,
regards,
Rob.
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RE: [Asterisk-Users] IAX2 VOIP HARDPHONE

[Kanuri, Seshu (Company IT)]

These are ATCOM Phones. Integrated Networks is an invented name and he
is a reseller from china like this IAXtalk guy in China. 

They are not the manufacturers. 

These ATCOM Phones are available and distributed in USA from
http://www.eezeephone.com
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Androtech
Sent: Thursday, March 17, 2005 5:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX2 VOIP HARDPHONE

Dear All

Does someone experienxed this phone using the IAX2 protocol with
Asterisk?
http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks
Regards, 

 
NOTICE: If received in error, please destroy and notify sender.  Sender does 
not waive confidentiality or privilege, and use is prohibited. 
 
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[Asterisk-Users] Atxfer not working for called party

[Ivan Barrera A.]

Hi.
I've been trying to develop this module since some time now.
CVS already has a dial version with atxfer. When trying this, using the 
modifiers tT and having configures features.conf accordingly, i havent 
been able to use such a feature in the called party.
I also tried using t and T separately.

I've tried to understand why this happens, and started to watch the 
"copy" of the frames in the bridge. I noticed only the frames originated 
from the calling party are analized for DTMF, dont know why
When using stable asterisk, with normal Dial, the transfer function 
works ok with both parties.

I Dont understand where is the problem, and if any have any ideas on how 
i can work it out, please tell me.

Thanks in advance
Bruce.-
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[Asterisk-Users] Re: Asterisk with Cisco Call Manager

[Maron Kristófersson]

At least in Call Manager 4 you can setup a SIP trunk between CCM and *. 
 Details are in the wiki.

Regards,
Maron
Mohamed Farid wrote:
Dear All :
We need to use the Conference Room Capability from Asterisk to use it 
with our IPT Solution which based on Cisco Call Manager..

Also we need to use most of Asterisk features in our IPT Network ..
How can I do this ? Any help will be grateful ..
 

Mohamed Farid ,,
Telecommunication & Security Administrator ,,
 

 

Notice:
This e-mail (including attachments) is confidential and is intended 
solely for the addressee. Unless you are the addressee, you may not 
read, copy, use or store this e-mail in any way, or permit others to. If 
you have received it in error, please contact Mediterranean Smart Cards 
Company :+202 333 1400


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[Asterisk-Users] Message waiting/station busy conflict?

[David Josephson]

Greetings list,
We are having a puzzle with * (asteriskathome 0.5) and SIP phones 
(SPA2000 ATA's). If callwaiting is enabled, everything (including call 
waiting) is normal. If callwaiting is turned off, the phone will not 
accept incoming calls and the call goes straight to whatever is 
programmed for the busy voicemail response. It doesn't matter whether 
reinvite is on or off, or whether the phones are registered or not.  
database show looks like

/CALLTRACE/2369   : 2368
/CALLTRACE/2370   : 2368
/CW/2368  : ENABLED
/CW/2369  : ENABLED
/SIP/Registry/2368: 
6x.xxx.xx.xx:5060:3600:2368:sip:[EMAIL PROTECTED]:5060
/SIP/Registry/2369: 
6x.xxx.xx.xx:5061:3600:2369:sip:[EMAIL PROTECTED]:5061

Here is the relevant snip from the log. What did I leave out?
Mar 17 12:12:09 VERBOSE[1348]:   == Manager 'admin' logged on from 127.0.0.1
Mar 17 12:12:09 DEBUG[1348]: Manager received command 'command'
Mar 17 12:12:09 DEBUG[1348]: Manager received command ''
Mar 17 12:12:09 DEBUG[1348]: Manager received command 'Logoff'
Mar 17 12:12:09 VERBOSE[1348]:   == Manager 'admin' logged off from 
127.0.0.1
Mar 17 12:12:09 DEBUG[1348]: Unable to find key '2369' in family 'CW'
Mar 17 12:12:09 VERBOSE[1348]:   dialparties.agi: Extension 2369 has 
call waiting disabled
Mar 17 12:12:09 DEBUG[1348]: Unable to find key '2369' in family 'CFB'
Mar 17 12:12:09 VERBOSE[1348]:   dialparties.agi: Max calls of 1 
exceeded - deleting from dial
Mar 17 12:12:09 VERBOSE[1348]:   dialparties.agi: Dial string is empty - 
nothing to do
Mar 17 12:12:09 VERBOSE[1348]:   dialparties.agi: Was direct call, 
jumping to priority 23
Mar 17 12:12:09 VERBOSE[1348]: -- AGI Script Executing Application: 
(NoOp) Options: ()
Mar 17 12:12:09 VERBOSE[1348]: -- AGI Script dialparties.agi 
completed, returning 0
Mar 17 12:12:09 VERBOSE[1348]: -- Executing 
ESC[1;36;40mWaitESC[0;37;40m("ESC[1;35;40mSIP/2368-f359ESC[0;37;40m", 
"ESC[1;3
5;40m2ESC[0;37;40m") in new stack
Mar 17 12:12:11 VERBOSE[1348]: -- Executing 
ESC[1;36;40mVoiceMailESC[0;37;40m("ESC[1;35;40mSIP/2368-f359ESC[0;37;40m", "
ESC[1;35;40mb2369ESC[0;37;40m") in new stack
Mar 17 12:12:11 DEBUG[1348]: Difference is 39520, ms is 4960
Mar 17 12:12:11 VERBOSE[1348]: -- Playing 
'voicemail/default/2369/busy' (language 'en')

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Re: [Asterisk-Users] PRI Cause Code Help

[Eric Knudson]

Trevor,

Nah, I think the response is correct. Take a look at the chart again:

http://www.lkn.ei.tum.de/lehre/kn2/anhangKap4.pdf

look at the incoming setup procedure(1 of 2) (user side).

If you get an incoming SETUP, then you MUST respond with one of the following:

CALL PROCEEDING, or ALERTING, or CONNECT, or RELEASE COMPLETE. Follow
the rest of the logic path to completion for the rest of the call.

I suspect that the carrier(since they mentioned a progress indicator)
expects your equipment to cut through audio and play a message
back("we're sorry, that extension is not available, blah, blah") or
play a busy tone or something.

Hope that helps, but unfortunately, I don't have enough experience
with * to troubleshoot this much more.

Eric




On Thu, 17 Mar 2005 11:26:12 -0800, Trevor Peirce <[EMAIL PROTECTED]> wrote:
> Trevor Peirce wrote:
> 
> > Anyhow, they are seeing the RELEASE COMPLETE message with cause code
> > 1, however the tech told me they expect a PROGRESS indicator with a
> > value between 1 and 10.
> 
> Okay after printing off a dozen pages and taking up tons of floor space
> I think I may have figured this out... but this being my first day in
> PRI-language I'd appreciate some feed back if I'm on the right track.
> 
> First, I am seeing
> 
> < SETUP
>  > CALL PROCEEDING
> < FACILITY (w/ CID Name)
>  > RELEASE COMPLETE (Cause: 1)
> 
>  From what I understand from Q.931's figure A.2, is that the CALL
> PROCEEDING should *not* be sent to do this... and if it is we've got to
> addto the dialog slightly
> 
> < SETUP
>  > CALL PROCEEDING
> < FACILITY (w/ CID Name)
>  > DISCONNECT (Cause: 1)
> < RELEASE
>  > RELEASE COMPLETE (Cause: 1)
> 
> Someone please let me know if I'm talking nonsense or if I am going
> somewhere with this...
> 
> Many thanks,
> Trevor Peirce
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RE: [Asterisk-Users] Backing up configurations and *@home list?

[Wiley Siler]

I cannot answer q1 and am interested in this myself.

Question 2 has a partial answer in that the AAH has a backup feature
located in the management portion of AMP.
The backup link is at the bottom.  The restore feature is located at the
linux command line on a AAH machine. 

help-aah will show the command.  It is restore-aah.  Have not used it
yet so I cannot attest to its efficacy...

W



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don Murray
Sent: Thursday, March 17, 2005 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Backing up configurations and [EMAIL PROTECTED] list?


Hello again,

I have [EMAIL PROTECTED] set up and working.  There are a couple of questions I
have about system administration that I couldn't find on the wiki
(although I may not be using the right buzzwords).

(1) is there an [EMAIL PROTECTED] specific mailing list?  I believe someone
on this list mentioned there was but I cannot find it at sourceforge,
the [EMAIL PROTECTED] web page, or the asterisk.org web page.

(2) is there a utility for backing up asterisk configurations and
current status?  I'd like to be able to do regular back-ups and if a big
problem happens, have a method to easily re-install the system on a
different machine.  Is there an *-friendly way of doing this or should I
just do directory dumps periodically and copy them back into place after
a fresh install?

Thanks

Don

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Re: [Asterisk-Users] RE: Caller ID on E&M Wink

[Dan Davis]

Caller ID is not available on a T1 CAS line because it does not support the
privacy indicator. however, ANI is available if configured and can be DTMF
or MF.

- Original Message - 
From: "Jason Kawakami" <[EMAIL PROTECTED]>
To: 
Sent: Thursday, March 17, 2005 4:54 PM
Subject: [Asterisk-Users] RE: Caller ID on E&M Wink


>
>
> -Original Message-
> 
> I know how Caller ID information is passed on an analog phone line
> (between the rings) but with a T1 line, I don't know technically how it
> is done.
>
> CallerID is not normally available on a T-1 but is more designed to come
in
> on a PRI.  Look into featd and featdmf signaling in Zapata.conf to get ANI
> on a t-1 span.
>
> Jason Kawakami
> www.optellabs.com
> Salt Lake City, UT
>
>
>
>
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[Asterisk-Users] IAX2 VOIP HARDPHONE

[Androtech]

Dear All
Does someone experienxed this phone using the IAX2 protocol with Asterisk?
http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks
Regards,
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Re: [Asterisk-Users] PRI Cause Code Help

[Trevor Peirce]

James Sizemore wrote:
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to 
;work
; with all telcos.
; outofband:  Signal Busy/Congestion out of band with 
;RELEASE/DISCONNECT
; inband: Signal Busy/Congestion using in-band tones
priindication = outofband
Asterisk wouldn't be sending a DISCONNECT or RELEASE COMPLETE message if 
that wasn't enabled... my configuration is fine, it's the telco and 
asterisk disagreeing on how to handle the call.


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[Asterisk-Users] softphones and extensions status

[Androtech]

Dear All,
there is a software phone that shows the extensions status when is 
registered to Asterisk?

Regards,
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[Asterisk-Users] Backing up configurations and *@home list?

[Don Murray]

Hello again,
I have [EMAIL PROTECTED] set up and working.  There are a couple of questions I 
have about system administration that I couldn't find on the wiki 
(although I may not be using the right buzzwords).

(1) is there an [EMAIL PROTECTED] specific mailing list?  I believe someone 
on this list mentioned there was but I cannot find it at sourceforge, 
the [EMAIL PROTECTED] web page, or the asterisk.org web page.

(2) is there a utility for backing up asterisk configurations and 
current status?  I'd like to be able to do regular back-ups and if a big 
problem happens, have a method to easily re-install the system on a 
different machine.  Is there an *-friendly way of doing this or should I 
just do directory dumps periodically and copy them back into place after 
a fresh install?

Thanks
Don
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RE: [Asterisk-Users] Caller ID on E&M Wink

[mattf]

With a RBS(Robbed-bit) T1(in other words, not a PRI) the CallerID(Called
ANI) is sent in the digits and come across in Asterisk as part of the
extension. It is not standard, you do need to ask for it to be enabled and
you usually have to specify how you want it.

A standard way of receiving ANI on a RBS T1 is this: *NXXNXX*DNIS where
NXXNXX is the callerID and the DNIS is the last 4 digits of the number
the caller dialed.

There is no option of receiving callerIDname with RBS T1s but you do get
that 24th channel to use for voice that you don't get with a PRI.

Hope this helps,

MATT---

-Original Message-
From: Michael Welter [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 17, 2005 4:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Caller ID on E&M Wink


Scott Nelson wrote:
> I am an Asterisk newby, and I cannot seem to get Caller ID information 
> from our T1 line.   When calls appear at the phones, they say the call 
> came from "asterisk" and unknown number.
> 
> I know how Caller ID information is passed on an analog phone line 
> (between the rings) but with a T1 line, I don't know technically how it 
> is done.
> 
With an E&M T-1, I think CallerID number comes across as DTMF digits. 
That's the difference with a PRI circuit--you get the caller name as 
well as caller number.


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[Asterisk-Users] ASTCC dialstatus confusing billing issue

[Clive]

Hi

I wonder if anyone else has noticed this, or has an explanation

When a call ends with dialstatus=cancel ,one would expect that the call 
never went through, BUT it seems that sometimes a call does go 
through sucessfully, and ends with dialstatus=cancel and I have no idea 
why.??..very strange.

The problem this introduces to ASTCC is that it does not bill for these 
calls if it sees the "Cancel". This can be fixed easily, but I am still 
baffled why the "cancel" comes through, and I am wondering if anyone 
else has had a similar experience.

Regards
Clive



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[Asterisk-Users] RE: Caller ID on E&M Wink

[Jason Kawakami]

-Original Message-

I know how Caller ID information is passed on an analog phone line 
(between the rings) but with a T1 line, I don't know technically how it 
is done.

CallerID is not normally available on a T-1 but is more designed to come in
on a PRI.  Look into featd and featdmf signaling in Zapata.conf to get ANI
on a t-1 span.

Jason Kawakami
www.optellabs.com
Salt Lake City, UT




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Re: [Asterisk-Users] Caller ID on E&M Wink

[Michael Welter]

Scott Nelson wrote:
I am an Asterisk newby, and I cannot seem to get Caller ID information 
from our T1 line.   When calls appear at the phones, they say the call 
came from "asterisk" and unknown number.

I know how Caller ID information is passed on an analog phone line 
(between the rings) but with a T1 line, I don't know technically how it 
is done.

With an E&M T-1, I think CallerID number comes across as DTMF digits. 
That's the difference with a PRI circuit--you get the caller name as 
well as caller number.

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Re: [Asterisk-Users] Undocumented "exten" syntax?

[Howard Lowndes]

On Fri, 2005-03-18 at 08:34, Asterisk wrote:
> John Goerzen wrote:
> > Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
> > extensions.conf lines:
> > 
> > exten => s,1,SetVar(SET_EMERG_FLAG=0)
> > exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
> > exten => s,n,SetGlobalVar(EMERGENCY=1)
> > exten => s,n,SetVar(SET_EMERG_FLAG=1)
> > exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
> > exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)
> > 
> > Now, I have several questions:
> > 
> >  * What is the "n" priority and how can they use it for several
> >different items?  Don't they need an increasing integer there?
> 
> n stands for the next available sequence after the previous one. This is 
>   used so that you can insert a step in the dial plan without having to 
> renumber all the following steps.
> 
> > 
> >  * What is the "(checkavail)" doing?
> > 
> 
> Thats a label. If you use n, you lose the ability to say "goto 102"

You can mix explicit priorities (123) with relative priorities (n)

> 
> >  * What does "s+2" mean?


This priority + 2, thus "n" == s+1

> 
> ok. That's fooled me.
> 
> > 
> > I've tried looking in docs and the wiki but can't figure it out.
> > 
> > Thanks!
> > 
> > -- John
> > 
> > ___
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Re: [Asterisk-Users] Caller ID on E&M Wink

[Eric Knudson]

Scott,

I'm new to *, but I know T1s pretty well - most likely you're not
getting that information with your wink-start trunk. Most of the time,
you're only getting the DNIS (dialed number), not the ANI (calling
party). Here's a Cisco doc that explains CAS circuits a bit:

http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800e2560.shtml

Eric


On Thu, 17 Mar 2005 13:17:37 -0600, Scott Nelson <[EMAIL PROTECTED]> wrote:
> I am an Asterisk newby, and I cannot seem to get Caller ID information
> from our T1 line.   When calls appear at the phones, they say the call
> came from "asterisk" and unknown number.
> 
> I know how Caller ID information is passed on an analog phone line
> (between the rings) but with a T1 line, I don't know technically how it
> is done.
> 
> I don't see the caller's number in the CDR records either.
> 
> What am I missing -- where shall I search?
> --
> Scott Nelson
> Thermeon Corporation - St. Louis
> 
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RE: [Asterisk-Users] Do you need to recompile the Linux 2.6kernelfor zaptel modules?

[Geoff Nordli]

Paul you are a genius!

Make prepare was the command.  Once I ran that I was able to compile the
zaptel modules with the make linux26 command.

Have a great day!

Geoff

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Paul Hewlett
> Sent: Thursday, March 17, 2005 12:55 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Do you need to recompile the Linux
> 2.6kernelfor zaptel modules?
> 
> On Thursday 17 March 2005 06:13, Geoff Nordli wrote:
> > > -Original Message-
> > > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > > [EMAIL PROTECTED] On Behalf Of David Uzzell
> > > Sent: Wednesday, March 16, 2005 4:15 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] Do you need to recompile the Linux 2.6
> > > kernelfor zaptel modules?
> > >
> > > Geoff Nordli wrote:
> > > > Hi Everyone.
> > > >
> > > > On the Linux 2.6 kernel do I need to recompile the kernel in order
> to
> > > > compile the zaptel modules?
> > >
> > > Using Mandrake 10.1 which runs standard a 2.6 kernel I did not have to
> > > recompile the kernel to get them working.
> > >
> > > cheers,
> > >
> > > David
> >
> > In order for me to get the zaptel source to compile from cvs I need to:
> >
> > 1) Copy the .config file from the /boot directory to the source
> directory,
> > run menuconfig, and run make.
> >
> > 2) After I run make I can then compile the zaptel modules.
> >
> > 3) I don't need to install the new kernel, but just run the make
> command.
> 
> Hi Guys
> 
>  I am new to this list but a similar problem came up at the asterisk
> training
> course I attended last week. When the linux kernel sources are freshly
> installed, the symbolic links for the kernel header files are not properly
> set up. All you have to do is type:
> 
>  make prepare
> 
> in /usr/src/linux to set this all up. You do not have to compile the
> kernel or
> anything else. I do not know whether this works for the 2.4 kernel but it
> does work on Fedora 2 and SuSe 9.1 and later.
> 
> Paul H
> 
> 
> --
> Paul Hewlett (Linux #359543)  Email:`echo [EMAIL PROTECTED] | rev`
> Tel: +27 21 852 8812  Cel: +27 72 719 2725  Fax: +27 86 672 0563
> --

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[Asterisk-Users] adding to asterisk db from a script

[Wolfgang S. Rupprecht]

It looks like asterisk isn't honoring EOF on stdin.

file add-phonelist: 

database put cidname 200551234 "name 1"
database put cidname 200551235 "name 2"
database put cidname 200551236 "name 3"
database put cidname 200551237 "name 4"

asterisk -rn < add-phonelist

What I see is an infinite stream of prompts as asterisk is banging
onto the EOF.

How is one supposed to add a bulk list of clid names to asterisk?

-wolfgang
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Re: [Asterisk-Users] Three way calling with X-Lite / MeetMe

[Ed Greenberg]

Try SJPhone. (www.sjlabs.com)
--On Tuesday, March 15, 2005 3:00 PM + Chris Blunt 
<[EMAIL PROTECTED]> wrote:


Hi All,

Does any one know of a way to make a three way call from Asterisk using
X-Lite.

I need the ability to be able to call someone on the PSTN using my IAX
provider then bring another person from a local extension into the call
if needs be?

I believe most three way calling is done using a feature of the phone,
and X-Lite doesn't look like it supports this.  Can this be achieved with
MeetMe or AppConference, if it can please tell me how J

Many thanks

Chris

--

Chris Blunt



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Re: [Asterisk-Users] OT: PC sound hardware for voice recording

[Matt Riddell]

snacktime wrote:
Thanks for the tip.  I was worried someone was going to say it
couldn't be done on a budget..  Those look pretty nice from what I was
reading on them.
I'd go as far as to say that the sound card isn't where you're going to 
lose quality.  Remembering that it goes down to 8000hz, I think the best 
thing for recording clean prompts is to use a) a reasonable microphone 
(I.E. even an SM58 would do), b) record at 24-bit, 44.1 if poss, c) 
multi-band compress the life out of it and d) record in a quiet environment.

That way you will have some good quality recordings that you can then 
downsample into gsm/g729/ulaw etc...

Also, don't forget to keep the original recordings before you did 
anything to them :)

For multiband compression you could use a) isotope ozone, b) T-Racks or 
c) TC Electronics plug-ins.

If you have all your files recorded and split (believe me splitting is 
the most time consuming part - I like to split with a few samples spare 
at each end and then I do a batch process to fade in the first 10 
samples and fade out the last 10 samples - this makes sure you don't 
have clicks when moving from one to the other), then you could send them 
to me and I will chuck them through a batch process (in Wavelab) for 
compression, topping and tailing.  Bear in mind that if your are going 
to be compressing, you would need to make sure that there is really no 
background noise as compression will bring up the noise floor.  The 
normal culprit is the computer fan noise.

Oh by the way I can do the compression et al for free if you want it.
If however you are unable to source a reasonable microphone for a day 
(you should be able to borrow one from a band or venue) then it would be 
a bad idea to compress everything.

Anyway let me know if you need any help or have any further questions.
Hope I haven't muddied the waters too much!
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Agent won't log out!

[Asterisk]

Oh, that took me a while to understand ... good one.
Julian
Tim Pushor wrote:
Go talk to the oracle 

[EMAIL PROTECTED] wrote:
Hey guys... one last thing.
I have set up agents in my Asterisk... and one agent refuses to log out.
I have tried to log out from Xlite.  I have tried from the console...
AGENT LOGOFF 1001.  It still gets the call.
If I shut down Xlite, it still tries to contact agent 1001, but gets a
congested message... if I bring Xlite back up, it gets the call.  If I
kill Asterisk and restart... its _STILL_ logged on.
How can I kill the agent that will NOT DIE!?!
Steve
Stephen Amadei
5114 Harbor Beach Blvd
Brigantine Beach, NJ 08203
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Re: [Asterisk-Users] Agent won't log out!

[Tim Pushor]

Go talk to the oracle 

[EMAIL PROTECTED] wrote:
Hey guys... one last thing.
I have set up agents in my Asterisk... and one agent refuses to log out.
I have tried to log out from Xlite.  I have tried from the console...
AGENT LOGOFF 1001.  It still gets the call.
If I shut down Xlite, it still tries to contact agent 1001, but gets a
congested message... if I bring Xlite back up, it gets the call.  If I
kill Asterisk and restart... its _STILL_ logged on.
How can I kill the agent that will NOT DIE!?!
Steve
Stephen Amadei
5114 Harbor Beach Blvd
Brigantine Beach, NJ 08203
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Re: [Asterisk-Users] Undocumented "exten" syntax?

[Scott Nelson]

I groked the answer and it is more than pseudo code, it actually works. 
 I'm using them in my extensions.conf now!

Priority n means 1+last defined priority.  Putting a name inside parens 
defines that name.

So, in the example you sent, checkavail is set to 2, dial is 5, and I 
am not sure what s+2 means if not 7, and that doesn't make sense since 
you will leave 6 undefined.

On Mar 17, 2005, at 3:23 PM, Sean Kennedy wrote:
John Goerzen wrote:
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten => s,1,SetVar(SET_EMERG_FLAG=0)
exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten => s,n,SetGlobalVar(EMERGENCY=1)
exten => s,n,SetVar(SET_EMERG_FLAG=1)
exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)
Now, I have several questions:
* What is the "n" priority and how can they use it for several
  different items?  Don't they need an increasing integer there?
* What is the "(checkavail)" doing?
* What does "s+2" mean?
I've tried looking in docs and the wiki but can't figure it out.
Thanks!
-- John
I have been curious about this as well.  I was thinking it may be 
pseudo code?  It seems easy enough to read, so that may be what it is.
*shrug* I'd like to know the answer to this as well.

Sean
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--
Scott Nelson
Thermeon Corporation - St. Louis
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Re: [Asterisk-Users] Undocumented "exten" syntax?

[Asterisk]

John Goerzen wrote:
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten => s,1,SetVar(SET_EMERG_FLAG=0)
exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten => s,n,SetGlobalVar(EMERGENCY=1)
exten => s,n,SetVar(SET_EMERG_FLAG=1)
exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)
Now, I have several questions:
 * What is the "n" priority and how can they use it for several
   different items?  Don't they need an increasing integer there?
n stands for the next available sequence after the previous one. This is 
 used so that you can insert a step in the dial plan without having to 
renumber all the following steps.

 * What is the "(checkavail)" doing?
Thats a label. If you use n, you lose the ability to say "goto 102"
 * What does "s+2" mean?
ok. That's fooled me.
I've tried looking in docs and the wiki but can't figure it out.
Thanks!
-- John
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[Asterisk-Users] Agent won't log out!

[asterisk]

Hey guys... one last thing.

I have set up agents in my Asterisk... and one agent refuses to log out.
I have tried to log out from Xlite.  I have tried from the console...
AGENT LOGOFF 1001.  It still gets the call.

If I shut down Xlite, it still tries to contact agent 1001, but gets a
congested message... if I bring Xlite back up, it gets the call.  If I
kill Asterisk and restart... its _STILL_ logged on.

How can I kill the agent that will NOT DIE!?!

Steve
Stephen Amadei
5114 Harbor Beach Blvd
Brigantine Beach, NJ 08203

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Re: [Asterisk-Users] PRI Test Equipment

[izo]

If you have some money to spend buy used aurora isdn tester they are
really handy and
they can do lots of stuff including access to voice channels.

Home grown will never be as good as ready to go equipment IMHO.

regards
m.
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Re: [Asterisk-Users] Undocumented "exten" syntax?

[Sean Kennedy]

John Goerzen wrote:
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten => s,1,SetVar(SET_EMERG_FLAG=0)
exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten => s,n,SetGlobalVar(EMERGENCY=1)
exten => s,n,SetVar(SET_EMERG_FLAG=1)
exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)
Now, I have several questions:
* What is the "n" priority and how can they use it for several
  different items?  Don't they need an increasing integer there?
* What is the "(checkavail)" doing?
* What does "s+2" mean?
I've tried looking in docs and the wiki but can't figure it out.
Thanks!
-- John
I have been curious about this as well.  I was thinking it may be pseudo 
code?  It seems easy enough to read, so that may be what it is. 

*shrug* I'd like to know the answer to this as well.
Sean
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Re: [Asterisk-Users] Channel name (and substring)

[Thomas Andrews]

On Thu, Mar 17, 2005 at 12:24:00PM -0800, Sean Kennedy wrote:

> Thomas Andrews wrote:
> 
> >How do I get the bit like "IAX2/white_phone" in extensions.conf eg from
> >pre-defined variables or variants thereof ?
> >
> >What I *do* get is strings like this "IAX2/[EMAIL PROTECTED]"
> >from ${CHANNEL}, but that's the full channel name.
>
> This should help:http://www.voip-info.org/wiki-Asterisk+variables

Thanks, that's more or less what I thought I would have to do. It just
seems such a long way around to get something that I would have expected
to be 'ready-made' in some pre-defined $variable.

Thanks,
Thomas
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RE: [Asterisk-Users] IAX Registration being lost

[Wiley Siler]

Tony,

Do you have port 4569 on your external firewall IP port-forwarded to
your internal IP on the * box?

You should create a port forward of the external eth1:4569 -->
192.168.100.183:4569

Assuming that you exxternal IP were something like 1.2.3.4, you should
see this when you run iax2 show registry.

Host  UsernamePerceived Refresh
State
210.80.176.12:45698990608214  1.2.3.4:4569 60
Registered 

Thanks,
Wiley



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Davidson
Sent: Thursday, March 17, 2005 1:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] IAX Registration being lost

Well, this is getting more interesting.  I started looking at this this
morning and realised that Asterisk had lost registration, yet my ADSL
connection has been up for almost 2 days - and it was working fine
yesterday.  Therefore it doesn't appear to be related to the IP address
changing.

I'm thinking it's more that the registration is lost for any reason
(such as an ADSL reconnect or the registration needing to be refreshed)
and it won't come back.  Get this message as before:

Host  UsernamePerceived Refresh  State
210.80.176.12:45698990608214   60  Request
Sent

I tried a ping and a traceroute and both working fine.  An ifconfig just
shows the internal address (192.168.100.183).


Tony Davidson CNA CA (IT) DCE
Director, Zero Effort Networking Pty Ltd
Ph: 0411 478 004, Fax: (02) 8569 2012
http://www.zeroeffortnetworking.com.au
 

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Wiley 
> Siler
> Sent: Thursday, 17 March 2005 9:19 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] IAX Registration being lost
> 
> What doesn't make sense about that is that if you are setup like 
> this...
> 
> DSL Router ---> Your Firewall/Router ---> Asterisk Box
> 
> Then the issue of being dynamic will not matter to the * box. 
>  IP storing is mute since the end point and start point are not 
> changing.
> All that is changing is the IP on the outside of your Firewall/Router 
> and thus a momentary loss of connectivity.
> AAH would not care about that in relation to what it has stored.  It 
> will just attempt the registration and pass data to the gateway 
> (inside interface of your
> FW/Router) just like before.  As far as it is concerned, nothing has 
> changed except now the attempt to communicate outward dies on the 
> first hop until the new IP is assigned to the external interface of 
> your FW.
> 
> Try this.  Start some IAX debug in the CLI the next time it happens.
> Tracert your IAX target and see if you can get to it.
> Ifconfig the interface to see what is setup.
> 
> Report back.
> 
> Thanks,
> Wiley
> 
> 


My mailbox is spam-free with ChoiceMail, the leader in personal and
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[Asterisk-Users] Undocumented "exten" syntax?

[John Goerzen]

Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:

exten => s,1,SetVar(SET_EMERG_FLAG=0)
exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten => s,n,SetGlobalVar(EMERGENCY=1)
exten => s,n,SetVar(SET_EMERG_FLAG=1)
exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)

Now, I have several questions:

 * What is the "n" priority and how can they use it for several
   different items?  Don't they need an increasing integer there?

 * What is the "(checkavail)" doing?

 * What does "s+2" mean?

I've tried looking in docs and the wiki but can't figure it out.

Thanks!

-- John

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