Re: [Asterisk-Users] Encrypted VOIP?

[Howard Lowndes]

On Sat, 2005-02-05 at 12:33, Steven P. Donegan wrote:
> I have done that extensively (H.323 and SIP over IPSEC tunnels) I was 
> more interested in the possibilities of 'native' support of some kind. 
> But thank you very much for the response.

Isn't there a fairly significant overhead with this, given the small
size if the IAX2 datagrams?

> 
> dean collins wrote:
> 
> >Just run point to point encryption over a vpn.
> >
> >
> >
> >-Original Message-
> >From: [EMAIL PROTECTED]
> >[mailto:[EMAIL PROTECTED] On Behalf Of Steven P.
> >Donegan
> >Sent: Friday, February 04, 2005 8:26 PM
> >To: asterisk-users@lists.digium.com
> >Subject: [Asterisk-Users] Encrypted VOIP?
> >
> >Is there any support in Asterisk for encryption of IAX and/or any other 
> >VOIP protocols? I haven't seen anything on this in the wiki or on the 
> >list. Just curious.
> >___
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-- 
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LANNet Computing Associates;
Your Linux people 
--
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Get rid of the Australian states."


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Re: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!

[Ben Hencke]

Wouldn't it be much simpler and effective to just boot them off the
list? I think they would get the picture pretty quick when they got
back...
They can always check the archives to read up on missed posts, and it
would save us all the trouble in the mean time ;-)

- Ben
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Re: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!

[Leif Madsen]

On Sat, 5 Feb 2005 00:07:59 -0500, dean collins <[EMAIL PROTECTED]> wrote:
> Lol - I already did do that 2 days ago and the 'associate' nominated as
> his replacement in the email.
> 
> Nothing has been done.
> 
> I figure seeing he is away until the 13th that we will be getting these
> messages for the next week and a half.

This was one of the things the mailing list moderators could take care
of, which I suggested in my post, "RFC: Moderating the Asterisk
Mailing lists". You can find posts regarding it here:

http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg08015.html
http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg08074.html

Thanks,
Leif Madsen.
http://www.leifmadsen.com
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[Asterisk-Users] ASTCC error on free calls

[Trevor G. Hammonds]

I set up certain 
routes in my ASTCC application to be free of charge.  When a user attempts 
to dial one of these numbers, the announcement plays the prompts "This 
call will cost", "nothing", and then terminates the script, dropping the 
call, leaving the card locked in the database as being in use.  

 
Any 
ideas?
 
    
Sincerely,
    Trevor 
Hammonds
 
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RE: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!

[dean collins]

Lol - I already did do that 2 days ago and the 'associate' nominated as
his replacement in the email.

Nothing has been done.

I figure seeing he is away until the 13th that we will be getting these
messages for the next week and a half.

Dean

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luki
Sent: Friday, February 04, 2005 11:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!

> Why don't you setup a filter rule like...
That's not the point. He isn't the only one getting them, so why
should EVERYONE add filters to cover all possible auto reply messages?
Just have the SENDER fix their end once for everyone.

--Luki
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Re: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!

[Steven Critchfield]

On Fri, 2005-02-04 at 20:34 -0800, Luki wrote:
> > Why don't you setup a filter rule like...
> That's not the point. He isn't the only one getting them, so why
> should EVERYONE add filters to cover all possible auto reply messages?
> Just have the SENDER fix their end once for everyone.

Because there are more moronic idiots than there are worthy people. I
submit the number of windows users here who are contributing to my spam
and virus folders as examples of moronic users.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!

[Luki]

> Why don't you setup a filter rule like...
That's not the point. He isn't the only one getting them, so why
should EVERYONE add filters to cover all possible auto reply messages?
Just have the SENDER fix their end once for everyone.

--Luki
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Re: [Asterisk-Users] MYSQL Failed

[Luki]

> How can I debug what this problem?
Make sure you have the correct user name and password, that the host
name and database is correct... and that the mySQL server is running
at the speciefied host and is reachable (if not localhost).

--Luki
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RE: [Asterisk-Users] astcc digit timeout

[Trevor G. Hammonds]

Karl H. Putz wrote on Thursday, 3 February 2005 12:14 PM:

> modify the agi to add the specific inter-digit timeout in
> milliseconds you would like after the prompt filename in the get_data
> calls.  
> 
> i.e. use:  $cardno = $AGI->get_data("astcc-accountnum",5000);
> 
> if you want a 5 second allowable delay between digits.

Karl,
This is a great suggestion!  I think it should be added to the ASTCC
distribution.  

Sincerely,
Trevor Hammonds

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[Asterisk-Users] Need some Advise

[Stephen Liew]

Hi All,
I am new to Asterisk and currently have one potential implementation.
It is to setup Intercom system using Asterisk for 1500 household.
They can call to each other using extension and can call out via PRI.
Can someone share your experience in implementing such a big systems 
(server sizing, uptime, fail over planning, cost involve)?

--
Thanks and Regards,
Stephen Liew,

begin:vcard
fn:Stephen Liew
n:Liew;Stephen
adr:Tmn Melodies;;14A, Jln Geronggang;Johor Bahru;Johor;80250;Malaysia
email;internet:[EMAIL PROTECTED]
tel;work:+(60) 7 334 9781
tel;fax:+(60) 7 334 5502
tel;cell:+(60)-12-7107350
url:http://www.revoltel.com
version:2.1
end:vcard

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[Asterisk-Users] MYSQL Failed

[Hariharan Gopalan]

Wondering if someone could please help me with this error:

Feb  4 22:10:58 WARNING[18502]: app_addon_sql_mysql.c:235 aMYSQL_connect: mysql_
real_connect(mysql,localhost,root,dbpass,mydb,...) failed

How can I debug what this problem?

Thanks 
Hari
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Re: [Asterisk-Users] Encrypted VOIP?

[Steven P. Donegan]

Well, not really 'looking for' anything except info - as an earlier 
gentleman said I can always just do stuff over a VPN - it's more a 
simple interest in where things might be going.

Thanks as always for the feedback folks
Nick Bachmann wrote:
[EMAIL PROTECTED] wrote:
If I remeber correctly, Mark Spencer is working on encryption in IAX2
 

Sort of.  Some IAX encryption code went into CVS a while back, but it 
was more of a "talking point" than anything else, meant to give 
interested developers a starting point.  The -dev and -security list 
archives have some discussion about this.  I don't think there's been 
too much continued work on it, but volunteers are always accepted!

For SIP, native encryption should be done through SRTP, which many 
people have asked for but nobody has really delivered.  Again, you 
will find some good background discussion in the list archives.

Running IAX over stunnel would probably be feasible if both sides of a 
tunnel were machines. That's at least a little closer to to native, 
and very easy to set up.  However, I don't think that's what you were 
looking for... :)

Nick
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RE: [Asterisk-Users] Callerid problems with 1.0.5

[mattf]

Hello,

patching v1.0.5 on my system removed the problem for me. But yes it seems
strange that this feature was inserted into a final release with very little
documentation of the wide implications that are caused by the change.

This was corrected in CVS with the addition of a diabling flag for the dial
command, but maybe this is a message that we should start an official beta
release period before a release so that people can test pre-releases even
for just a week to report problems before it is unleashed upon the world as
an official release

MATT---


-Original Message-
From: Mark Eissler [mailto:[EMAIL PROTECTED]
Sent: Friday, February 04, 2005 9:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Callerid problems with 1.0.5


Yikes!

On Feb 4, 2005, at 1:26 PM, Jay Milk wrote:

> Can someone clarify what's going on here?
>
> I'm running 1.0.5, and I see caller-id come through just fine from one
> extension to the other, as well as for incoming and outgoing calls
> (iax2).  What are you folks seeing there?
>

The behavior that was reported by Kevin is/was exactly the same 
behavior that I was experiencing with 1.0.5 and reported in another 
thread. I switched back to 1.0.2 to resolve that problem and another I 
was experiencing (SIP calls ringing forever instead of disconnecting 
even when voicemail had already picked up).

Reading through the bug tracker on this one I must say I'm a bit 
confused. I understand the concept of showing useful/relevant callerid 
when a call is transferred (from park or some other extension) but I 
don't understand why a call should ever show the recipient extension's 
callerid. My understanding is that this is the default behavior when no 
other callerid is present and for some reason inbound callerid is 
getting wiped out because it's not correct.

That some people are experiencing problems with this while others are 
not leads me to believe that it might be a problem that is exacerbated 
depending upon the dialplan setup. I'm just thinking this at the top of 
my head now, haven't looked back at my dialplan yet.

What's annoying, either way, is that when this change was made the 
behavior of existing, functioning setups broke. I don't recall seeing 
any documentation for 1.0.5 that noted this might be the case and if 
the documentation is lacking...well, that's a problem.

-mark

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

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RE: [Asterisk-Users] vicidial and mysql ........help

[mattf]

Hello,

First, there is a mailing list for the astGUIclient suite:
https://lists.sourceforge.net/lists/listinfo/astguiclient-users

As for your problem, If you have everything set up correctly you should just
be able to run the AST_VDhopper.pl script from your Asterisk server to fill
your lead hopper. Make sure you have the dialing restrictions in the
campaign screen set to 24 hours for testing and the dial level set to 1 or
higher and the hopper level to 1 or greater.

If you are still having problems, send the output of the AST_VDhopper script
to the astguiclient-users list and you'll get some help.

MATT---

-Original Message-
From: Hussain Umair [mailto:[EMAIL PROTECTED]
Sent: Friday, February 04, 2005 4:45 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] vicidial and mysql help


hi all.well ive installed each and everything according to the scratch 
installation but the problem is when i try to login a user through 
vicidialgui application it gives an error that there are 0 leads in the 
hopper to dialwell im pasting the result of a few queries .plz if 
any one can do help me out 
this is what the error looks like...

SELECT count(*) FROM vicidial_hopper where campaign_id = '13' and 
status='READY';
0 - leads left to call in hopper
so the login fails..


but these queries at the mysql cmd line tell me i got 7 leads in the 
database

select * from vicidial_list;
7 rows in set (0.00 sec)


but when i use count it gives this
mysql>select count(1) from vicidial_hopper;
+--+
| count(1) |
+--+
|0 |
+--+
1 row in set (0.00 sec)

mysql> select status,count(1) from vicidial_hopper group by status;
Empty set (0.00 sec)

mysql> select campaign_id from vicidial_hopper group by campaign_id;
Empty set (0.00 sec)


please any ideas or help woud be greatly appreciatedim kinda lost in 
this mysql lala land cant figure out whats going on.help me out here 
guys thankss

kurt...
Network Engineer plus Asterisk Newbie u can say

_
Express yourself instantly with MSN Messenger! Download today it's FREE! 
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Re: [Asterisk-Users] Callerid problems with 1.0.5

[Mark Eissler]

Yikes!
On Feb 4, 2005, at 1:26 PM, Jay Milk wrote:
Can someone clarify what's going on here?
I'm running 1.0.5, and I see caller-id come through just fine from one
extension to the other, as well as for incoming and outgoing calls
(iax2).  What are you folks seeing there?
The behavior that was reported by Kevin is/was exactly the same 
behavior that I was experiencing with 1.0.5 and reported in another 
thread. I switched back to 1.0.2 to resolve that problem and another I 
was experiencing (SIP calls ringing forever instead of disconnecting 
even when voicemail had already picked up).

Reading through the bug tracker on this one I must say I'm a bit 
confused. I understand the concept of showing useful/relevant callerid 
when a call is transferred (from park or some other extension) but I 
don't understand why a call should ever show the recipient extension's 
callerid. My understanding is that this is the default behavior when no 
other callerid is present and for some reason inbound callerid is 
getting wiped out because it's not correct.

That some people are experiencing problems with this while others are 
not leads me to believe that it might be a problem that is exacerbated 
depending upon the dialplan setup. I'm just thinking this at the top of 
my head now, haven't looked back at my dialplan yet.

What's annoying, either way, is that when this change was made the 
behavior of existing, functioning setups broke. I don't recall seeing 
any documentation for 1.0.5 that noted this might be the case and if 
the documentation is lacking...well, that's a problem.

-mark
--
Mark Eissler, [EMAIL PROTECTED]
Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com
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Re: [Asterisk-Users] Encrypted VOIP?

[Nick Bachmann]

[EMAIL PROTECTED] wrote:
If I remeber correctly, Mark Spencer is working on encryption in IAX2
 

Sort of.  Some IAX encryption code went into CVS a while back, but it 
was more of a "talking point" than anything else, meant to give 
interested developers a starting point.  The -dev and -security list 
archives have some discussion about this.  I don't think there's been 
too much continued work on it, but volunteers are always accepted!

For SIP, native encryption should be done through SRTP, which many 
people have asked for but nobody has really delivered.  Again, you will 
find some good background discussion in the list archives.

Running IAX over stunnel would probably be feasible if both sides of a 
tunnel were machines. That's at least a little closer to to native, and 
very easy to set up.  However, I don't think that's what you were 
looking for... :)

Nick
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[Asterisk-Users] RE: Same Extensions in Multiple contexts

[Jason Brown]

Revisiting this: I was able to get this to work as well, but voicemail doesn’t work as intended. A user in companya-internal cannot get his voicemail when in the office. It gives login incorrect. However if the user dials in from outside through a zap channel the vm login works. Any ideas? Jason  I was able to get this to work.  Not sure if it is the best way or the onlyway, but this is how I did it.  Including contexts do not give you thedesired result.  You must keep the contexts separate and use the goto to getit to work. Use the internal context in sip.conf.  If you don't people calling into thesystem will be allowed to dial an out bound line as if it were an extension. You also must take care with voicemail. Notice the @context, if the @ symbolis missing it will match any context with that extension.  You will get veryunpredictable results. Call Parking does not appear to work.  It does not support contexts.  Extensions.conf [incoming-calls]exten => _2125551212,1,Goto,companya|${EXTEN}|1exten => _2025551212,1,Goto,companyb|${EXTEN}|1  [companya]exten => _2125551212,1,Macro(auto-attendant) exten => _202,1,Dial(device)exten => _203,1,Dial(device) exten => _8500,1,VoicemailMain(@companya) [companya-internal]include => companyainclude => outgoing [companyb]exten => _2025551212,1,Macro(auto-attendant) exten => _202,1,Dial(device)exten => _203,1,Dial(device) exten => _8500,1,VoicemailMain(@companyb)  [companyb-internal]include => companybinclude => outgoing  sip.conf[phone1]context=companya-internal   -- Message: 9Date: Mon, 8 Nov 2004 15:43:10 -0500From: "Uma S. Pandey" Subject: [Asterisk-Users] Same Extensions in Multiple contextsTo: Message-ID: <200411081433493.SM02180 at UMA>Content-Type: text/plain; charset="us-ascii" Hi   For a customer, I am trying to setup 3 different companies on one asteriskbox, and I need to assign extension 200 in three different companies. I wasusing different contexts, but was unable to get it to work. So, my basicquestion is -    In Asterisk, Can we have same extension number in different contexts?    For example:   [Context_company_1] exten => 200,1,,,  [context_company_2] Exten =>200,1,..  [context_company_3] Exten =>200,1,..   Thanks  Uma Pandey

 






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Re: [Asterisk-Users] Call pickup across technologies (SIP, IAX, MGCP)?

[Philipp von Klitzing]

Hi again!

> > it appears that call pick-up only works _within_ a technolgoy, i.e. with 
> > a SIP phone when another SIP phone is ringing. Is that correct, or is my 
> > configuration faulty?
> > 
> > * Case 2:
> > IAX phone ringing - SIP phone can't pick the call up:
> > NOTICE[10250]: Nothing to pick up
> 
> This seems less a matter of technology than a matter of implementation.
>  From the SIP phones, I can pickup ANY call, no matter if between ISDN, 
> SIP or cross-channel. From the ISDN phones, I can pickup NO calls 
> ("unknown extension *8 in context from_ISDN").

Hm... with the help of the bristuff PickUp() app I was able to solve this 
"unkown extension" for 2 of my 3 cases, but trying to pickup a ringing 
IAX phone with SIP still fails with error "no channel found 2" (bristuff 
"exten => *8,1,PickUp(1)"). All clients have callgroup=1 and 
pickupgroup=1.

If I do "ship show peer " I get:

  Callgroup: 1 (2)
  Pickupgroup  : 1 (2)

and I wonder what the (2) is supposed to mean in both cases, the 
errormessage as well as the peer info. Maybe there is a difference in 
implementation of callgroup= in iax.conf where one starts couting at 0 
and the other at 1?

Hm... too bad there is no "iax2 show peer "...

Philipp


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Re: [Asterisk-Users] Encrypted VOIP?

[timebandit001]

If I remeber correctly, Mark Spencer is working on encryption in IAX2

hth
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Re: [Asterisk-Users] Encrypted VOIP?

[Steven P. Donegan]

I have done that extensively (H.323 and SIP over IPSEC tunnels) I was 
more interested in the possibilities of 'native' support of some kind. 
But thank you very much for the response.

dean collins wrote:
Just run point to point encryption over a vpn.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven P.
Donegan
Sent: Friday, February 04, 2005 8:26 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Encrypted VOIP?
Is there any support in Asterisk for encryption of IAX and/or any other 
VOIP protocols? I haven't seen anything on this in the wiki or on the 
list. Just curious.
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RE: [Asterisk-Users] Encrypted VOIP?

[dean collins]

Just run point to point encryption over a vpn.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven P.
Donegan
Sent: Friday, February 04, 2005 8:26 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Encrypted VOIP?

Is there any support in Asterisk for encryption of IAX and/or any other 
VOIP protocols? I haven't seen anything on this in the wiki or on the 
list. Just curious.
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[Asterisk-Users] Encrypted VOIP?

[Steven P. Donegan]

Is there any support in Asterisk for encryption of IAX and/or any other 
VOIP protocols? I haven't seen anything on this in the wiki or on the 
list. Just curious.
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[Asterisk-Users] Re: Can't get Polycom auto-answer to work (Solved)

[Noah Miller]

So I guess the problem is in my config for the phone?  Or maybe  
asterisk
has to send "alert-info" more than just once?  Does anybody have this
auto-answer config working reliably on a Polycom phone?

Thanks!
Noah
Noah,
	Please see my Polycom config files at
http://www.kriscompanies.com/modules.php? 
name=Downloads&d_op=viewdownload&cid=1

I have a setup for a Ring-answer and autoanswer.
Thanks for the configs Kristian!  I'm learning a lot from them.
Kevin, thanks for the debug tips and the heads up on the alert-info  
variable (or rather _ALERT_INFO) in CVS HEAD.  I also recompiled with  
Alert-info changed to Alert-Info.  That may have helped, too.

In the end, the holdup was my bad configs on the phone.  I had been  
messing around with them trying to make the phone ring on simultaneous  
calls with SetGroup/CheckGroup.  I thought I had changed everything  
back, but I guess I hadn't.   I restored the default configs, and the  
auto-answer worked just fine.  D'uh.

Thanks!
Noah
BTW:  Has anybody figured out how to make the phones ring when a second  
or third simultaneous call comes in when using SetGroup/CheckGroup?

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Re: [Asterisk-Users] how to add more TDM04B

[Lyle Giese]

Take a look at /etc/zaptel.conf

You need to modify this file.


- Original Message - 
From: "Martin Roy" <[EMAIL PROTECTED]>
To: 
Sent: Wednesday, February 02, 2005 1:46 PM
Subject: [Asterisk-Users] how to add more TDM04B


> I already have one Digium TDM04B installed in my server working fine. I 
> just received 2 more so I did added them to my server. When I booted 
> again linux told me that it found new hardware I said ignore. Then I log 
> in as usual. I did ztcfg -vv to see if it sees 12 channels now instead 
> of 4 but it only see 4. So then I did again modprobe zaptel, then 
> modprobe wcfxo and wcfxs hoping it would see the new cards but nothing 
> happen... it still see the 4 channels only.. What I have to do to make 
> it see the new card? Asterisk is not running but I know zaptel load at 
> startup is it the problem?
> 
> Thanks
> 
> Martin
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Re: [Asterisk-Users] AU caller ID with Sipura SPA-3000

[Howard Lowndes]

On Sat, 2005-02-05 at 08:28, Eric Bishop wrote:
> Hi All,
> 
> I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and
> out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN
> Line" tab) so Asterisk answers the call rather than the SPA-3000. It
> is all working perfectly except I can't get the SPA-3000 to pass
> caller ID to Asterisk. It passes "Display Name", "User ID" and any
> "PSTN CID Number Prefix" I have configured.
> 
> I have adjusted "PSTN Ring Thru Delay" to 10 as I realise caller ID is
> not presented until the second ring in oz. I have also verified that
> caller ID is enabled on the line (with an analogue LCD handset).
> 
> Has any aussie out there had success getting the SPA-3000 to pass
> caller ID to Asterisk?

Not specifically the Sipura, but check that your circuit has caller id
presentation enabled (it is off by default).  Check the wiki about
Australian Caller ID, I posted there the other day.

> 
> The only settings I havn't played with yet is "Caller ID Method" (on
> the Regional tab). It is set to the default of "Bellcore (N.Amer,
> China)" which I beleive is correct.
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-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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Re: [Asterisk-Users] RE:Terrible inbound call quality vs. outbound

[Brian Dingman]

I have contacted VP regarding this issue and have included links to
this thread. My ticket number is [Incident: 050120-92] for
reference. Might want to fire off an email referencing it

On Fri, 4 Feb 2005 18:27:18 -0500, Daryl G. Jurbala
<[EMAIL PROTECTED]> wrote:
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Gene Willingham
> > Sent: Tuesday, February 01, 2005 6:49 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] RE:Terrible inbound call quality
> > vs. outbound
> >
> >
> >
> > I am experiencing the same problem, except I do not use
> > Voicepulse outbound.
> > I have 100 Mbps connection, so it should not be a bandwidth
> > issue.   Last
> > Thursday they had a 4 hour outage on inbound calls.  The call
> > quality has deteriorated since.  I am in the process of
> > looking for another provider.
> [...]
> 
> Not to just "me too", butme too.  I've contacted their support on
> numerous occasions, and have been given busywork to do (run ping plotter
> for 24 hours, send us the results, etc) and never receive a response
> that acknowledges a problem of any sort.
> 
> Daryl
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RE: [Asterisk-Users] Limit MOH processes

[Chamberland-Larose, Guillaume]

> 
> > - use one of the many patches for native MOH without mpg123
> 
> Well, doesn't that mean, I have to convert all the mp3s to 
> another bitrate/format? *sigh*
> 

I'm using a whole bunch of my mp3s and I didn't have to convert any of
them. It seems to be working fine. All I had to do is install the addon.

Guills
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Re: [Asterisk-Users] Nortel i2004 support asterisk?

[Leo Ann Boon]

You were able to complete calls from one phone to another?
The installation doesn't look that difficult.
 

Yes. from one i2004 to another i2004 works. Most of the time was spent 
figuring out what goes into the sql tables.

It looks like it was a lot of work to reverse engineer it.
 

Yup, totally agree with you. The protocol is pure binary. I was told 
might be a variant of Megaco/H.248.
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[Asterisk-Users] codec0 = 516 is not codec1 = 216

[Matthew Boehm]

I got this today when using XTen Pro. What does it mean? What is codec 216?

-Matthew
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[Asterisk-Users] Answering machine detection

[Ben Hencke]

Hi all,
I would like to have basic answering machine detection for a
notification service (not spam, not telemarketing) so that I know when
I get a machine, and also to avoid starting the message during the
machine greeting.

I have found the WaitForSilence app in CVS, and this seems like it
would do exactly what I need, but does not seem to work in my setup.

I am making the call over VoIP (tryed AIX and SIP with GSM), but keep
getting errors where it calls app_waitfor(). I get these messages on
the console:

"One waitfor failed. "
"No audio available on ??"


If I use IAX, the progress will wait for some silence, but still
display the error after it gets some silence. If I use SIP, it will
never wait more than a few seconds and doesn't seem to wait for any
silence at all.

Does anyone know of some other ways (ie applications)  to detect
answering machines?

Thanks in advance,
  Ben
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Re: [Asterisk-Users] Digium X100P FXO Asterisk for Australia ?

[Gary]

On Sat, 5 Feb 2005 08:56:34 +1000, Peter Illmayer wrote:

>Anyone in Australia using these cards ?  I've been using a sipura and had to
>do a lot of research to get the thing to detect hangup and other subtilties on
>the Telstra network.
>
>These boards work OK in Australia ?

They can be made to work quite ok, but they have their own
peculiarities.

NB: These boards do not have an A-tick and thus cant be sold within
Australia
unless they are NO-ticked and thus are not legal to attach to the PSTN.

Now having said that, I have used them extensiively as PABX extenders
linking multiple analogue PABX's and Key systems with any dramas.

Gary

.


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[Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!

[Daryl G. Jurbala]

PLEASE CONFIGURE YOUR 
AUTORESPONSERS TO NOT SEND MESSAGES TO PEOPLE WHO POST IN MAILING LISTS YOU 
SUBSCRIBE TO.
 
This is an extremely rude 
thing to allow, and is becoming increasingly common, especially with users of 
the Asterisk-Users list.
 
Daryl

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] Sent: Friday, February 04, 
  2005 6:41 PMTo: Daryl G. JurbalaSubject: AUTOREPLY RE: 
  [Asterisk-Users] Zap channel occasionally misses di...
  
  Vielen Dank für Ihre Email!
   
  Ich bin vom 02.02.05 bis einschließlich 13.02.05 ausser Haus.
   
  Ihre Email wird bis dahin nicht bearbeitet oder weitergeleitet.
   
  Bei dringenden Fragen wenden Sie sich bitte an meinen Kollegen
   
  Herrn Rüdiger Hoog
  Email: [EMAIL PROTECTED]
  Telefon: 02331/473101-11Telefax: 02331/473101-19
  
  
  Für weitere Fragen stehe ich Ihnen gerne zur Verfügung und 
  verbleibe
   
  mit freundlichem Gruß,
   
  Stefan SpeckenheuerTechnische Leitung
   
  POS Service, Logistik & Handels GmbHAuf dem Graskamp 
  2D-58099 Hagen
   
  Tel. +49 2331 473101-21FAX  +49 2331 473101-39
   
  mailto:[EMAIL PROTECTED]http://www.posservice.de
   
  Sitz der Gesellschaft: Walter-Rathenau-Ring 9-11, 59581 Warstein 
  BeleckeHandelsregister Arnsberg: HRB 2958Ust.IdNr.: DE 198 933 
  818Geschäftsführer: Martin Menzel, Christian Woelke
   
  Diese E-Mail einschließlich aller Anhänge ist vertraulich.Wir 
  bitten, eine fehlgeleitete Mail unverzüglich vollständigzu löschen und uns 
  eine Nachricht zukommen zu lassen.Wir haben die Mail beim Ausgang auf 
  Viren geprüft;wir raten jedoch, auf Grund der Gefahr auf 
  denÜbertragungswegen, zu einer Eingangskontrolle.Eine Haftung für 
  Virenfreiheit schließen wir aus.
   
  This e-mail and any attachments are confidential.If you are not the 
  intended recipient of this e-mail,please immediately delete its contents 
  and notify us.This e-mail was checked for virus contamination 
  beforebeing sent; nevertheless, it is advisable to checkfor any 
  contamination occuring during transmission.We cannot accept any liability 
  for virus contamination.
  > -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
> Sent: Tuesday, February 01, 2005 11:26 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Zap channel occasionally misses 
> dialing thefirst digit
> 
> have a look at http://www.voip-info.org/wiki-Asterisk+zap+channels
> 
> Here it tells you that you can specify a wait period.
[...]

Don't know if it will apply to those having issues with BRI/PRI, but in
my case, a ww in front of the dial string has worked witout fail for the
last few days.

Thanks to all who helped,
Daryl
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RE: [Asterisk-Users] Zap channel occasionally misses dialing thefirst digit

[Daryl G. Jurbala]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
> Sent: Tuesday, February 01, 2005 11:26 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Zap channel occasionally misses 
> dialing thefirst digit
> 
> have a look at http://www.voip-info.org/wiki-Asterisk+zap+channels
> 
> Here it tells you that you can specify a wait period.
[...]

Don't know if it will apply to those having issues with BRI/PRI, but in
my case, a ww in front of the dial string has worked witout fail for the
last few days.

Thanks to all who helped,
Daryl
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RE: [Asterisk-Users] RE:Terrible inbound call quality vs. outbound

[Daryl G. Jurbala]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Gene Willingham
> Sent: Tuesday, February 01, 2005 6:49 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] RE:Terrible inbound call quality 
> vs. outbound 
> 
> 
> 
> I am experiencing the same problem, except I do not use 
> Voicepulse outbound.
> I have 100 Mbps connection, so it should not be a bandwidth 
> issue.   Last
> Thursday they had a 4 hour outage on inbound calls.  The call 
> quality has deteriorated since.  I am in the process of 
> looking for another provider.
[...]

Not to just "me too", butme too.  I've contacted their support on
numerous occasions, and have been given busywork to do (run ping plotter
for 24 hours, send us the results, etc) and never receive a response
that acknowledges a problem of any sort.

Daryl
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[Asterisk-Users] toll-free anonymous

[Andrew Thompson]

Hi, I'm Andrew.
(Hi Andrew)
I'm a toll-free number junkie.
I've had an account with iax.cc/sixtel for about a week, and every few 
days, I find myself sitting at the DID menu clicking the link that reads 
"Click here to get a random toll free number".

I have three toll-free numbers now, and I don't know if it will stop...
Is there any hope for me?
--
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
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[Asterisk-Users] Polycom Auto-Answer and Call Transfers

[Jared Armstrong]

I have my * and polycom system setup to do
Auto-Answer for internal SIP/Staff calls, and I am running into an issue with
this and the polycom call transfer feature. * is seeing a new call come through
from the polycom and is then transferring the call over. I need to know if
there is some way I can grab a message from the SIP header or something to
determine if I should not set the ALERT_INFO tag to A-A. I would greatly
appreciate it if someone could help me out with this, I need to have this
resolved by Monday.

 

Thanks,

 

Jared Armstrong






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RE: [Asterisk-Users] Bristuff and incoming call problems

[Stefan Reuter]

On Fri, 2005-02-04 at 23:50 +0100, Remco Barende wrote:
> I haven't tried that. Where did you find about this option? Are there any 
> docs about bristuff? The tarball doesn't give much info and the Junghanns 
> website even less.

maybe we should start some doc on the wiki on bri stuffs additional
features.
what comes to my mind is:

- +201 priority for dial
- 'R' option for dial
- PickUp/PickDown/Steal/PickupChan apps
- changes for unique id syntax (includes uniquename configured in
asterisk.conf and pid of asterisk process)
- chan_iax2 (added Hangup cause signalling implementation by Levent
Guendogdu <[EMAIL PROTECTED]>)
- chan_zap (cid handling, ...)
- res_features (autoanswer and autoanswerlogin)
- manager (new actions dbget, dbput, dbdel, changes on status action,
returning uniqueid on originate)
- pbx (hangup with cause parameter)

and maybe more that i didn't get yet.

any further ideas?

stefan

-- 
stefan reuter

[EMAIL PROTECTED], http://www.reucon.net, phone: 0700 88 REUTER
neusserstr. 110, d-50670 koeln, germany, fax: 0221 130569990

pgp key 1024D/9A1FA605 http://www.reucon.de/pgp.asc
2C89 AA8A 956F 5E7E A5F6   D397 4D46 7B5D 9A1F A605

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[Asterisk-Users] Digium X100P FXO Asterisk for Australia ?

[Peter Illmayer]

Anyone in Australia using these cards ?  I've been using a sipura and had to
do a lot of research to get the thing to detect hangup and other subtilties on
the Telstra network.

These boards work OK in Australia ?

Many Thanks Pete

--
Open WebMail Project (http://openwebmail.org)


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Re: [Asterisk-Users] Call forwarding

[Andrew Thompson]

Ryan Courtnage wrote:
One question - let's say someone specifies their home phone number and
their cell number.  How do you take into the account if the cell VM
picks up (ie. if cell is out of coverage and VM greeting is played)?

AFAIK, there isn't much you can do in this scenario - other than ringing
your house for a few rings before ringing your house AND the cell.  Even
then, the cell provider's 'out of the service area' message would answer
the call.
That's where you move on to building/adding the logic for: "Press 1 to
accept this call"
--
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
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Re: [Asterisk-Users] callback on busy

[Andrew Thompson]

Bartosz Jozwiak wrote:
Hello everybody,
I would like to implement "callback" function.
When I call a person and his extension is busy I can press, for example, 5
and get a callback when his phone is not busy anymore.
When I create a call file and copy it to spool call folder
asterisk makes a call. One problem is that when extension is still busy
my phone rings and I get busy tone of the person who I am trying to reach.
Is there another way to do it ? Or do I need to check first if channel 
is free or
still busy ? Can anybody give me some hints ?
Are you passing the Dial line in your .call file?
Try building a context that has your logic in it, and directing the call 
file to it, after setting some appropriate parameters.

You should be able to discern from Dial() why you can't get ahold of the 
other party. Once you do, test for that case as a part of your logic. If 
the case still exists, write a new .call file and exit. If the case no 
longer exists, connect to the original caller and be done. (Assuming you 
are not on the phone now, as well!)

Seems like I read that you could date a .call file a little bit of time 
in the future and asterisk would wait to run it until then. This would 
be a way for you to buy some time between tries. If I'm just making this 
up, use cron to straighten it out.

--
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
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RE: [Asterisk-Users] Bristuff and incoming call problems

[Remco Barende]

On Fri, 4 Feb 2005, Philipp von Klitzing wrote:
Hi!
The weird thing is that sometimes tyhe caller does hear the phone
ringing, and sometimes the line is "dead".
Question: Have you tried R instead of r in the dial command? This option
comes with bristuff as a patch:
'R' -- indicate ringing to the calling party when the called party
indicates
Maybe that solves it for you?
I haven't tried that. Where did you find about this option? Are there any 
docs about bristuff? The tarball doesn't give much info and the Junghanns 
website even less.

exten => s,1,Ringing
seems to work too
Thanks!
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[Asterisk-Users] No ring tone on Outgoing calls

[Carlos M]

Hi there i have some problems with some clients of my asterisk box, i
have some cases when a client tried to make a call and there is no
ring back only a silence and then the call hung up. I dont know why
this is happening. I have the following stable asterisk version:
CVS-v1-0-01/18/05-19:49:31

I did the an update a few days ago, the version that i had installed
before was: CVS-HEAD-10/08/04-12:44:50, I had some troubles with that
version related with IAX trunk=yes so i had to make the update.

Any help would be useful.

Thanks a lot for your help.

Carlos Andres Medina
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Re: [Asterisk-Users] Limit MOH processes

[Rich Adamson]

> is there a way to limit the number of spawned mpg123 processes. I have 
> 22 mpg123 in my tasklist, consuming 97% of CPU power. That seems a 
> little too much for me, considering, I have just 4 external lines...

Seems to have been a common occurance on past * implementations. Just
stop asterisk, then kill all mpg123 sessions that exist, and restart
asterisk.

Seems to me that I read something recently where coding changes were
made to help resolve this; could be just my imagination though.



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Re: [Asterisk-Users] External Callforward (Vanity CLI)

[Andrew Thompson]

magnus wrote:
Hello all, 
We have been asked if we can forward (for vanity reasons) one number to
another number whilst retaining the original callers Caller ID. For example
caller (ie 02027 xxx ) comes in on ISDN pri, and is then auto forwarded
to 0208 xxx  can the original caller id e.g. 02027 xxx  be presented
to remote site? 
We have tried a variety of options, but can not achieve this, checked the
wiki, but we are arriving at the conclusion that this is not possible unless
the carrier allows complete control on setting CLI, currently they only seem
to allow the CLI to be set as one of the DDI number on the PRI. Yet this can
be done when you divert on a GSM handset and if I remember correctly on my
old office definity pabx. Have I missed something? Can asterisk send Qsig
call divert information? Thanks for any and all thoughts - Magnus
Pick up an account with one of the many SIP/IAX providers that are 
asterisk compatible. They generally allow setting of the outbound CallerID.

I have personally tested that this feature works on connect.voicepulse.com.
I just tested iax.cc/sixtel a moment a go and it worked there as well.
Before you Dial, do a SetCallerID().
--
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
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RE: [Asterisk-Users] Limit MOH processes

[Chamberland-Larose, Guillaume]

You could try to use the native mp3 support for MOH if you really want
mp3 support. It is a lot better than using mpg123 IMHO. mpg123 kept
doing nasty things to my system :)

See
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musicon
hold.conf there is a section about the native support.

Guillaume 

> -Original Message-
> From: Stefan Gofferje [mailto:[EMAIL PROTECTED] 
> Sent: Friday, February 04, 2005 1:43 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Limit MOH processes
> 
> Hi folks,
> 
> is there a way to limit the number of spawned mpg123 processes. I have
> 22 mpg123 in my tasklist, consuming 97% of CPU power. That 
> seems a little too much for me, considering, I have just 4 
> external lines...
> 
> Regards,
>Stefan
> 
> 
> -- 
>   (o_   Stefan Gofferje  | Linux Systems Specialist
>   //\   Reg'd Linux User #247167 | SuSE Certified Linux Trainer
>   V_/_  Linux is like a Wigwam - No gates, no windows, Apache inside
> 
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Re: [Asterisk-Users] autoAnswer and autoAnswerLogin?

[Stefan Reuter]

hi,

> bristuff comes with these two applications - and too little info to 
> understand what they are for. Anyone has a clue and is willing to share 
> it?

i don't really know what they are good for but i had a look at the
source and think i know what they do.

imagine you have two extension 8767 and 8768 like

exten => 8767,1,Autoanswer
exten => 8768,1,AutoanswerLogin(8767)

when you dial 8768 you get are logged in to the autoanswer channel for
extension 8767. You will hear music on hold until someone calls 8767.
When someone calls 8767 this call is directly bridged (autoanswered) to
8768 so can immediately talk to him.

AutoanswerLogin takes an extension and optionally a context as arguments
(default context is "default").

If someone else logs in the first caller to 8768 is automatically logged
of.

hope that helps

stefan

P.S. You can look at the implementation of this apps in
res/res_config.c. search for autoanswer.

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Re: [Asterisk-Users] MeetMe & ztdummy

[Kristian Kielhofner]

Leif Madsen wrote:
On Fri, 04 Feb 2005 13:43:54 -0600, Kristian Kielhofner <[EMAIL PROTECTED]> 
wrote:
Alright, one more question before I try this So not having a zaptel
interface, is this why MeetMe didn't build when I initially made Asterisk?
Now that I have not heard off.  Perhaps if you built Asterisk without
Zaptel installed that might happen.  But I don't even know if that is
possible...

From the Makefile in ./asterisk/apps/
APPS+=$(shell if [ -f /usr/include/linux/zaptel.h ]; then echo
"app_zapras.so app_meetme.so app_flash.so app_zapbarge.so
app_zapscan.s
o" ; fi)
The ZapRAS, MeetMe, Flash, ZapBarge and ZapScan applications are not
compiled and installed unless Zaptel has been installed.
Thanks,
Leif Madsen.
http://www.leifmadsen.com
Thanks for that!
--
Kristian Kielhofner
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Re: [Asterisk-Users] HP ProLiant server for Asterisk

[Dana Olson]

On Fri, 04 Feb 2005 15:53:13 -0600, Steven Critchfield
<[EMAIL PROTECTED]> wrote:
> On Fri, 2005-02-04 at 16:02 -0500, Dana Olson wrote:
> 
> > Out of curiosity, do any of you think that the ML350 would be a better
> > choice, with similar options? They're a lot cheaper too, and I haven't
> > found any negative reports (no solid positive reports either) yet. The
> > Debian install will be a bit annoying, but that's alright, as long as
> > it's do-able.
> 
> I'm curious as to why you think debian wouldn't be anything but straight
> forward to install o that machine. I didn't see any hardware on there
> that shouldn't be super easy to get working. Also Debian installers have
> come a ways in the last series. I was disappointed to not have at least
> a little hiccup on a brand new machine I installed recently.
> 
> --
> Steven Critchfield <[EMAIL PROTECTED]>


Hey Steven,

I'm talking about Debian Stable, if that makes a difference. The
particular disc that I want to standardise on includes the 2.4.18bf24
kernel, and apparently the onboard ethernet is not supported by this
kernel. Also, if we're using the 64x SmartArray controller, this is
also not supported by the default kernel. I don't believe these
options showed up until 2.4.21 or so.

This means that I'd have to install to a SCSI drive off of the onboard
SCSI first, and then cp the data to the SmartArray after I've upgraded
the kernel, and then modify LILO and /etc/fstab to point to the new
devices.

I've never liked Debian Testing for regular use, and Sid is great on
my home desktop and laptop, but even on my own home server I run
Woody.
--
Dana
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Re: [Asterisk-Users] HP ProLiant server for Asterisk

[Steven Critchfield]

On Fri, 2005-02-04 at 16:02 -0500, Dana Olson wrote:

> Out of curiosity, do any of you think that the ML350 would be a better
> choice, with similar options? They're a lot cheaper too, and I haven't
> found any negative reports (no solid positive reports either) yet. The
> Debian install will be a bit annoying, but that's alright, as long as
> it's do-able.

I'm curious as to why you think debian wouldn't be anything but straight
forward to install o that machine. I didn't see any hardware on there
that shouldn't be super easy to get working. Also Debian installers have
come a ways in the last series. I was disappointed to not have at least
a little hiccup on a brand new machine I installed recently.

-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] vicidial and mysql ........help

[Hussain Umair]

hi all.well ive installed each and everything according to the scratch 
installation but the problem is when i try to login a user through 
vicidialgui application it gives an error that there are 0 leads in the 
hopper to dialwell im pasting the result of a few queries .plz if 
any one can do help me out 
this is what the error looks like...

SELECT count(*) FROM vicidial_hopper where campaign_id = '13' and 
status='READY';
0 - leads left to call in hopper
so the login fails..

but these queries at the mysql cmd line tell me i got 7 leads in the 
database

select * from vicidial_list;
7 rows in set (0.00 sec)
but when i use count it gives this
mysql>select count(1) from vicidial_hopper;
+--+
| count(1) |
+--+
|0 |
+--+
1 row in set (0.00 sec)
mysql> select status,count(1) from vicidial_hopper group by status;
Empty set (0.00 sec)
mysql> select campaign_id from vicidial_hopper group by campaign_id;
Empty set (0.00 sec)
please any ideas or help woud be greatly appreciatedim kinda lost in 
this mysql lala land cant figure out whats going on.help me out here 
guys thankss

kurt...
Network Engineer plus Asterisk Newbie u can say
_
Express yourself instantly with MSN Messenger! Download today it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[Asterisk-Users] External Callforward (Vanity CLI)

[magnus]

Hello all, 
We have been asked if we can forward (for vanity reasons) one number to
another number whilst retaining the original callers Caller ID. For example
caller (ie 02027 xxx ) comes in on ISDN pri, and is then auto forwarded
to 0208 xxx  can the original caller id e.g. 02027 xxx  be presented
to remote site? 
We have tried a variety of options, but can not achieve this, checked the
wiki, but we are arriving at the conclusion that this is not possible unless
the carrier allows complete control on setting CLI, currently they only seem
to allow the CLI to be set as one of the DDI number on the PRI. Yet this can
be done when you divert on a GSM handset and if I remember correctly on my
old office definity pabx. Have I missed something? Can asterisk send Qsig
call divert information? Thanks for any and all thoughts - Magnus

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[Asterisk-Users] AU caller ID with Sipura SPA-3000

[Eric Bishop]

Hi All,

I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and
out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN
Line" tab) so Asterisk answers the call rather than the SPA-3000. It
is all working perfectly except I can't get the SPA-3000 to pass
caller ID to Asterisk. It passes "Display Name", "User ID" and any
"PSTN CID Number Prefix" I have configured.

I have adjusted "PSTN Ring Thru Delay" to 10 as I realise caller ID is
not presented until the second ring in oz. I have also verified that
caller ID is enabled on the line (with an analogue LCD handset).

Has any aussie out there had success getting the SPA-3000 to pass
caller ID to Asterisk?

The only settings I havn't played with yet is "Caller ID Method" (on
the Regional tab). It is set to the default of "Bellcore (N.Amer,
China)" which I beleive is correct.
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RE: [Asterisk-Users] Call forwarding

[Adam Robins]

Here is the dial plan. Uses CVS-HEAD features for call screening.  This
is a pre-production system.  We will be replacing our existing Comdial
within the next few weeks.


;=
; extensions.conf file
;=
; Office extensions are 10-59
; Directory-by-name is *
; Parking extensions are 70-75
; Voicemail extension is 77 or #
; Music-on-hold test is 76
; MeetMe conference rooms are 81,82,83
; Polycom conference phone is 61
; Call Forward is 91
; DISA is 99
;

[general]
static=yes
writeprotect=no

[globals]
ZAPPHONE = Zap/1
ZAPLINE  = Zap/4
IAXNET = IAX2/iaxtel_out
;
; Macro for personal office extensions
;
[macro-ext]
exten => s,1,GotoIf($[${LEN(${INCOMING})} > 0]?2:211)  ;
If call is from outside, execute screening
exten => s,2,System(test -e
/var/spool/asterisk/voicemail/default/${ARG1}/greet.gsm)   ; check if
recorded name exists
exten =>
s,3,Playback(/var/spool/asterisk/voicemail/default/${ARG1}/greet)
; play recorded name
exten => s,4,Goto(s,104)   ;
skip playing number if played name
exten => s,103,SayNumber(${ARG1},f);
else play extension number
exten => s,104,Wait(1)
exten => s,105,Playback(screen-record) ;
ask caller to id self
exten => s,106,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH});
set file name
exten => s,107,Record(${SCREEN_FILE}.gsm|3|10) ;
record response to file
exten => s,108,DBget(fn=CFIM/${ARG1})  ;
is number forwarded?
exten =>
s,109,Dial(Local/[EMAIL PROTECTED]/n|24|gM(screen^${SCREEN_FILE})) ;
forwarded - screen to new number
exten => s,110,Goto(s,313)
exten => s,209,Dial(SIP/${ARG1}|24|tgM(screen^${SCREEN_FILE})) ;
not forwarded - screen to extension
exten => s,210,Goto(s,313)

; call is from inside - no screening
exten => s,211,DBget(fn=CFIM/${ARG1})  ;
is number forwarded?
exten => s,212,Dial(Local/[EMAIL PROTECTED]/n|24)  ;
forwarded, so send forwarding number back in
exten => s,213,Goto(s,313)
exten => s,312,Dial(SIP/${ARG1}|24|t)  ;
not forwarded, normal dial

exten => s,313,Voicemail(u${ARG1}) ;
voicemail if unavailable or busy
exten => s,314,Hangup

;
; Macro for call screening
;
[macro-screen]
exten => s,1,Wait(1)
exten => s,2,Playback(screen-from)
exten => s,3,Playback(${ARG1})
exten => s,4,Read(ACCEPT|screen-accept|1)
exten => s,5,GotoIf($[${ACCEPT} = 1 ] ?7:6)
exten => s,6,SetVar(MACRO_RESULT=CONTINUE)
exten => s,7,System(/bin/rm ${ARG1})

;
; Call forwarding Macro
;
[macro-forwarding]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,DigitTimeout(3)
exten => s,4,ResponseTimeout(10)
exten => s,5,Read(fwext,fw-extension,2); ask
extension (2 digits)
exten => s,6,Authenticate(/etc/asterisk/authFWD)   ; only
authorized individuals
exten => s,7,Playback(fw-extension); repeat back
extension
exten => s,8,SayNumber(${fwext},f)
exten => s,9,DBget(fwnum=CFIM/${fwext}); check if
already forwarded

; ext is forwarded
exten => s,10,Playback(fw-is-forwarded-to) ; play
forwarded number from database
exten => s,11,SayDigits(${fwnum})
exten => s,12,Read(resp,fw-cancel-1-change-2,1); 1 to cancel
fwd, 2 to change #
exten => s,13,GotoIf($[${resp} = 1]?17:14) ; 1 entered,
goto delete
exten => s,14,GotoIf($[${resp} = 2]?111:15); 2 entered,
jump to change number
exten => s,15,Playback(fw-invalid-response); invalid
response, loop back
exten => s,16,Goto(s,12)
exten => s,17,DBdel(CFIM/${fwext}) ; delete
entry from database
exten => s,18,Playback(fw-call-fwd-canceled)   ; give status
& end call
exten => s,19,Playback(fw-goodbye)
exten => s,20,Hangup

; ext is not forwarded
exten => s,110,Playback(fw-is-not-currently-forwarded) ; say number
is not forwarded
exten => s,111,Playback(fw-enter-new-forwarding-number); ask for new
number
exten => s,112,Read(fwnum,fw-press-pound-when-finished); accept new
number, since variable length, ask for #
exten => s,113,GotoIf($[${LEN(${fwnum})} < 2]?114:116) ; if len < 2
then bad number
exten => s,114,Playback(fw-invalid-response)
exten => s,115,Goto(s,111)
exten => s,116,Playback(fw-you-entered); repeat back
number
exten => s,117,SayDigits(${fwnum})
exten => s,118,Read(resp,fw-if-corr-press-1-otherwise-2,1) ; confirm 1
if correct, 2 if not
exten => s,119,GotoIf($[${resp} = 1]?120:111)  ; if 1,
proceed and update db, else loop back
exten => s,120,DBdel(CFIM/${fwext}); delete db
exten => s,121,DBput(CFIM/${fwext}=${fwnum})   ; add new db
entry
exten =>

[Asterisk-Users] Conference Bridge?

[Nash, Jason]

Hello,

Newbie needs some help J

I read on the list of features for Asterisk that it can work
as a Conference bridge.  Does anyone currently use this?  How well
does it work compared to like an AT&T conference bridge?

Thanks

Jason

 









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Re: [Asterisk-Users] Server Criteria

[Scott Laird]

On Feb 4, 2005, at 11:21 AM, Spencer Nassar wrote:
I've been doing a lot of background reading/searching of this list, 
voip-info.org, and Google, looking to define a good candidate for a 
server platform.  I'm very interested in thoughts from others!  So 
here goes...

Axiom 1:  if you are not doing doing much transcoding (converting 
between codecs), the bottleneck for supporting high volumes of 
simultaneous calls is system bus speed, not CPU power
---> points to a 64 bit AMD Opteron system, and maybe just one of the 
two processor slots populated.  Bus is twice as wide as a 32 bit 
system, and operates at 1.8GHz (a lot faster than a 64 bit Zeon 
system).  Then add the second processor to the board if you see you 
need it.
Er, not really.  First, the closest thing to a "system bus" in the 
Opteron system is really only 16 bits wide at its widest, and it 
probably only 8 bits wide by the point that it talks to your PCI 
bridge.  But, measuring the system's throughput on the basis of how 
many wires its internal plumbing uses is kind of nutty--the Opteron can 
move more data over an 8-bit Hypertransport link then a Pentium III 
could over its 64-bit bus.  And, in *either* case, the system bus isn't 
going to be a problem for any sort of telephony application.  I mean, 
even an 8-bit 800 MHz HT link is good for 12.8 Gbps each direction.  
That's a lot of T1s.

The real problem seems to be either CPU performance for transcoding or 
(more frequently) PCI bus bandwidth and latency.  Low-end systems with 
a single 32-bit PCI bus are going to have problems with more then one 
Digium card.  Systems with multiple busses *should* be able to scale 
further, but I haven't seen any sort of testing to back this up.

Axiom 2:  Get lots of memory
---> I haven't seen this quantified, and plan to do some testing.  
I'll post results here, but can anyone share any insights?  I'm 
planning to start at 2GB, and go up from there if I see swap getting 
used.
   - what would an alaw to alaw connection consume (if it didn't hand 
off)?
   - what about a 5 call alaw meetme bridge (and how much memory per 
incremental caller)
Again, not really.  Asterisk doesn't use a whole lot of RAM.  Make sure 
that it's not swapping, but even 256 MB is probably enough most of the 
time.

Axiom 3: Don't allow any disk IO
---> I'm assuming this is related to #2 - get lots of memory to avoid 
swap to disk.  Other issues or thoughts?
Well, if you have a really nasty IDE bus with DMA and interrupts 
disabled, then disk I/O could probably be a problem.  Other then that, 
it shouldn't matter.  As others have said, don't run a big DB server on 
the same box, but a bit of disk I/O isn't a problem.

Axoim 4: Come codecs will take advantage of the faster floating point 
of a 64 bit system
---> unknown... has anyone seen this?  Will Asterisk, compiled in a 64 
bit Linux environment, reap these or other benefits from being on a 64 
bit system (other than the system bus speed)?
Dunno.  I suspect that some codecs *could* benefit from 64-bit math, 
but I doubt that any of the current codecs are tuned for 64-bit CPUs.

Scott
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[Asterisk-Users] autoAnswer and autoAnswerLogin?

[Philipp von Klitzing]

Hi there,

bristuff comes with these two applications - and too little info to 
understand what they are for. Anyone has a clue and is willing to share 
it?

Thanks, Philipp


  -= Info about application 'Autoanswer' =-

[Synopsis]:
Autoanswer a call

[Description]:
Autoanswer(exten):Used to autoanswer a call for an extension.


  -= Info about application 'AutoanswerLogin' =-

[Synopsis]:
Log in for autoanswer

[Description]:
AutoanswerLogin(exten):Used to login to the autoanswer application for an 
extension.


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[Asterisk-Users] Call pickup across technologies (SIP, IAX, MGCP)?

[Philipp von Klitzing]

Hi there,

it appears that call pick-up only works _within_ a technolgoy, i.e. with 
a SIP phone when another SIP phone is ringing. Is that correct, or is my 
configuration faulty?


* Case 1:
SIP phone 1 ringing - SIP phone 2 can pick the call up with *8
We are happy! :-)

* Case 2:
IAX phone ringing - SIP phone can't pick the call up:
NOTICE[10250]: Nothing to pick up

* Case 3: 
SIP phone ringing - IAX phone can't pick the call up:
NOTICE[12300]: Rejected connect attempt from 192.168.x.y, reque
st '[EMAIL PROTECTED]' does not exist

The same applies to MGCP and SIP phone interaction.


[features.conf]
pickupexten = *8

[sip.conf and iax.conf]
callgroup=1
pickupgroup=2


Cheers, Philipp


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Re: [Asterisk-Users] MeetMe & ztdummy

[Leif Madsen]

On Fri, 04 Feb 2005 13:43:54 -0600, Kristian Kielhofner <[EMAIL PROTECTED]> 
wrote:
> > Alright, one more question before I try this So not having a zaptel
> > interface, is this why MeetMe didn't build when I initially made Asterisk?
> 
> Now that I have not heard off.  Perhaps if you built Asterisk without
> Zaptel installed that might happen.  But I don't even know if that is
> possible...

>From the Makefile in ./asterisk/apps/

APPS+=$(shell if [ -f /usr/include/linux/zaptel.h ]; then echo
"app_zapras.so app_meetme.so app_flash.so app_zapbarge.so
app_zapscan.s
o" ; fi)

The ZapRAS, MeetMe, Flash, ZapBarge and ZapScan applications are not
compiled and installed unless Zaptel has been installed.

Thanks,
Leif Madsen.
http://www.leifmadsen.com
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[Asterisk-Users] patch for chan_capi error condition report when receiving CAPI_CONF:CAPI_LISTEN message

[Luis Vazquez]

Hello,
here I submit a (previously mentioned) patch that fix a (false) error 
report when usin chan_capi.
I have checked this is not included in the patch at leviogo.de

Best regards
Luis
Pd: Just in case,if anybody need it I have made a version of 
chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2  compatible with the asterisk 
stable version at http://200.59.203.76/pub/chan_capi-0.3.5-patch.stable.diff

diff -ruN --ignore-all-space chan_capi-0.3.5/chan_capi.c chan_capi-0.3.5.ipcontact/chan_capi.c
--- chan_capi-0.3.5/chan_capi.c	2005-02-04 15:04:29.0 -0300
+++ chan_capi-0.3.5.ipcontact/chan_capi.c	2005-02-04 18:32:45.0 -0300
@@ -2161,6 +2161,18 @@
 		PLCI = INFO_CONF_PLCI(CMSG);
 //		ast_log(LOG_ERROR,"INFO_CONF PLCI=%#x INFO=%#x\n",PLCI,INFO_CONF_INFO(CMSG));
 	break;
+	case CAPI_LISTEN:
+	  if (LISTEN_CONF_INFO(CMSG)!=0) {
+		char * message = capi_info2str(LISTEN_CONF_INFO(CMSG));
+		if(!message) {
+		  asprintf(&message, "CAPI returned an unknown error! Please ask your manufacturer for assistance (error code=0x%X)\n", LISTEN_CONF_INFO(CMSG));
+		  ast_log(LOG_ERROR, message);
+		  free(message);
+		} else {
+		  ast_log(LOG_ERROR, "%s\n", message);
+		}
+	  }
+	  break;
 	case CAPI_CONNECT:
 		PLCI = CONNECT_CONF_PLCI(CMSG);
 		if (option_verbose > 3) {
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RE: [Asterisk-Users] Bristuff and incoming call problems

[Philipp von Klitzing]

Hi!

> The weird thing is that sometimes tyhe caller does hear the phone
> ringing, and sometimes the line is "dead".

Question: Have you tried R instead of r in the dial command? This option 
comes with bristuff as a patch:

'R' -- indicate ringing to the calling party when the called party 
indicates

Maybe that solves it for you?

Cheers, Philipp


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Re: [Asterisk-Users] HP ProLiant server for Asterisk

[Dana Olson]

On Fri, 4 Feb 2005 14:35:14 -0500, Dana Olson <[EMAIL PROTECTED]> wrote:
> On Fri, 4 Feb 2005 19:07:14 +0100 (CET), Christoph Rothe
> <[EMAIL PROTECTED]> wrote:
> > On Fri, 4 Feb 2005, Dana Olson wrote:
> >
> > > I'm looking at ordering a server from HP. I checked around on Google
> > > and found in the Wiki that the ProLiant DL380 is supposed to be known
> > > to work with *.
> >
> > > HP ProLiant DL380 G4 Server w/ the following options:
> > > Intel Xeon 3.20GHz/1MB
> > > 2GB REG PC2-3200 (2 X 1GB)
> > > HP ProLiant Battery Backed Write Cache Enabler for SA6i
> > > RAID 1 drive set
> > > HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive
> > > HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive
> > > Hot Plug Redundant Power Supply Module
> > > HP Redundant Fan Option Kit (3 fans)
> > > 1.44MB Floppy Disk Drive
> > > Slimline 24X CD-ROM
> >
> > Nice setup I think. The only thing with the DL380G3 and G4 is that it
> > freezes in some situations. This has been reported in the company
> > where I work and is a known bug to HP which will be solved by some
> > firmware upgrade.
> >
> > However the problem occurs only in some special situation noone can
> > really describe (cosmic rays ?! ;-) and at that location where I work
> > we installed SAP on a DL380G3 and it never froze.
> >
> > And at the end let me say that personally I like the DL380s. You can
> > even resize Logical Drives of the RAID (e.g. when putting in
> > some new harddisc) without shutting down the server!
> >
> > Good Luck!
> >
> > Christoph
> >
> 
> Thanks for all of the replies, both on and off list.
> 
> I should have also mentioned that I'll be installing Debian Stable on
> it. Are you using Debian or something else?
> 
> There seems to be an issue with the G4 and the Digium TE cards, so I'm
> unsure about placing the order at this time. I've been eyeing the
> Sangoma A104 card and the ipVolution TDM120 (not out yet, I believe)
> as an alternative.
> 
> I have a service contract with an HP provider, so I'd prefer to stick
> with an HP server, if at all possible. However, if the T1 interfaces
> cause issues, there may be no choice.
> --
> Dana




Out of curiosity, do any of you think that the ML350 would be a better
choice, with similar options? They're a lot cheaper too, and I haven't
found any negative reports (no solid positive reports either) yet. The
Debian install will be a bit annoying, but that's alright, as long as
it's do-able.
--
Dana
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[Asterisk-Users] DTMF Problem with analog phones

[Glenn McLelland]

Hi, I'm running a Voicetronix openswitch12 card 
under linux with asterisk. It's configured to have 8 loop start and 4 
station ports. I've got a few ariavoice and grandstream phones running 
off it without any problems. I've also got 4 analog phones running off 
it too. They work fine, except for a strange problem with the sending 
of dtmf tones once a call has been established. I've been calling our 
phone provider's voice mail service, and it seems to be very 
inconsistent in passing the correct dtmf key presses through. If I 
press the numbers at normal speed (2-3 per second) it's pretty much 
guaranteed not to work, however if I press each number approx 1 
second apart, it'll send the correct tones through. I've 
monitored the console with level 3 verbosity, and here's the 
output from entering the tones at normal speed, then at slow speed. 
I've dialled the voicemail number of 021 700700, and am trying to enter 
my mailbox number of 021 1271779#. The first time was unsuccessful, but 
the second time worked. Has anyone come across this problem 
before?  Here's the console 
log:    > vpb/1-5: 
handle_notowned: playing dialtone    -- Executing 
Dial("vpb/1-5", "vpb/g1/021700700") in new stack  ==  g1 
requested, got: [vpb/1-9]  == vpb/1-9: Calling 021700700 on 
vpb/1-9  == vpb/1-9: Dial parms for vpb/1-9 
1/2000ms/4000ms/4000ms/12ms  == vpb/1-9: Dial parms for vpb/1-9 
tone 7->0  == vpb/1-9: Dial parms for vpb/1-9 tone 0->1  
== vpb/1-9: Dial parms for vpb/1-9 tone 4->2  == vpb/1-9: Dial parms 
for vpb/1-9 tone 7->3  == vpb/1-9: Dial parms for vpb/1-9 tone 
3->4    -- vpb/1-9: VPB Calling 021700700 [t=12] on 
vpb/1-9 returned 0vpb/1-9: chanreads: starting thread    
-- Called g1/021700700    -- vpb/1-9 is ringing  == 
vpb/1-9: Dialend    -- vpb/1-9 answered vpb/1-5  == 
vpb/1-5: Answered call on vpb/1-5 [FXS]vpb/1-5: chanreads: starting 
thread  == vpb/1-5:Now listening for DTMF  == vpb/1-5: 
Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR]  == 
vpb/1-9:Now listening for DTMF  == vpb/1-9: Starting record mode 
(codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR]    -- Attempting 
native bridge of vpb/1-5 and vpb/1-9  == vpb_bridge: Bridging call 
entered with [vpb/1-5, vpb/1-9]  == Bridging call done with [vpb/1-5, 
vpb/1-9] => 0    -- Attempting native bridge of vpb/1-5 
and vpb/1-9  == vpb_bridge: Bridging call entered with [vpb/1-5, 
vpb/1-9]  == vpb/1-5: chanreads: Played DTMF 0  == Bridging 
call done with [vpb/1-5, vpb/1-9] => 0    -- Attempting 
native bridge of vpb/1-5 and vpb/1-9  == vpb_bridge: Bridging call 
entered with [vpb/1-5, vpb/1-9]  == vpb/1-5: chanreads: Not playing 
DTMF frame on native bridge  == Bridging call done with [vpb/1-5, 
vpb/1-9] => 0    -- Attempting native bridge of vpb/1-5 
and vpb/1-9  == vpb_bridge: Bridging call entered with [vpb/1-5, 
vpb/1-9]  == vpb/1-5: chanreads: Played DTMF 1  == Bridging 
call done with [vpb/1-5, vpb/1-9] => 0    -- Attempting 
native bridge of vpb/1-5 and vpb/1-9  == vpb_bridge: Bridging call 
entered with [vpb/1-5, vpb/1-9]  == vpb/1-5: chanreads: Not playing 
DTMF frame on native bridge  == Bridging call done with [vpb/1-5, 
vpb/1-9] => 0    -- Attempting native bridge of vpb/1-5 
and vpb/1-9  == vpb_bridge: Bridging call entered with [vpb/1-5, 
vpb/1-9]  == vpb/1-5: chanreads: Not playing DTMF frame on native 
bridge  == Bridging call done with [vpb/1-5, vpb/1-9] => 
0    -- Attempting native bridge of vpb/1-5 and 
vpb/1-9  == vpb_bridge: Bridging call entered with [vpb/1-5, 
vpb/1-9]  == vpb/1-5: chanreads: Not playing DTMF frame on native 
bridge  == Bridging call done with [vpb/1-5, vpb/1-9] => 
0    -- Attempting native bridge of vpb/1-5 and 
vpb/1-9  == vpb_bridge: Bridging call entered with [vpb/1-5, 
vpb/1-9]  == vpb/1-5: chanreads: Not playing DTMF frame on native 
bridge  == Bridging call done with [vpb/1-5, vpb/1-9] => 
0    -- Attempting native bridge of vpb/1-5 and 
vpb/1-9  == vpb_bridge: Bridging call entered with [vpb/1-5, 
vpb/1-9]  == vpb/1-9: chanreads: Played DTMF 0  == Bridging 
call done with [vpb/1-5, vpb/1-9] => 0    -- Attempting 
native bridge of vpb/1-5 and vpb/1-9  == vpb_bridge: Bridging call 
entered with [vpb/1-5, vpb/1-9]  == vpb/1-9: chanreads: Played DTMF 
2  == Bridging call done with [vpb/1-5, vpb/1-9] => 
0    -- Attempting native bridge of vpb/1-5 and 
vpb/1-9  == vpb_bridge: Bridging call entered with [vpb/1-5, 
vpb/1-9]  == vpb/1-5: chanreads: Played DTMF 1  == Bridging 
call done with [vpb/1-5, vpb/1-9] => 0    -- Attempting 
native bridge of vpb/1-5 and vpb/1-9  == vpb_bridge: Bridging call 
entered with [vpb/1-5, vpb/1-9]  == vpb/1-9: chanreads: Played DTMF 
1  == Bridging call done with [vpb/1-5, vpb/1-9] => 
0    -- Attempting native bridge of vpb/1-5 and 
vpb/1-9  == vpb_bridge: Bridging call entered with [vpb/1-5, 
vpb/1-9]  == vpb/1-5: chanreads: Played DTMF 2  == Bridging 
call done with [vpb/1-5, vpb/1-9] => 0    -- Attempting 
native bridge of vpb/1-5 and vpb/1-9  == vpb_bri

RE: [Asterisk-Users] SIP Challenge response bug?

[Keith Burns]

Did you get an answer on this ?

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matt Schulte
> Sent: Tuesday, February 01, 2005 6:56 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] SIP Challenge response bug?
> Importance: High
> 
> 
> Ok, here's an odd one. I would have opened a bug on this but last time
I
> tried that I got flamed.. :)
> 
> Problem: When proxy requests digest challenge (SIP) Asterisk responds
> normally with the exception that for some reason it changes the FROM:
> (Also changes Contact: )to what's in the original TO: line. Why on
earth
> is it doing this?! It must be a bug, I've gone over my extensions.conf
> several times to no avail. It seems older vers of ast don't do this.
> 
> near end is error:
> 
> Sip read:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK243afb2c
> From: "Matt S" ;tag=as5f22d23c
> To:
>
;tag=b27e1a1d33761e85846fc98f5f3a7e
> 58.2b61
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> Proxy-Authenticate: Digest realm="sipfarm.netlogic.net", nonce="blah
> blah", qop="auth"
> Server: Sip EXpress router (0.8.14 (i386/linux))
> Content-Length: 0
> Warning: 392 206.80.70.46:5060 "Noisy feedback tells:  pid=27326
> req_src_ip=206.80.66.138 req_src_port=5060
> in_uri=sip:[EMAIL PROTECTED]
> out_uri=sip:[EMAIL PROTECTED] via_cnt==1"
> 
> 
> 10 headers, 0 lines
> Transmitting:
> ACK sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK243afb2c
> From: "Matt S" ;tag=as5f22d23c
> To:
>
;tag=b27e1a1d33761e85846fc98f5f3a7e
> 58.2b61
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Content-Length: 0
> 
>  (no NAT) to 206.80.70.46:5060
> We're at 206.80.66.138 port 60922
> Answering with capability 0x100 (g729)
> Answering with non-codec capability 0x1 (telephone-event)
> Reliably Transmitting:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK670cb15b
> From: "Matt S" ;tag=as5f22d23c
> To: 
> Contact: 
> 
> --snip--
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Re: [Asterisk-Users] IAX2 register Refresh

[Liaan vd Merwe]

yes, that is true, but luckely all my dns etc is
sitting on firewall 
machine, so no natting except for iax. i've been
monitoring this for last 
few hours, and the only connections remaining is the
iax.

thanks
liaan

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

Sent: Friday, February 04, 2005 10:20 PM
Subject: Re: [Asterisk-Users] IAX2 register Refresh


>> > I been looking into the whole code strugture of
chan_iax and i see 
>> > there is
>> > a option to specify the refresh rate of
registrations: But there is no 
>> > code
>> > to actually load this from the config file thus i
changed the setting 
>> > in
>> > chan_so.h, and recompiled. But still my refresh
rate is 60 sec.
>> >
>> > I need to get this down to 15 sec (nat /pat
firewall issue)
>>
>> Fix the NAT/PAT device?  15s timeout is insane,
most things have many, 
>> many
>> minutes for timeout.
>
> Carefull. Short duration timeouts for udp packets
are very common
> otherwise tables grow beyond available memory rather
quickly due to
> dns, etc, type requests.
>
>
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Re: [Asterisk-Users] Server Criteria

[Steven Critchfield]

On Fri, 2005-02-04 at 11:21 -0800, Spencer Nassar wrote:
> I've been doing a lot of background reading/searching of this list, 
> voip-info.org, and Google, looking to define a good candidate for a 
> server platform.  I'm very interested in thoughts from others!  So here 
> goes...
> 
> Axiom 1:  if you are not doing doing much transcoding (converting 
> between codecs), the bottleneck for supporting high volumes of 
> simultaneous calls is system bus speed, not CPU power
> ---> points to a 64 bit AMD Opteron system, and maybe just one of the 
> two processor slots populated.  Bus is twice as wide as a 32 bit 
> system, and operates at 1.8GHz (a lot faster than a 64 bit Zeon 
> system).  Then add the second processor to the board if you see you 
> need it.

This I guess depends mostly on what channels you are using. Since a
fully loaded E1 is only 2mbps and you are unlikely to want to place more
than a few per machine before you start building redundancy into your
systems, You will be moving less data than a 100mb ethernet card, but on
a realtime basis instead of being able to delay service. On the other
hand, if you are talking all VoIP, it might help to service that NIC a
little faster to keep all your callers happy with latency.  

> Axiom 2:  Get lots of memory
> ---> I haven't seen this quantified, and plan to do some testing.  I'll 
> post results here, but can anyone share any insights?  I'm planning to 
> start at 2GB, and go up from there if I see swap getting used.
> - what would an alaw to alaw connection consume (if it didn't hand 
> off)?
> - what about a 5 call alaw meetme bridge (and how much memory per 
> incremental caller)

Other than a few structures relating to each call and it's state, you
only pass about 160 samples around to be worked on at a time and in
signed linear thats only 320 bytes. So per call incremental memory
shouldn't be very much, probably on the order of a few K at best. We run
our main systems right now with only 256m and they handle T1 capacity.

> Axiom 3: Don't allow any disk IO
> ---> I'm assuming this is related to #2 - get lots of memory to avoid 
> swap to disk.  Other issues or thoughts?

Disk IO isn't a killer. You just have to be careful about not pounding
the drives. Many of us record calls directly to the main drives of the
asterisk machine. Voice mail goes straight to main drives. Just don't
put a database that is going to see much more than extremely sparse
access on the same machine.

> Axoim 4: Come codecs will take advantage of the faster floating point 
> of a 64 bit system
> ---> unknown... has anyone seen this?  Will Asterisk, compiled in a 64 
> bit Linux environment, reap these or other benefits from being on a 64 
> bit system (other than the system bus speed)?

Is the floating point unit that much faster than an equivalent clocked
non 64 bit system? You will probably see similarly improved performance
on the same clock speed rises. Otherwise, I don't know much about the
math being used internally to help on that question.

Hope that sharpens some of your questions to get the data you are
looking for.

-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] IAX2 register Refresh

[Rich Adamson]

> > I been looking into the whole code strugture of chan_iax and i see there is
> > a option to specify the refresh rate of registrations: But there is no code
> > to actually load this from the config file thus i changed the setting in
> > chan_so.h, and recompiled. But still my refresh rate is 60 sec.
> >
> > I need to get this down to 15 sec (nat /pat firewall issue)
> 
> Fix the NAT/PAT device?  15s timeout is insane, most things have many, many 
> minutes for timeout.

Carefull. Short duration timeouts for udp packets are very common
otherwise tables grow beyond available memory rather quickly due to
dns, etc, type requests.


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[Asterisk-Users] Updateing to Stable from CVS

[AJ Grinnell]

I am looking to update our Asterisk system from Asterisk
CVS-D2004.07.03.04.00.00-08/26/04-04:27:13 to whatever is the latest
stable version. Looks like maybe 1.0.5? Is this in fact the latest
stable version, and if so, is there anything that I should look out
for/ be aware of besides features.conf? Thank you.
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Re: [Asterisk-Users] Server Criteria

[Sergey Kuznetsov]

I have Dual Opteron 2 GHz with 4 Gb memory.
I don't have huge load right now, and system load is almost 0.00 even if 
it uses slinear to G.729 transcoding. I have Wildcard 410P installed.
Works good for me.

Spencer Nassar wrote:
I've been doing a lot of background reading/searching of this list, 
voip-info.org, and Google, looking to define a good candidate for a 
server platform.  I'm very interested in thoughts from others!  So 
here goes...

Axiom 1:  if you are not doing doing much transcoding (converting 
between codecs), the bottleneck for supporting high volumes of 
simultaneous calls is system bus speed, not CPU power
---> points to a 64 bit AMD Opteron system, and maybe just one of the 
two processor slots populated.  Bus is twice as wide as a 32 bit 
system, and operates at 1.8GHz (a lot faster than a 64 bit Zeon 
system).  Then add the second processor to the board if you see you 
need it.

Axiom 2:  Get lots of memory
---> I haven't seen this quantified, and plan to do some testing.  
I'll post results here, but can anyone share any insights?  I'm 
planning to start at 2GB, and go up from there if I see swap getting 
used.
   - what would an alaw to alaw connection consume (if it didn't hand 
off)?
   - what about a 5 call alaw meetme bridge (and how much memory per 
incremental caller)

Axiom 3: Don't allow any disk IO
---> I'm assuming this is related to #2 - get lots of memory to avoid 
swap to disk.  Other issues or thoughts?

Axoim 4: Come codecs will take advantage of the faster floating point 
of a 64 bit system
---> unknown... has anyone seen this?  Will Asterisk, compiled in a 64 
bit Linux environment, reap these or other benefits from being on a 64 
bit system (other than the system bus speed)?

Also, any experience with Asterisk on an Opteron out there?  Any 
unexpected issues?  How about card drivers?

Thanks!  I hope this spurs an interesting exchange of ideas that is of 
value to many.

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--
All the Best!
Sergey.
=
Sergey Kuznetsov
President/CEO
High Intellectual Technologies, Inc.
  Web: http://www.hitcalls.com
   E-mail: [EMAIL PROTECTED]
Business phone: (416) 548-9700
 Mobile phone: (647) 287-8448
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Re: [Asterisk-Users] MeetMe & ztdummy

[Kristian Kielhofner]

Matthew Laird wrote:
On Fri, 4 Feb 2005, Kristian Kielhofner wrote:

You do not need ztdummy.  Just zaprtc.

Alright, one more question before I try this So not having a zaptel
interface, is this why MeetMe didn't build when I initially made Asterisk?
I noticed th app_meetme.so wasn't even in the modules directory after I
installed it, unlike on my machine at home which does have a zaptel
interface.  With zaprtc installed, do I then rebuild Asterisk to get the
MeetMe app to build and install?
Thanks.
Now that I have not heard off.  Perhaps if you built Asterisk without 
Zaptel installed that might happen.  But I don't even know if that is 
possible...

--
Kristian Kielhofner
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Re: [Asterisk-Users] MeetMe & ztdummy

[Matthew Laird]

On Fri, 4 Feb 2005, Kristian Kielhofner wrote:

> You do not need ztdummy.  Just zaprtc.

Alright, one more question before I try this So not having a zaptel
interface, is this why MeetMe didn't build when I initially made Asterisk?
I noticed th app_meetme.so wasn't even in the modules directory after I
installed it, unlike on my machine at home which does have a zaptel
interface.  With zaprtc installed, do I then rebuild Asterisk to get the
MeetMe app to build and install?

Thanks.
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Re: [Asterisk-Users] HP ProLiant server for Asterisk

[Dana Olson]

On Fri, 4 Feb 2005 19:07:14 +0100 (CET), Christoph Rothe
<[EMAIL PROTECTED]> wrote:
> On Fri, 4 Feb 2005, Dana Olson wrote:
> 
> > I'm looking at ordering a server from HP. I checked around on Google
> > and found in the Wiki that the ProLiant DL380 is supposed to be known
> > to work with *.
> 
> > HP ProLiant DL380 G4 Server w/ the following options:
> > Intel Xeon 3.20GHz/1MB
> > 2GB REG PC2-3200 (2 X 1GB)
> > HP ProLiant Battery Backed Write Cache Enabler for SA6i
> > RAID 1 drive set
> > HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive
> > HP 36.4GB Pluggable Ultra320 SCSI 15,000 rpm (1") Universal Hard Drive
> > Hot Plug Redundant Power Supply Module
> > HP Redundant Fan Option Kit (3 fans)
> > 1.44MB Floppy Disk Drive
> > Slimline 24X CD-ROM
> 
> Nice setup I think. The only thing with the DL380G3 and G4 is that it
> freezes in some situations. This has been reported in the company
> where I work and is a known bug to HP which will be solved by some
> firmware upgrade.
> 
> However the problem occurs only in some special situation noone can
> really describe (cosmic rays ?! ;-) and at that location where I work
> we installed SAP on a DL380G3 and it never froze.
> 
> And at the end let me say that personally I like the DL380s. You can
> even resize Logical Drives of the RAID (e.g. when putting in
> some new harddisc) without shutting down the server!
> 
> Good Luck!
> 
> Christoph
> 

Thanks for all of the replies, both on and off list.

I should have also mentioned that I'll be installing Debian Stable on
it. Are you using Debian or something else?

There seems to be an issue with the G4 and the Digium TE cards, so I'm
unsure about placing the order at this time. I've been eyeing the
Sangoma A104 card and the ipVolution TDM120 (not out yet, I believe)
as an alternative.

I have a service contract with an HP provider, so I'd prefer to stick
with an HP server, if at all possible. However, if the T1 interfaces
cause issues, there may be no choice.
--
Dana
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Re: [Asterisk-Users] MeetMe & ztdummy

[Leif Madsen]

On Fri, 4 Feb 2005 10:50:15 -0800 (PST), Matthew Laird <[EMAIL PROTECTED]> 
wrote:
> 
> On Fri, 4 Feb 2005 [EMAIL PROTECTED] wrote:
> 
> > I've heard that 2.6 kernel does not need usb hardware for ztdummy to
> > function. Maybe someone else can confirm... although this would require a
> > complete reinstall for you.
> 
> Yes, the unfortunate part is this is a headless box somewhere in Virginia
> I don't have access to.  So I'm limited on what I can do I don't know
> how else I can get conference calls to work.  I'm open to opinions. :)

The 2.4 kernel requires the usb-uhci module in order to obtain timing
from the USB module on your motherboard. This has to be a UHCI and not
OHCI based chip or ztdummy will not work. As was mentioned earlier,
you can also use a 2.6 kernel as the timing can be obtained from the
kernel.

If neither of these options are possible you are out of luck for a
timing interface for conferencing.

Thanks,
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Re: [Asterisk-Users] MeetMe & ztdummy

[Kristian Kielhofner]

Matthew Laird wrote:
On Fri, 4 Feb 2005, Kristian Kielhofner wrote:

I have rmmod'd rtc many times.  You shouldn't have a problem.  Just
rmmod rtc and modprobe zaprtc.  That will get your basic rtc
functionality back.  Then run "rtcsetup &" and you should be okay.

Alright, I'm trying to fully understand so I remove rtc and add
zaprtc.  Then do I still need ztdummy on top of this?  Or will MeetMe then
work with just zaprtc?

P.S. - However, it goes without saying that if this hoses your box for
whatever reason, it's not my fault.  But you really should be just fine!

Well, fortunately as long as I do it manually and not change the config
files this machine has a "rapid reboot" function on their web control
pannel I can always use. :)  So you're off the hook.
Thanks again.
You do not need ztdummy.  Just zaprtc.
--
Kristian Kielhofner
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Re: [Asterisk-Users] why asterisk and ser

[Philipp von Klitzing]

Hi!

> Why would u use asterisk and ser together and what is the big
> difference? 

Why would you opt to not use Google?

Cheers, Philipp


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[Asterisk-Users] Server Criteria

[Spencer Nassar]

I've been doing a lot of background reading/searching of this list, 
voip-info.org, and Google, looking to define a good candidate for a 
server platform.  I'm very interested in thoughts from others!  So here 
goes...

Axiom 1:  if you are not doing doing much transcoding (converting 
between codecs), the bottleneck for supporting high volumes of 
simultaneous calls is system bus speed, not CPU power
---> points to a 64 bit AMD Opteron system, and maybe just one of the 
two processor slots populated.  Bus is twice as wide as a 32 bit 
system, and operates at 1.8GHz (a lot faster than a 64 bit Zeon 
system).  Then add the second processor to the board if you see you 
need it.

Axiom 2:  Get lots of memory
---> I haven't seen this quantified, and plan to do some testing.  I'll 
post results here, but can anyone share any insights?  I'm planning to 
start at 2GB, and go up from there if I see swap getting used.
   - what would an alaw to alaw connection consume (if it didn't hand 
off)?
   - what about a 5 call alaw meetme bridge (and how much memory per 
incremental caller)

Axiom 3: Don't allow any disk IO
---> I'm assuming this is related to #2 - get lots of memory to avoid 
swap to disk.  Other issues or thoughts?

Axoim 4: Come codecs will take advantage of the faster floating point 
of a 64 bit system
---> unknown... has anyone seen this?  Will Asterisk, compiled in a 64 
bit Linux environment, reap these or other benefits from being on a 64 
bit system (other than the system bus speed)?

Also, any experience with Asterisk on an Opteron out there?  Any 
unexpected issues?  How about card drivers?

Thanks!  I hope this spurs an interesting exchange of ideas that is of 
value to many.

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

[Erick Perez]

Has anyone on this list have a way to contact ServerWorks? they make
the mobos for the G4.
I dont have a G4 but i do know HP in the G line uses ServerWorks

I have to make a full stop ordering on 2 G4 monsters because of this
thread...However one friend is using a sangoma card without
problems


TE410P/ServerWork motherboard combo not working because of bus problems

my less than 1 cent



On Mon, 31 Jan 2005 20:42:47 +1100, Eric Bishop <[EMAIL PROTECTED]> wrote:
> Did anyone get anywhere with this thread? Any HP G4 series servers working?
> 
> 
> On Wed, 26 Jan 2005 09:46:31 +1100, Eric Bishop <[EMAIL PROTECTED]> wrote:
> > Has anyone had any luck with this issue and new Asterisk/Zaptel
> > releases (1.05/1.04)? I am still searching for a solution and waiting
> > for that Eureka! moment..
> >
> >
> > On Thu, 20 Jan 2005 09:20:09 +0100, Tais M. Hansen <[EMAIL PROTECTED]> 
> > wrote:
> > > On Wednesday 19 January 2005 23:15, Eric Bishop wrote:
> > > > Well guys this is truly bizarre. I managed to get a DL360 G3 to show
> > > > interrupts with FC2 but not FC3. Exact same config and setup
> > > > proceedure. Ofcourse neither FC2 or FC3 show interrupts with the DL360
> > > > G4. I think TE410P is just a flakey card.
> > > > Anyone got a DL360 G3 going with a TE410P and FC3?
> > >
> > > I did manage to get a TE110P running on the DL380 G4. Still can't get the
> > > TE410P working in the G4 though. Supports your theory.
> > >
> > > Sadly we're now being forced to look elsewhere for PRI cards.
> > >
> > > --
> > > Regards,
> > > Tais M. Hansen
> > > ComX Networks A/S
> > > Tel: +45-70257474
> > > Fax: +45-70257374
> > >
> > >
> > >
> >
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-- 

---
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)
Panama, Republic of Panama
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Re: [Asterisk-Users] MeetMe & ztdummy

[Matthew Laird]

On Fri, 4 Feb 2005, Kristian Kielhofner wrote:

> I have rmmod'd rtc many times.  You shouldn't have a problem.  Just
> rmmod rtc and modprobe zaprtc.  That will get your basic rtc
> functionality back.  Then run "rtcsetup &" and you should be okay.

Alright, I'm trying to fully understand so I remove rtc and add
zaprtc.  Then do I still need ztdummy on top of this?  Or will MeetMe then
work with just zaprtc?

> P.S. - However, it goes without saying that if this hoses your box for
> whatever reason, it's not my fault.  But you really should be just fine!

Well, fortunately as long as I do it manually and not change the config
files this machine has a "rapid reboot" function on their web control
pannel I can always use. :)  So you're off the hook.

Thanks again.
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RE: [Asterisk-Users] Call forwarding

[Jay Milk]

With GSM cell phones you have a few options.  My cell-phone is set up to
forward unavailable (busy, no answer, etc) calls to a special DID on my
* box.  When calls came in through that DID, I know they're forwarded
from the cell and they go into voice-mail after a few rings.

I'm planning on scripting this a little better so that calls which ring
our home extensions as well as my cell will be ignored when they come in
on my cell-DID within 3 seconds of ringing on the initial DID.

> -Original Message-
> From: Ryan Courtnage [mailto:[EMAIL PROTECTED] 
> Sent: Friday, February 04, 2005 12:55 PM
> To: Pedro; Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Call forwarding
> 
> 
> On Fri, 2005-04-02 at 13:11 -0500, Pedro wrote:
> > Cool idea.
> > 
> > One question - let's say someone specifies their home phone 
> number and 
> > their cell number.  How do you take into the account if the cell VM 
> > picks up (ie. if cell is out of coverage and VM greeting is played)?
> 
> AFAIK, there isn't much you can do in this scenario - other 
> than ringing your house for a few rings before ringing your 
> house AND the cell.  Even then, the cell provider's 'out of 
> the service area' message would answer the call.
> 
> 
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Re: [Asterisk-Users] MeetMe & ztdummy

[Kristian Kielhofner]

Matthew Laird wrote:
On Fri, 4 Feb 2005 [EMAIL PROTECTED] wrote:

I've heard that 2.6 kernel does not need usb hardware for ztdummy to
function. Maybe someone else can confirm... although this would require a
complete reinstall for you.

Yes, the unfortunate part is this is a headless box somewhere in Virginia
I don't have access to.  So I'm limited on what I can do I don't know
how else I can get conference calls to work.  I'm open to opinions. :)
Thanks.

I have rmmod'd rtc many times.  You shouldn't have a problem.  Just 
rmmod rtc and modprobe zaprtc.  That will get your basic rtc 
functionality back.  Then run "rtcsetup &" and you should be okay.

P.S. - However, it goes without saying that if this hoses your box for 
whatever reason, it's not my fault.  But you really should be just fine!

And yes, ztdummy does not need usb-uhci in 2.6 because the default 
kernel timer is already 1000hz.  I have heard that you can do this on 
FreeBSD if you have HZ=1000 in your kernel config.  Not really related 
to this post, but interesting nonetheless...

--
Kristian Kielhofner
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RE: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

[Robert Webb]

What we have been discussing with no ringback, is if you have a caller
call in through your DID line and say dials an extension, then after
using the dial command, the caller hears silence and no ringing tone.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, February 04, 2005 1:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

I currently use LiveVoip for service (2 DIDs, 1 Toll Free 8ZZ and
outgoing).  Although it took longer to get installed than was expected
(about a day and a half), I find their service to be quite acceptable.
Could you clarify the "no ringback" condition?  There was a situation at
first in which I got fast busies (reorder) when calling the incoming
services.  I suspect that it took a while to provision the service with
THEIR service providers (Lever 3, Quest, etc.)

I reported the problem via email, and was pleased with the support I
received.  At least they ARE willing to quickly refund money (unlike
some other providers I have read about on the list) and seem sincere
about their desire to provide quality service.

Norm Zimon
Globex Telecom
www.globextele.com



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Eissler
Sent: Thursday, February 03, 2005 7:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Brian
Dingman
Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

Based on the support and management responses that have been posted to
this list it doesn't sound to me (at least) like LiveVoip really wants
business from * users anyhow. They blame a lot of problems on * and are
quick to offer a refund. There are plenty of DID providers that are more
asterisk-friendly.

-mark

On Feb 2, 2005, at 1:25 PM, Brian Dingman wrote:

> Finally got a reply from LV support. Not what I was hoping for.
> Hopefully they will file a bug with Digium since they investigated the

> issue not holding my breath.
>
> Since this is such basic * functionality that they can't seem to
> accomplish I would think twice before aquiring DID's from them.
>
>  LiveVoip Support
>
> Our people have looked into this matter over the past few days. They
> tell me that it is a problem with Asterisk.
> We are not going to be able to help you with this. If you would like a

> refund so that you can migrate to another service provider we will be
> happy to do so. With each rev. of Asterisk

> more
> and more improvements are made.
> At some point these issues may resolve but, for the time being it is
> not a problem we can help you with.
>
>
> On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier
> <[EMAIL PROTECTED]> wrote:
>> I just got a couple of numbers (activated Friday) from livevoip, I am

>> having
>> similar issues.
>>
>> When you call the number, I get ring back, but as soon as IVR picks
>> up, I
>> should here "extensioni" I don't hear that but then I dial an
>> extension
>> number and there is no ring back.  I don't have this issue from other

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

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Re: [Asterisk-Users] Call forwarding

[Andrew Thompson]

Adam Robins wrote:
Works beautifully!  Thanks. 
Could you post here, or to the wiki, or just back to myself the 
configuration you're using to implement this?

Thanks.
--
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
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Re: [Asterisk-Users] MeetMe & ztdummy

[David Coulson]

[EMAIL PROTECTED] wrote:
I've heard that 2.6 kernel does not need usb hardware for ztdummy to 
function. Maybe someone else can confirm... although this would require 
a complete reinstall for you.
I have ztdummy loaded under 2.6 without USB hardware and it works fine.
David
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Re: [Asterisk-Users] Re: Can't get Polycom auto-answer to work

[Kristian Kielhofner]

Noah Miller wrote:
-- Executing SetVar("SIP/27-8b27", "ALERT_INFO=Ring Answer") in
new
stack
-- Executing Dial("SIP/27-8b27", "SIP/19|20") in new stack
-- Called 19
-- SIP/19-e562 is ringing
and the phone just rings.  Any ideas?  Is it the firmware version?
Am I
not setting ALERT_INFO correctly?

It's possible, but hard to say since you didn't tell us what version of
Asterisk you are using. If you are using CVS HEAD (and not the stable
series), you will have to set _ALERT_INFO instead, so it will be
carried
down into the channel that the Dial app creates.

It's an older version of CVS HEAD:
CVS-HEAD-11/03/04-14:59:37
I tried with _ALERT_INFO also - the result is the same.

Here's what is actually working with a spa3k and CVS-HEAD-01/08/05:
exten => 3020,1,SetVar(_ALERT_INFO=bellcore-r3) ; selects Ringer
exten => 3020,2,Dial(SIP/3020,25,r)
exten => 3020,3,Voicemail(u3020)
exten => 3020,103,Voicemail(b3020)
exten => 3020,104,Hangup
I don't have a Polycom handy to test with.

I did a "sip debug peer 19", and here is part of that debug that shows 
that asterisk is trying to send the alert-info variable to the phone.

Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK304b1809
From: "Noah Miller" ;tag=as44d2096b
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 04 Feb 2005 17:39:53 GMT
Alert-info: Ring Answer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 160
So I guess the problem is in my config for the phone?  Or maybe asterisk 
has to send "alert-info" more than just once?  Does anybody have this 
auto-answer config working reliably on a Polycom phone?

Thanks!
Noah
Noah,
	Please see my Polycom config files at 
http://www.kriscompanies.com/modules.php?name=Downloads&d_op=viewdownload&cid=1

I have a setup for a Ring-answer and autoanswer.
--
Kristian Kielhofner
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Re: [Asterisk-Users] Call forwarding

[Ryan Courtnage]

On Fri, 2005-04-02 at 13:11 -0500, Pedro wrote:
> Cool idea.
> 
> One question - let's say someone specifies their home phone number and
> their cell number.  How do you take into the account if the cell VM
> picks up (ie. if cell is out of coverage and VM greeting is played)?

AFAIK, there isn't much you can do in this scenario - other than ringing
your house for a few rings before ringing your house AND the cell.  Even
then, the cell provider's 'out of the service area' message would answer
the call.


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RE: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

[nzimon]

I currently use LiveVoip for service (2 DIDs, 1 Toll Free 8ZZ and
outgoing).  Although it took longer to get installed than was expected
(about a day and a half), I find their service to be quite acceptable.
Could you clarify the "no ringback" condition?  There was a situation at
first in which I got fast busies (reorder) when calling the incoming
services.  I suspect that it took a while to provision the service with
THEIR service providers (Lever 3, Quest, etc.)

I reported the problem via email, and was pleased with the support I
received.  At least they ARE willing to quickly refund money (unlike
some other providers I have read about on the list) and seem sincere
about their desire to provide quality service.

Norm Zimon
Globex Telecom
www.globextele.com



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Eissler
Sent: Thursday, February 03, 2005 7:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Brian
Dingman
Subject: Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

Based on the support and management responses that have been posted to 
this list it doesn't sound to me (at least) like LiveVoip really wants 
business from * users anyhow. They blame a lot of problems on * and are 
quick to offer a refund. There are plenty of DID providers that are 
more asterisk-friendly.

-mark

On Feb 2, 2005, at 1:25 PM, Brian Dingman wrote:

> Finally got a reply from LV support. Not what I was hoping for.
> Hopefully they will file a bug with Digium since they investigated the
> issue not holding my breath.
>
> Since this is such basic * functionality that they can't seem to
> accomplish I would think twice before aquiring DID's from them.
>
>  LiveVoip Support
>
> Our people have looked into this matter over the past few days. They 
> tell me
> that it is a problem with Asterisk.
> We are not going to be able to help you with this. If you would like a
> refund so that you can migrate to another
> service provider we will be happy to do so. With each rev. of Asterisk

> more
> and more improvements are made.
> At some point these issues may resolve but, for the time being it is 
> not a
> problem we can help you with.
>
>
> On Sun, 30 Jan 2005 18:15:10 -0500, Steven Frazier 
> <[EMAIL PROTECTED]> wrote:
>> I just got a couple of numbers (activated Friday) from livevoip, I am

>> having
>> similar issues.
>>
>> When you call the number, I get ring back, but as soon as IVR picks 
>> up, I
>> should here "extensioni" I don't hear that but then I dial an 
>> extension
>> number and there is no ring back.  I don't have this issue from other

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

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Re: [Asterisk-Users] Snom Phones Volume

[timebandit001]

> They work great, EXCEPT I have to have the volume turned all the
> way up in order to hear the conversation on the other end.

I don't know if there's a fix, but I experiences exactly the same
thing. This is the only negative thing I can say about the snom
phones. Other thant this, they are really great phones

If somebody know how to fix this, it would be more than welcome
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Re: [Asterisk-Users] MeetMe & ztdummy

[Richard Lyman]

Matthew Laird wrote:
I'm running into a bit of a problem setting up conference calls.  The box
I rent at a colo doesn't seem to have USB hardware When I try to load
usb-uhci I receive a "device does not exist" error.  Which means I can't
load ztdummy
The system has a rtc clock module, so zaprtc won't work... (which I'm
scared to unload rtc because I don't have physical access to the
machine...)
I've tried app_conference, but it doesn't seem to have been updated for
the latest release of Asterisk.
So, I'm stuck can anyone advise me on how I can get some form of
conference calling given all these factors?
Thanks.
sadly your choices are limited.
either...
make sure there is someone at the colo to help in case you hose rtc.
send them a digium card to toss in it.
buy your own box to be colo'd and mod it before it goes.
(i'm sure there are other methods)
but if you can't do the above, i think you will be in for a rough 
ride.  because, when something goes wrong, and it will... you'll 
be in the same boat.

good luck
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Re: [Asterisk-Users] MeetMe & ztdummy

[Matthew Laird]

On Fri, 4 Feb 2005 [EMAIL PROTECTED] wrote:

> I've heard that 2.6 kernel does not need usb hardware for ztdummy to
> function. Maybe someone else can confirm... although this would require a
> complete reinstall for you.

Yes, the unfortunate part is this is a headless box somewhere in Virginia
I don't have access to.  So I'm limited on what I can do I don't know
how else I can get conference calls to work.  I'm open to opinions. :)

Thanks.
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[Asterisk-Users] FC2 RPMS are updated

[Andrew McRory]

Hello,
I've uploaded current builds to
   ftp://ftp.linuxsys.com/pub/releases/FC2/asterisk-CVS
   ftp://ftp.linuxsys.com/pub/releases/FC2/asterisk-1.0
Regards,
--
Andrew McRory - President and Chief Techincal Officer
Linux Systems Engineers, Inc. - http://www.linuxsys.com
Located in beautiful Tallahassee, Florida
Office  850-224-5737
Office  850-575-7213
Mobile  850-294-7567
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Re: [Asterisk-Users] MeetMe & ztdummy

[rsenykoff]

I'm running into a bit of a problem setting up conference
calls.  The box
I rent at a colo doesn't seem to have USB hardware When I try to load
usb-uhci I receive a "device does not exist" error.  Which
means I can't
load ztdummy

The system has a rtc clock module, so zaprtc won't work... (which I'm
scared to unload rtc because I don't have physical access to the
machine...)

I've tried app_conference, but it doesn't seem to have been updated for
the latest release of Asterisk.

So, I'm stuck can anyone advise me on how I can get some form of
conference calling given all these factors?


I've heard that 2.6 kernel does not
need usb hardware for ztdummy to function. Maybe someone else can confirm...
although this would require a complete reinstall for you.

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[Asterisk-Users] MeetMe & ztdummy

[Matthew Laird]

I'm running into a bit of a problem setting up conference calls.  The box
I rent at a colo doesn't seem to have USB hardware When I try to load
usb-uhci I receive a "device does not exist" error.  Which means I can't
load ztdummy

The system has a rtc clock module, so zaprtc won't work... (which I'm
scared to unload rtc because I don't have physical access to the
machine...)

I've tried app_conference, but it doesn't seem to have been updated for
the latest release of Asterisk.

So, I'm stuck can anyone advise me on how I can get some form of
conference calling given all these factors?

Thanks.
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RE: [Asterisk-Users] X-lite to Cisco ATA - no RTP

[Keith Burns]

Title: X-lite to Cisco ATA - no RTP









Interesting,
SUSE firewall allows SIP but not RTP out of the box.

 



-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Keith Burns
Sent: Friday, February 04, 2005
8:02 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] X-lite
to Cisco ATA - no RTP

 

Hi there,

I have X-lite and a Cisco ATA
on the same hub (i.e. no NAT, no ACLs) as my Asterisk box.

Ethereal shows normal SIP
signaling when I call from
X-lite to the
ATA.

Ethereal also shows
RTP is passed from X-lite to Asterisk, and RTP is passed from the ATA to Asterisk, but no RTP from Asterisk to
either device.

(Note that Ethereal does show
the SIP signaling
packets originating from
Asterisk, so nothing funky with my Ethereal filter either)

Has anyone run into anything
similar? Any pointers? I set up all the extensions using AMP.

If you need specific configs,
I am happy to provide.

Cheers

Keith.








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Re: [Asterisk-Users] BRI in the US?

[Michael Welter]

Michael Graves wrote:
OK, I asked this about a week back and met with no repsonse at all. But
perhaps its worth trying again. 

Does anyone on-list have * running BRI to their local telco? I'm
considering this as an alternative to my TDM400p card.
I had an HFC card and a BRI circuit from Qwest, but I would never make 
it work.  As I recall, I could receive incoming calls but could never 
make outgoing calls.  Had to do with SPID (service provisioning id?) 
was/is not supported in the software.

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Re: [Asterisk-Users] No Playback() when Digicom TE110P enabled

[Asterisk]

oh bugger. Just when I thought that I was clever, I re-read the question.
Forget what I spewed.
:(
Julian
Asterisk wrote:
A guess - because zaptel is not running, you don't have a timer. No 
timer, no sound ... use ztdummy or enable zaptel.

Julian
Gareth Blades wrote:
I have a Digicom TE110p card installed in our exchange. I have compiled
and installed libpri, zaptel and recompiled and installed asterisk.
I have configured udev as I am running Fedora Core 3.
The problem that I have is that when zaptel is not running everything
works fine. However when I start zaptel (service start zaptel) then I
can make normal calls ok but the 'Playback()' function,voicemail etc...
dont work and all I get on the IAX client is silence.
In /etc/sysconfig/zaptel I have:-
TELEPHONY=yes
MODULES="$MODULES wcte11xp" # TE110P - Single Span T1/E1 Card
Any idea what is going wrong?
/etc/asterisk/zapata.conf has not been altered (card is not currently
connected to anything).
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RE: [Asterisk-Users] Call forwarding

[Adam Robins]

Works beautifully!  Thanks. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Friday, February 04, 2005 10:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call forwarding

Adam Robins wrote:

> What I would like to do is take the variable containing the number 
> retrieved from CFIM, place it on the stack as the called number, and 
> have it reenter the dial plan, similar to the WAITEXTEN command.

Easy-peasy. That's what Local channels are for.

If you have retrieved the CFIM value into a variable called CFIM, and
the user's phone would normally dial via a context called
"customer-dial", then:

exten => ...,...,Dial(Local/[EMAIL PROTECTED])

This will process the call exactly the same way as if the phone user had
dialed that number from their phone.
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Re: [Asterisk-Users] Re: Can't get Polycom auto-answer to work

[Kevin P. Fleming]

Noah Miller wrote:
User-Agent: Asterisk PBX
Date: Fri, 04 Feb 2005 17:39:53 GMT
Alert-info: Ring Answer
This looks odd... that header name should be "Alert-Info". Try changing 
that in chan_sip to see if it makes any difference.
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RE: [Asterisk-Users] Callerid problems with 1.0.5

[Jay Milk]

Can someone clarify what's going on here?

I'm running 1.0.5, and I see caller-id come through just fine from one
extension to the other, as well as for incoming and outgoing calls
(iax2).  What are you folks seeing there?


> -Original Message-
> From: mattf [mailto:[EMAIL PROTECTED] 
> Sent: Friday, February 04, 2005 11:12 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Callerid problems with 1.0.5
> 
> 
> Supposedly when a call is parked and/or transferred they 
> wanted the callerid to reflect the person who is on that 
> phone call. That's the only reason I saw mentioned for the 
> change. As for why it was made default I have no idea, but 
> now in CVS_HEAD at least you can turn that feature off. And 
> in v1.0.5 you can patch your system to remove that feature.
> 
> MATT---
> 
> 
> -Original Message-
> From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
> Sent: Friday, February 04, 2005 12:01 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Callerid problems with 1.0.5
> 
> 
> mattf wrote:
> 
> > Also in CVS_HEAD preserving original callerid has been given a flag 
> > 'o' in the dial string.
> 
> I have to wonder why the default behavior was changed to this 
> non-standard usage though; in what situations do we want the 
> CLID/CNAM 
> of the _recipient_ to be passed to them? 
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Re: [Asterisk-Users] No Playback() when Digicom TE110P enabled

[Asterisk]

A guess - because zaptel is not running, you don't have a timer. No 
timer, no sound ... use ztdummy or enable zaptel.

Julian
Gareth Blades wrote:
I have a Digicom TE110p card installed in our exchange. I have compiled
and installed libpri, zaptel and recompiled and installed asterisk.
I have configured udev as I am running Fedora Core 3.
The problem that I have is that when zaptel is not running everything
works fine. However when I start zaptel (service start zaptel) then I
can make normal calls ok but the 'Playback()' function,voicemail etc...
dont work and all I get on the IAX client is silence.
In /etc/sysconfig/zaptel I have:-
TELEPHONY=yes
MODULES="$MODULES wcte11xp" # TE110P - Single Span T1/E1 Card
Any idea what is going wrong?
/etc/asterisk/zapata.conf has not been altered (card is not currently
connected to anything).
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[Asterisk-Users] callback on busy

[Bartosz Jozwiak]

Hello everybody,
I would like to implement "callback" function.
When I call a person and his extension is busy I can press, for example, 5
and get a callback when his phone is not busy anymore.
When I create a call file and copy it to spool call folder
asterisk makes a call. One problem is that when extension is still busy
my phone rings and I get busy tone of the person who I am trying to reach.
Is there another way to do it ? Or do I need to check first if channel is 
free or
still busy ? Can anybody give me some hints ?

thx in advance.
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Re: [Asterisk-Users] Call forwarding

[Pedro]

Cool idea.

One question - let's say someone specifies their home phone number and
their cell number.  How do you take into the account if the cell VM
picks up (ie. if cell is out of coverage and VM greeting is played)?


On Fri, 04 Feb 2005 10:41:28 -0700, Kevin P. Fleming
<[EMAIL PROTECTED]> wrote:
> Ryan Courtnage wrote:
> 
> > Can multiple Local channels be safely used in a single Dial command?
> > ie:
> >
> > exten
> > => ...,...,Dial(Local/[EMAIL PROTECTED],Local/[EMAIL 
> > PROTECTED],Local/[EMAIL PROTECTED])
> 
> Yes, using the standard "&" connector like you would use if you were
> dialing multiple SIP peers or any other peers.
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