Re: [Asterisk-Users] Asterisk & Meridian Tie Line

[Johann Steinwendtner]

The BT guy should check LD 73 block LPTI and prompt AFF.
If it is crc then you need crc4 as well.

Best regards

Hans

Steve Totaro schrieb:

Andy Kirby wrote:



I am new to the group but have searched the doc's FAQ's etc before 
posting here.


 We are attempting tie our asterisk server/service to the building's 
PBX, the building is in the UK and the local PBX is a meridian option 
11 installed and mainteined by BT.


BT Have installed a NTBK50AA E1-PRI card in the meridian with daughter 
cards NTBK51AA (D channel) and NTAK20BD (Clocking)


 I have asked BT to configure the card as a Master (Exchange end) E1 
Euro ISDN (Just like a standard ISDN30e)


 They claim to have done this in line with the model they use to 
interface to Cisco routers etc.


 I have installed a Digium TE411P in our server looped back the span 2 
port (Gives a green light and OK with same config as span 1) and am 
using a crossover cable to link the PBX to our server. (We tried a 
pucker BT cross over cable with exactly the same results as mine and a 
striahgt through gives us nothing at all, I guess as you might expect)


I have configured the Zap span for 1 clocking (Primary) 1 line build 
out, with the framing etc as CCS, HDB3, CRC4


 


But they don't appear to want to synchronise/talk to each other.

 

ZTTool claims that the span is up and down more times than a fiddlers 
elbow  and the clocking source is internal.


 ( Might I expect the alarm state to be constant if the framing etc 
was matched and the clock source to show as external ??)


 

The alarms are cylcling from red to red/yellow and finaly to 
red/yellow/recover before falling back to red and starting again.


 I think I may be missing something that is probably blindingly 
obvious to someone in the know.


 The BT guy has been very good and is trying to help us get this going 
but seems rather nonplussed with the terms CRC4, CSS and HDB3...


Please can somone help and point me (and I guess by extension the BT 
guy) in the right direction.


 


Cheers

 


Andy
 



Try your end with every different combination of settings applicable to 
EuroISDN.  I would start by removing the CRC4.  Try something, if it 
doesnt work, put it back and change another setting.



s
  



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Re: [Asterisk-Users] Quad BRI card

[Hadley Rich]

On Thursday 18 May 2006 18:35, stoffell wrote:
> Aside from being available.. What driver does it use?
> Will it be needing bristuff ? (that wouldn't work I guess)
>
> Or will the near future integrate BRI ( and hfc?) drivers in asterisk?
> And thus, making bristuff obsolete? (wich means, BRI users will be
> able to use cvs easily..)
>
> Just to make clear I'm very curious on this card. And yes I'm in europe ;)

I'm curious too, unfortunately I don't know anything more about it sorry.

hads.

-- 
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without any means.
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RE: [Asterisk-Users] Plan to free myself from AAH

[Mimmus]

AMP dialplan is full of garbage and perpaphs is not fully 1.2 compatible but
it is anyway an Asterisk, working dialplan!
I already tried to copy config files and Asterisk starts without warnings:
gradually I will clean out them from fax, queues, devices, ring groups,
weather reports, etc 



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Colin Anderson
> Sent: Wednesday, May 17, 2006 6:29 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Plan to free myself from AAH
> 
> What I did with AMP was take the best parts of it and 
> copy/paste to a clean extensions.conf, then add my 
> modifications onto it. Worked for me. 
> 
> -Original Message-
> From: Strom Carlson [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, May 17, 2006 10:21 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Plan to free myself from AAH
> 
> 
> On 5/17/06, Mimmus <[EMAIL PROTECTED]> wrote:
> 
> > I was thinking to this plan:
> > - install another server with Red Hat 4 U3
> > - install PHP, MySQL and other usefuls stuffs
> > - download latest version of Asterisk and third parts 
> applications I 
> > use
> > - compile all
> > - copy /etc/asterisk from old server to new, change only what is 
> > needed
> > - start and try
> >
> > Do you think is it OK?
> 
> I doubt it.  The problem I have with AAH / AMP / FreePBX is 
> that the configuration files are absolutely full of useless 
> garbage and are really not at all suitable for moving to a 
> standard asterisk install.
> 
> Set up a new server from scratch and start learning how to 
> configure asterisk manually.  Rebuild everything one step at 
> a time so that the functionality remains as you'd like it to 
> be, but that the actual configs aren't full of that FreePBX garbage :)
> 
> --
> Strom Carlson

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Re: [Asterisk-Users] Meetme conf

[Gavin Henry]

> hi there,
>
> i am wondering can meetme.conf able to support diffferent context. Cause
> currently, it has [rooms] context. ]
> is it possible to have same conference number with different context?
>
> thanks

Try it and see ;-)

-- 
Kind Regards,

Gavin Henry.

Open Source. Open Solutions(tm).
http://www.suretecsystems.com/
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Re: [Asterisk-Users] soekris hadware

[Jonathan Gonzalez]

Hi Christopher,

i know the place, in fact i've been reading a lot before post here.
The problem is that even if there are a lot of good documents,
personallly i can't see answered my doubts, and this is the reason i
wrote.

If you can be a bit more explicit and give me some light over my
questions i would really love it.

Thanks in advance.
Kind regards,

Jonathan GF



On 5/17/06, Christopher Snell <[EMAIL PROTECTED]> wrote:

Google and voip-info.org will have answers to all of your questions.


On 5/17/06, Jonathan Gonzalez < [EMAIL PROTECTED]> wrote:
>
Hi group,

i'm brand new and i would like to ask about soekris hardware. I read
along the web but i have some doubts that i think can be solved here.
My question are the following:

[...]


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Re: [Asterisk-Users] soekris hadware

[Jonathan Gonzalez]

Hi Oliver,

i understand you use astlinux, even if the version nunber shows the
product is quite new. If you have to decide between astlinux and
[EMAIL PROTECTED], thinking in use the pbx for basic thinks like MOH, IVR,
an advanced dialplan, 1 FXO, 3FXS, 3 SIP and no much more, which one
would you select?

Any thoughts will be welcomed.
Kind regards,

Jonathan GF



On 5/17/06, olivier.taylor <[EMAIL PROTECTED]> wrote:


 more kindly :

 http://www.astlinux.org/

 Olivier

 Christopher Snell a écrit :
Google and voip-info.org will have answers to all of your questions.


On 5/17/06, Jonathan Gonzalez < [EMAIL PROTECTED]> wrote:
> Hi group,
>
> i'm brand new and i would like to ask about soekris hardware. I read
> along the web but i have some doubts that i think can be solved here.
> My question are the following:
>
> [...]
>




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[Asterisk-Users] Unable to set channel to linear mode

[Koen Van Impe]

I have a TE110P connected in euroisdn as pri-cpe.
When I dial out from a sip phone to a number over the pri, I get an error
 
Unable to set channel 1 (index 0) to linear mode
 
On the destination phone, I only get a terrible noise when answering the call.
There doesn't seem to be a speech path...
 
Config: libpri 1.2.2 - zaptel 1.2.5 - asterisk 1.2.6 on Fedora Core 4 (kernel 2.6.11)
 
Anybody knows what this is all about?
 
K
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Re: [Asterisk-Users] soekris hadware

[Jonathan Gonzalez]

Hi Michael,

your document is very good. In fact this was one of the first i read.
I googled looking for soekris and asterisk and you appreared.

Anyway, your document do not cover the same setup i have. You point
the problem of the digium cards don't using it, while i have or i
think i require one of this card with a soekris.

I'm thinking buy only de SBC and look for another chassis where all
equipment fit fine. I say you don't cover, or this i think, because i
need to use FXO<->PBX<->FXS/SIP and you don't use this setup.

You call forward all calls over an ip-to-pstn account while i want to
receive incoming calls via pstn and call from internal sip phones to
sip phones and to pstn trhu asterisk (of course, paying for the call,
like a normal call).

Another difference is that you use CF with the SBC, while i want to
use the mini-hard disk option and make a complete installation of
[EMAIL PROTECTED]

It seems the product is quite stable and i couldn't see this about
astlinux in any place. I would appreciate your thougts about astlinux
and some recomendations will be welcomed.

Thanks for you tu answer and for your magnificent document.

Kind regards,

Jonathan GF

On 5/18/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:

Also...an article I wrote earlier this year

http://www.tomsnetworking.com/2006/01/13/how_to_asterisk_pbx/

Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
o(800) 905-6412
f(713) 864-8668
c(713) 201-1262



>  Original Message 
> Subject: Re: [Asterisk-Users] soekris hadware
> From: "olivier.taylor" <[EMAIL PROTECTED]>
> Date: Wed, May 17, 2006 11:12 am
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
>
>more kindly :
>
>  http://www.astlinux.org/
>
>  Olivier
>
>  Christopher Snell a écrit : Google and voip-info.org will have answers to 
all of your questions.
>
>
> On 5/17/06, Jonathan Gonzalez < [EMAIL PROTECTED]> wrote: Hi group,
>
>  i'm brand new and i would like to ask about soekris hardware. I read
>  along the web but i have some doubts that i think can be solved here.
>  My question are the following:
>
>  [...]
>
>
>
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Re: [Asterisk-Users] SIP debugging

[Klaus Darilion]

Andres wrote:

Hi Klaus,

The response to a CANCEL should be a "487 Request Terminated", not  a 
"200 OK".  Maybe your innovaphone Server is to blame.


Hi Andres.

No. The reply to the CANCEL is a 200 Ok. The reply to the cancelled 
INVITE is a 487.


regards
klaus
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Re: [Asterisk-Users] Quad BRI card

[Wayne Gemmell]

On Thursday 18 May 2006 03:35, Mark Coccimiglio wrote:
>  Otherwise the Diva server cards
> are a good option (extensive, but come highly recomended from most that
> I hear).  Good luck and happy hunting.
Ouch, you weren't joking. 1453 Euro!
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[Asterisk-Users] DM/V1200-4E1 with asterisk

[mohamed kerbachi]

Hello every body.

I have this PCI card : DM/V1200-4E1 

spec in this site:
http://www.intel.com/network/csp/products/3967web.htm

Can i use it with Asterisk, is it compatible ?

Thank you in advance.






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RE: [Asterisk-Users] Plan to free myself from AAH

[Mimmus]

> AMP dialplan is full of garbage and perpaphs is not fully 1.2 
> compatible but it is anyway an Asterisk, working dialplan!

As example:

May 17 18:35:40 WARNING[8625] app_db.c: This application has been
deprecated, please use the ${DB(family/key)} function instead.
May 17 18:35:40 WARNING[8625] app_setcidname.c: SetCIDName is deprecated,
please use Set(CALLERID(name)=value) instead.
May 17 18:35:40 WARNING[8625] pbx.c: SetVar is deprecated, please use Set
instead.
May 17 18:35:40 WARNING[8625] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP
or TOKEN; Input:
May 17 18:35:40 WARNING[8625] ast_expr2.fl: If you have questions, please
refer to doc/README.variables in the asterisk source.
May 17 18:35:49 WARNING[8631] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP
or TOKEN; Input:
May 17 18:35:49 WARNING[8631] ast_expr2.fl: If you have questions, please
refer to doc/README.variables in the asterisk source.
May 18 10:59:08 WARNING[9144] app_groupcount.c: The SetGroup application has
been deprecated, please use the GROUP() function.
May 18 10:59:08 WARNING[9144] app_cut.c: The application Cut is deprecated.
Please use the CUT() function instead.


DV

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Re: [Asterisk-Users] Quad BRI card

[Avi Miller]

On 18/05/2006, at 6:51 PM, Wayne Gemmell wrote:

are a good option (extensive, but come highly recomended from most  
that

I hear).  Good luck and happy hunting.

Ouch, you weren't joking. 1453 Euro!


But worth every penny, imo. I have a few servers running Eicon Diva  
Server V-4BRI cards and they are easy to install, run great with  
Armin's chan_capi-cm and the onboard hardware echo cancellation is  
excellent.


cYa,
Avi


--
National Manager - Special Projects

< Sydney / Melbourne / Canberra / Hobart / London />
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. > > Open Source - Own It - Squiz.net .. />




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[Asterisk-Users] Trunk Si without autetification

[Rabii NOUR]

Hello, 

I am trying to use a trunk SIP between my Asterisk server and a Biling
prepaid server.


Problem: I would like to disable the authentication trunk what is the
command for that request.

In my log server I have: "proxy authentification required"

Regards

Rabii 



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Re: [Asterisk-Users] Quad BRI card

[Michiel van Baak]

On 19:16, Thu 18 May 06, Avi Miller wrote:
> 
> On 18/05/2006, at 6:51 PM, Wayne Gemmell wrote:
> 
> >>are a good option (extensive, but come highly recomended from most  
> >>that
> >>I hear).  Good luck and happy hunting.
> >Ouch, you weren't joking. 1453 Euro!
> 
> But worth every penny, imo. I have a few servers running Eicon Diva  
> Server V-4BRI cards and they are easy to install, run great with  
> Armin's chan_capi-cm and the onboard hardware echo cancellation is  
> excellent.

We use the junghanns.net quadbri cards.
They work great too, and roughly 1/3 of the price.

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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[Asterisk-Users] Please help me...Urgent

[Crazy Boy]

Hi Friends,  Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone.   Now, I want to make calls to US using VoIP Asterisk. I think that there is no need of any external hardware to implement pure VoIP solution. Am I right?  I have registered with Vebtel (VoIP IP Telephony Service provider). They had given me one VoIP Modem called "Voice Finder AP 200" and the below values:  Inbound Number: 123456789 Public IP Number: 55.23.789.145 Password: xyz  (These values are dummy values)  Currently we are making US calls using VoIP provided by "Vebtel". Now, I want to make US calls using this Vebtel service from Asterisk. How can I do this?  I am unable to understand where to give above mentioned values? What configuration files I should use to implement this using the Vebtel SIP provider? Do I need to provide any more values to implement this using
 Asterisk from Vebtel?  Waiting for your quick response. Thank you.    Regards, Chandra.
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Re: [Asterisk-Users] SIP debugging

[Klaus Darilion]

Kevin P. Fleming wrote:

Klaus Darilion wrote:


Shouldn't there be some error indication if Asterisk discards a response?


Probably, although it's not clear here that Asterisk actually discarded
anything. Without seeing the entire dialog, there's no way to be sure
whether there were multiple Call-IDs, multiple tags, etc. involved.


The problem is caused be a forked call with pedantic=yes.

Asterisk --SIP--> Proxy ---SIP> Sipura
  \
   ---> Cisco phone

The SIPURA sends the first 180 Ringing back. Then, Asterisk ignores the 
responses from the Cisco phone (180+200).


When setting pedantic=no, it works (I guess with pedantic=no Asterisk 
does not check the To tag (ugly)).


Is Asterisk not able of handling multiple early dialogs with pedantic=yes?

regards
Klaus

PS: Following the call flows

pedantic=yes:

-- Executing Set("Zap/50-1", "[EMAIL PROTECTED]") 
in new stack

-- Executing GotoIf("Zap/50-1", "0?103:3") in new stack
-- Goto (frompbx,059966366102,3)
-- Executing SetCIDNum("Zap/50-1", "00431234600265") in new stack
-- Executing Dial("Zap/50-1", "SIP/[EMAIL PROTECTED]|90") in 
new stack

-- parse_srv: SRV mapped to host sip.at43.at, port 5060
We're at 213.174.230.213 port 10392
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (NAT) to 83.136.32.160:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport
From: "00431234600265" ;tag=as6ce265a8
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 18 May 2006 09:31:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 293

v=0
o=root 9803 9803 IN IP4 213.174.230.213
s=session
c=IN IP4 213.174.230.213
t=0 0
m=audio 10392 RTP/AVP 8 0 3 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called [EMAIL PROTECTED]
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
From: "00431234600265" ;tag=as6ce265a8
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Server: OpenSer (1.0.0-tls (i386/linux))
Content-Length: 0

--- (8 headers 0 lines)---
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 180 Ringing
t: ;tag=f1d48eba29dc7f4i0
f: "00431234600265" ;tag=as6ce265a8
i: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
Record-Route: 
,

Server: Sipura/SPA2000-3.1.2(NTb)
Contact: 
Content-Length: 0

--- (10 headers 0 lines)---
-- SIP/enum.at43.at-3323 is ringing
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
From: "00431234600265" ;tag=as6ce265a8
To: ;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: [EMAIL PROTECTED]
Date: Thu, 18 May 2006 09:31:25 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: 
Record-Route: 
ontent-Length: 0

--- (11 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'


poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
From: "00431234600265" ;tag=as6ce265a8
To: ;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: [EMAIL PROTECTED]
Date: Thu, 18 May 2006 09:31:36 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: 
Record-Route: 
Content-Type: application/sdp
Content-Length: 196

v=0
o=Cisco-SIPUA 14377 7540 IN IP4 83.136.33.21
s=SIP Call
c=IN IP4 83.136.33.21
t=0 0
m=audio 21174 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (12 headers 9 lines)---
Destroying call '[EMAIL PROTECTED]'
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
From: "00431234600265" ;tag=as6ce265a8
To: ;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: [EMAIL PROTECTED]
Date: Thu, 18 May 2006 09:31:36 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: 
Record-Route: 
Content-Type: application/sdp
Content-Length: 196

v=0
o=Cisco-SIPUA 14377 7540 IN IP4 83.136.33.21
s=SIP Call
c=IN IP4 83.136.33.21
t=0 0
m=audio 21174 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (12 headers 9 lines)---
Destroying call '[EMAIL PROTECTED]'
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
From: "00431234600265" ;tag=as6ce265a8
To: ;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: [EMAIL PROTECTED]
Date: Thu, 18 May 2006 09:31:36 GMT
CSeq: 1

[Asterisk-Users] Re: Ringing indication not working as expected

[Sebastian Kayser]

* Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
> "R" is not a valid Dial option.  

Sure about that? My Asterisk installation lists it as a valid option.

asterisk*CLI> show application Dial
[...]
R- indicate ringing to the calling party when the called
   party indicates
[...]

> "r" is the option you wanted.  HOWEVER, if you are not hearing ringback,
> "r" will almost never fixes the issue.

"r" leads to the ringing being indicated right from the start. So if i
call my cellphone from a SIP-connected snom, ringing is indicated to me
immediately, whereas the cellphone starts to ring not until about 3-5
seconds later. =/

> Make sure you have a /etc/asterisk/indications.conf   In some situations 
> if you do not have that file you will not hear ringback.

Thanks for the advice. indications.conf is now existent and Asterisk is
reloaded but the problem still persists.

/etc/asterisk/indiciations.conf:

[general]
country=de

[de]
description = Germany
ringcadance = 1000,4000
dial = 425
ring = 425/1000,0/4000
busy = 425/480,0/480
congestion = 425/480,0/480
callwaiting = 425/2000,0/6000
dialrecall = 425/500,0/500,425/500,0/500,425/500,0/500,1600/100,0/900
record = 1400/500,0/15000
info = 950/330,0/200,1400/330,0/200,1800/330,0/1000

- Sebastian
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RE: [Asterisk-Users] Re: Ringing indication not working as expected

[Mimmus]

> Thanks for the advice. indications.conf is now existent and 
> Asterisk is reloaded but the problem still persists.
Reloaded? Peraphs restarted is better...

DV

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[Asterisk-Users] SNOM, g722 and 16 kHz audio

[Philipp von Klitzing]

Hi there,

I've been playing with a SNOM 360 and 190 trying to get them talk to each 

other using g722 with 16 kHz. However all I see in the SIP log codec 
negotiation is "g722/8000" which makes me believe that this is only a 8 
kHz link (and that's what it sounds like).

Anyone every managed to establish a 16 kHz wideband call between SNOM 
phones?

Cheers, Philipp


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[Asterisk-Users] Failing SIP registration brings * down

[Philipp von Klitzing]

Hi there,

this is now the second time I've seen an issue like this with 1.2.7.1, 
the first time it was a DNS hickup, today its some Internet congestion: 

When one (!) or more register statements in sip.conf fail the entire 
Asterisk becomes very unresponsive and does not accept registrations from 

local phones. This is bad because I could live with being temporarily 
unreachable thru my SIP carrier, but the local phones should not be 
affected (and therefore still work with BRI and internally).

There's masses of these to be seen (btw DNS and ping work) on the CLI:

chan_sip.c:5322 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' 
timed out, trying again (Attempt #24)

Question: Wasn't there a new sip.conf entry introduced recently that 
limited registration attempts to, say, 5 tries? I can't find any 
reference to this, and am not sure if this made it into 1.2.

Cheers, Philipp

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[Asterisk-Users] Re: Ringing indication not working as expected

[Sebastian Kayser]

* Sebastian Kayser <[EMAIL PROTECTED]> wrote:
> are there any caveats regarding ringing indication with Asterisk?
> 
> I have got an asterisk installation with a quadBRI driven by BRIstuff.
> Internal phones are various snoms (320 / 360) connected via SIP and
> Idefisk softphones connected via IAX2. Outgoing calls are "routed"
> through the Zap interfaces.
> 
> When i set up the action for an external extension as
> 
> Dial(Zap/g2/,60,R)
> 
> or
> 
> Dial(Zap/g2/,60)
> 
> and initiate an outgoing call, Asterisk tells me that the called party
> is ringing (Zap/4-1 is ringing) but there is no ringing indicated to the
> calling party. No matter whether the calling party is a snom hardphone
> or an idefisk softphone.

I tried to narrow down the problem. My installation looks like this


PSTN <-- 3 x BRI --> POTS (NEC) <-- 3 x BRI --> Asterisk
  ^ ^ 
  | |
POTS telephone sets  snom SIP phones

With no options set in the Dial command, i.e. Dial(Zap/g2/,60),
the ringing behaviour is as follows.

- snom -> snom - OK
- snom -> POTS telephone set - OK
- snom -> PSTN - NOK

OK = ringing is signalled to the calling party as soon as Asterisk
indicates it on the console (... is ringing).

NOK = no ringing is signalled to the calling party _although_ Asterisk
indicates it on the console.

So although the Zap interface is used for both types of "external" calls
(snom -> POST, snom -> PSTN) the ringing indication to my snoms fails
for calls to the PSTN.

Any ideas on how to further debug / troubleshoot this behaviour?

- Sebastian
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[Asterisk-Users] DM/V1200-4E1 with asterisk

[mohamed kerbachi]

Hello every body.

I have this PCI card : DM/V1200-4E1 

spec in this site:
http://www.intel.com/network/csp/products/3967web.htm

Can i use it with Asterisk, is it compatible ?

Thank you in advance.






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[Asterisk-Users] Unable to register channel

[Emre BALCI]

Hii all 
I bought te110p card.
I configured zaptel.com ;

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16  and zapata.conf is;

switchtype=euroisdn
signalling=pri_net
group=1
context=nortel
dchannel => 16
channel => 1-15,17-31 I receving the following error
on my gentoo;

May 18 13:48:44 WARNING[8755]: Ignoring canpark
May 18 13:48:44 WARNING[8755]: Ignoring dchannel
May 18 13:48:44 ERROR[8755]: Channel 24 is reserved
for D-channel.
May 18 13:48:44 ERROR[8755]: Unable to register
channel '1-15'
May 18 13:48:44 WARNING[8755]: chan_zap.so:
load_module failed, returning -1
May 18 13:48:44 WARNING[8755]: Loading module
chan_zap.so failed! 
lsmod output is;
wcte11xp   25120  0
zaptel182820  1 wcte11xp
crc_ccitt   2176  1 zaptel
and red led is not light up on card


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[Asterisk-Users] Re: Ringing indication not working as expected

[Sebastian Kayser]

* Sebastian Kayser <[EMAIL PROTECTED]> wrote:
> * Sebastian Kayser <[EMAIL PROTECTED]> wrote:
> > are there any caveats regarding ringing indication with Asterisk?
> PSTN <-- 3 x BRI --> POTS (NEC) <-- 3 x BRI --> Asterisk
>   ^ ^ 
>   | |
> POTS telephone sets  snom SIP phones
> 
> With no options set in the Dial command, i.e. Dial(Zap/g2/,60),
> the ringing behaviour is as follows.
> 
> - snom -> snom - OK
> - snom -> POTS telephone set - OK
> - snom -> PSTN - NOK
> 
> OK = ringing is signalled to the calling party as soon as Asterisk
> indicates it on the console (... is ringing).
> 
> NOK = no ringing is signalled to the calling party _although_ Asterisk
> indicates it on the console.

I gave "brig debug span 1" a try and it led to the following obvious
difference (sorry for the long lines).

snom -> POTS telephone set:
1 < Message type: ALERTING (1)
1 < [1 181  011  891 ]
1 < Channel ID (len= 3) [ Ext: 1  IntID: Implicit, Other Spare: 0, Exclusive 
Dchan: 0
1  PSTN telephone:
1 < Message type: ALERTING (1)
1 < [1 181  011  891 ]
1 < Channel ID (len= 3) [ Ext: 1  IntID: Implicit, Other Spare: 0, Exclusive 
Dchan: 0
1  PSTN
telephone calls?

- Sebastian
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[Asterisk-Users] tdm21B in china

[LIU.ANDY]

Hello Gentelmen,
 
I am in china, just ordered a tdm21B card (2 fxs and 1 fxo), still waiting for its delivery.
 Whether anybody already tried this kind of card  here in china,  will you please tell me how it works, any issue?
esp.
1)    caller id
2)   dtmf
 3)  busy tone
  4) hang up
Thanks a lot!
 
Regards ÄúµÄÅóÓÑkamnpapÕýÔÚÏíÓÃ21CNµÄ10GÓÊÏ䣬ÄúÒ²¸Ï½ô¼ÓÈ룬ÏíÊÜ»ý·Ö´øÀ´µÄÎÞÏÞ¾ªÏ²°É¡£Óá°Vgo ²¥°É¡± ¿´Ãâ·ÑµçÓ°¡¢µçÊÓ¡¢ÌåÓýÖ±²¥!£¨µã»÷ÏÂÔØ£©

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Re: [Asterisk-Users] Please help me...Urgent

[Benchev]

>  Thank you for your quick response. I have successfully implemented
> Intercom (Dialling within my office LAN) using Asterisk. To implement this,
> I am using X-Lite Softphone.
>
>  Now, I want to make calls to US using VoIP Asterisk. I think that there is
> no need of any external hardware to implement pure VoIP solution. Am I
> right?
>
>  I have registered with Vebtel (VoIP IP Telephony Service provider). They
> had given me one VoIP Modem called "Voice Finder AP 200" and the below
> values:
>
>  Inbound Number: 123456789
>  Public IP Number: 55.23.789.145
>  Password: xyz
>
>  (These values are dummy values)
>
>  Currently we are making US calls using VoIP provided by "Vebtel". Now, I
> want to make US calls using this Vebtel service from Asterisk. How can I do
> this?
>
>  I am unable to understand where to give above mentioned values? What
> configuration files I should use to implement this using the Vebtel SIP
> provider? Do I need to provide any more values to implement this using
> Asterisk from Vebtel?
>
>  Waiting for your quick response. Thank you.
Hi,
You have sent this 10 times and received at least 20 answers
but there is no development in you query! 

Some people have email filters for Urgent. The only
"Urgent" in your case is that you urgently need to 
go to the wiki and have a long reading.

Please, show some efforts, benevolence or something... 

Benchev
P.S.
http://www.voip-info.org/wiki/view/Asterisk+Configurations+for+connecting+with+VOIP+providers





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[Asterisk-Users] just softphone

[Ralph Liebessohn]

Hi,I'm trying to start with Asterisk, but I could not put 2 softphones to talk.The asterisk server rejects the connections always when I dial.May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 

192.168.0.106What is necessary to put it to work?There is no need to configure external lines.extensions.conf [internal1]exten => 311000,1,Dial(SIP/teste1)[internal2] 
exten => 312000,1,Dial(SIP/teste2) 
[internal3]exten => 313000,1,Dial(SIP/teste3) 
[teste1]sip.conf[teste1]type=friendusername=teste1secret=123
qualify=yesnat=no host=dynamiccanreinvite=no
context=internal[teste2]type=friendusername=teste2 secret=123
qualify=yesnat=nohost=dynamiccanreinvite=no
context=internal2[teste3]type=friendusername=teste3secret=123
qualify=yesnat=nohost=dynamiccanreinvite=no
context=internal3-- Ralph LiebessohnICQ: 74835911Skype: liebessohn
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[Asterisk-Users] Re: Ringing indication not working as expected

[Sebastian Kayser]

* Mimmus <[EMAIL PROTECTED]> wrote:
> > Thanks for the advice. indications.conf is now existent and 
> > Asterisk is reloaded but the problem still persists.
> Reloaded? Peraphs restarted is better...

The reload messages informed about the re-reading of indications.conf.
However, even restart doesn't change anything about the ringing
indication problem.

See my other reply for further debug information i have gathered.

- Sebastian
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Re: [Asterisk-Users] Unable to register channel

[Leo Ann Boon]

Emre BALCI wrote:




May 18 13:48:44 ERROR[8755]: Channel 24 is reserved
for D-channel.

did you change the jumper setting to E1 as per 
http://www.digium.com/en/docs/TE110P/te110p_config.php?


Looks like the card thinks its a T1 card.

Leo

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Re: [Asterisk-Users] Unable to register channel

[Emre BALCI]

Yes I changed jumper settings but I receiving
following error;

May 18 14:43:43 WARNING[8481]: Ignoring canpark
May 18 14:43:43 WARNING[8481]: Ignoring dchannel
May 18 14:43:43 WARNING[8481]: Unable to specify
channel 1: No such device or ad
dress
May 18 14:43:43 ERROR[8481]: Unable to open channel 1:
No such device or address
here = 0, tmp->channel = 1, channel = 1
May 18 14:43:43 ERROR[8481]: Unable to register
channel '1-15'
May 18 14:43:43 WARNING[8481]: chan_zap.so:
load_module failed, returning -1
May 18 14:43:43 WARNING[8481]: Loading module
chan_zap.so failed!

--- Leo Ann Boon <[EMAIL PROTECTED]> wrote:

> Emre BALCI wrote:
> 
> >
> >
> >May 18 13:48:44 ERROR[8755]: Channel 24 is reserved
> >for D-channel.
> >
> did you change the jumper setting to E1 as per 
>
http://www.digium.com/en/docs/TE110P/te110p_config.php?
> 
> Looks like the card thinks its a T1 card.
> 
> Leo
> 
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Re: [Asterisk-Users] AAH not getting IP address, likely to be network card?

[Bruce Reeves]

Does your AAH box have a static IP address or is it a DHCP client? Run ifconfig to check the IP address on the card.  On 5/17/06, Brian McCarey <
[EMAIL PROTECTED]> wrote:






Hi all,
 
We use AAH to run our office telecoms 
registered with two Sipgate accounts.
 
Today, Sipgate appeared to have had problems with 
their server with oneway audio on every call. In order to cause the Sipgate 
message service to pick up in stead of our AAH box, I simply unplugged the 
network cable.
 
We now have problems where AAH does not seem to 
access the network. I plugged the network cable back in and rebooted 
AAH.
 
AAH boots up, I log in as Root and AAH does not 
give me an IP address. I've used different cables. Everything else can access 
the network.
 
Network card in the AAH box lights up 
green!
 
Before I naff around changing the network card, as 
anyone got any useful thoughts. I think when I pulled out the cable, the card 
went on the blink..!
 
AAH has been running faultlessly 
before..!
 
Should have left the bloody cable at 
start.
 
Regards
 
Brian.

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http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks
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Re: [Asterisk-Users] Re: Bristuffed Asterisk: Hangup problems

[Jeroen Zwarts]

Looks like you're on the right track with this.
I have just diffed the 'i' version with the 'p' version and found the same
line as you found below.

I've changed the 'peercallstate' line in the 'p' version with the
'ourcallstate' one and compiled.

I tested the outbound and inbound dialling over BRI, and * hangs up when it
needs to!
I will have to test some more to see if this little patch doesn't break
anything else, but so far so good.

Thanks for the tip!

Jeroen


> We had the exact same problem. It started happening for us starting at
> the 'k' release of bristuff (i mailed a msg on it in february i think
> to junghanns).
>
> So, the 'i' release worked fine, while 'k' has the problem as described.
>
> A quick diff of 'i' vs. 'k' showed me this (among other things):
>
> diff -U0 -r -x '*.o' -x '*.so'
> bristuff-0.3.0-PRE-1i/libpri-1.2.2/q931.c
> bristuff-0.3.0-PRE-1k/libpri-1.2.2/q931.c
> --- bristuff-0.3.0-PRE-1i/libpri-1.2.2/q931.c   2006-05-17
> 19:54:51.0 +0200
> +++ bristuff-0.3.0-PRE-1k/libpri-1.2.2/q931.c   2006-05-17
> 20:04:07.0 +0200
> @@ -4428,3 +4428,3 @@
> -   if (c->ourcallstate != c->sugcallstate) {
> -   pri_error(pri, "updating callstate, ourcallstate
> %d to %d\n", c->ourcallstate, c->sugcallstate);
> -   c->ourcallstate = c->sugcallstate;
> +   if (c->peercallstate != c->sugcallstate) {
> +   pri_error(pri, "updating callstate, peercallstate
> %d to %d\n", c->peercallstate, c->sugcallstate);
> +   c->peercallstate = c->sugcallstate;
>
> This was such a close match, that i reversed that change in the 'k'
> release and voila! problem disappeared.
> Now, i have no clue what kind of side-effects this has, if any, nor if
> this is the proper solution, but it made the problem disappear for us.
>
> I haven't tried to apply the same to later bristuff releases (all
> releases up to 'p' give us the same hangup problem)
>
> Hope this helps.
>
> marcel
>
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Re: [Asterisk-Users] small form factor WAS soekris hadware

[Darrick Hartman]

Jonathan Gonzalez wrote:

I'm thinking buy only de SBC and look for another chassis where all
equipment fit fine. I say you don't cover, or this i think, because i
need to use FXO<->PBX<->FXS/SIP and you don't use this setup.


Don't attempt to use a Digium or other FXO and FXS card on a soekris 
board, especially if you plan on using a full-featured distribution like 
CentOS (which is what AAH uses).  See more below...



You call forward all calls over an ip-to-pstn account while i want to
receive incoming calls via pstn and call from internal sip phones to
sip phones and to pstn trhu asterisk (of course, paying for the call,
like a normal call).

Another difference is that you use CF with the SBC, while i want to
use the mini-hard disk option and make a complete installation of
[EMAIL PROTECTED]


If you are going to use a hard drive AND an FXS/FXO card (assume 
TDM11B), you'd be happier using an EPIA board.  If you need dual 
ethernet ports, go with the PD1. If not, the V1 would work fine 
as well.



It seems the product is quite stable and i couldn't see this about
astlinux in any place. I would appreciate your thougts about astlinux
and some recomendations will be welcomed.


For information about AstLinux, go to http://www.astlinux.org  Note that 
there have been some pretty substantial changes in the past month and 
some of the documentation hasn't quite caught up to the 0.4.0 image.


Darrick

--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
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Re: [Asterisk-Users] AAH not getting IP address, likely to be network card?

[Justin Biggs]

 
Before I naff around changing the network card, as  anyone got any useful thoughts. I think when I pulled out the cable, the card  went on the blink..!
 Run ifconfig as root and see what that tells you about your eth connections.-- Justin BiggsOwner, Biggs Computer Consulting
[EMAIL PROTECTED]740.501.4781 
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[Asterisk-Users] Re: DISA & SPA3000 issues

[Dave Hawkes]

Thanks for the feedback, but the route that that I'm finding doesn't 
work is:


Asterisk -> SPA3000 -> ZAP/BRI -> Asterisk -> DISA

The problem appears to be on outbound calls from the SPA3000 where the 
second dial tone seems to stop audio transmission, changing the DTMF 
method make no difference. :(


Thanks
Dave Hawkes


Philippe Lindheimer wrote:

Just tried it on mine, worked fine:
 
Cellphone Call -> POTS -> SPA3000 -> Asterisk -> DISA -> Telasip
 
As an FYI, I have my SPA3000 setup with INFO for the DTMF. When I 
originally installed it, I couldn't get the DTMF digits to work coming 
in using AUTO, which is why I have it using INFO (needs to be set on 
both the SPA and in Asterisk).
 
I'm running the SPA300 with Software Version: 2.0.13(GWg), Hardware 
Version 2.0.1(4e16).
 
philippe


 



From: Dave Hawkes <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Date: Wed, 17 May 2006 13:44:43 -0400
Subject: [Asterisk-Users] Re: DISA & SPA3000 issues

I have this exact same issue with the SPA3000, I'm assuming it
must be a
SPA3000 bug?

Dave Hawkes

Alchaemist wrote:
 > Hi,
 >
 > These days I run into something quite odd.
 > I have an [EMAIL PROTECTED] that was modified to meet our 
requirements.
 > We have a completely funtional DISA which we use pretty much
all the
 > time.
 > I works flawlessly with incomming SIP calls from several
providers,
 > IAX calls from FWD and with ZAP.
 >
 > Recently we came out with a situation where it doesn't
work... with
 > a SPA3000 PSTN Line.
 > You can call, navigate de IVR, log in into our app, and then
when
 > you go to DISA, and DISA plays the dialtone... whatever you
dial is not
 > recognized...
 >
 > This was REALLY odd... so I made a network capture with
Ethereal,
 > and... the SPA actually STOPS sending the RTP Events after
the second
 > dialtone...
 >
 > To verify this, I created an IVR which played the dialtone, and
 > verified that it was true no RTP DTMF events (RFC2833)
are sent after
 > the SPA listens the second dialtone.
 >
 > I just reviewed the 87 pages PDF of the SPA3000... and didn't
find
 > anything about such "feature".
 > Now I am going to try to figure out if it has something to do
with
 > the tones recognition of the SPA.
 > I the meanwhile I had to write a little DISA-like app, based on
 > something I found on this forum, without the dialtone.
 >
 > Did anyone find out anything about this issue before?
 >
 > REGARDS!!!
 > Alchaemist
 >
 >
 >
 >
 >
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 >


Feel free to call! Free PC-to-PC calls. Low rates on PC-to-Phone. Get 
Yahoo! Messenger with Voice 
 






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Re: [Asterisk-Users] just softphone

[Benchev]

> I'm trying to start with Asterisk, but I could not put 2 softphones to
> talk. The asterisk server rejects the connections always when I dial.
>
> May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106
>
> What is necessary to put it to work?
> There is no need to configure external lines.
>
> extensions.conf
>
> [internal1]
> exten => 311000,1,Dial(SIP/teste1)
>
> [internal2]
> exten => 312000,1,Dial(SIP/teste2)
>
> [internal3]
> exten => 313000,1,Dial(SIP/teste3)
> [teste1]
>
>
> sip.conf
>
> [teste1]
> type=friend
> username=teste1
> secret=123
>
> qualify=yes
> nat=no
> host=dynamic
> canreinvite=no
> context=internal
>
> [teste2]
> type=friend
> username=teste2
> secret=123
>
> qualify=yes
> nat=no
> host=dynamic
> canreinvite=no
> context=internal2
>
> [teste3]
> type=friend
> username=teste3
> secret=123
>
> qualify=yes
> nat=no
> host=dynamic
> canreinvite=no
> context=internal3
Debug/verbose is too short, but 
probably your peers cannot "meet" in
a mutual context.

Try:
extensions.conf
[default]
include => internal
[internal]
exten => 311000,1,Dial(SIP/teste1)
exten => 311000,2,Hangup ; Hangup is good

exten => 312000,1,Dial(SIP/teste2)
exten => 312000,2,Hangup

exten => 313000,1,Dial(SIP/teste3)
exten => 313000,2,Hangup

Put context=internal or default in all your sip "friends".

Hope that would do.

Benchev
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[Asterisk-Users] multiple calls using IAX

[Ever Zalazar]

Hi. I have a aplication for web, when u press on 
the link, the application log into an asterisk(user, password), and call to one 
extension(ex 201). How can I do to that call go to 201, if busy, go to 202, and 
so on? I want to implement in a call center.
 
 
Best Regards
 
 
Thanks
 
 
Ever
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Re: [Asterisk-Users] small form factor WAS soekris hadware

[Andrew Kohlsmith]

On Thursday 18 May 2006 08:07, Darrick Hartman wrote:
> Don't attempt to use a Digium or other FXO and FXS card on a soekris
> board, especially if you plan on using a full-featured distribution like
> CentOS (which is what AAH uses).  See more below...

I read below, but you don't explain why not to use an FXS/FXO card with a 
Sokeris board if I'm planning on using a full-featured distro.  You only say 
that the EPIA systems are "better."

Can you elaborate?

-A.
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[Asterisk-Users] ACD Light on Phone?

[Matt]

Hi,
Is there anyway I can make a softkey light on a sip phone (aastra
9133i) light up when the agent is logged into a queue?  Even if I have
to do it via some call in a dialplan.  I guess the question is more...
what command do I need to send to a sip phone to turn a light on a
softkey on/off?
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Re: [Asterisk-Users] just softphone

[Stefan Märkle]

> I'm trying to start with Asterisk, but I could not put 2 
> softphones to talk.
> The asterisk server rejects the connections always when I dial.
> 
> May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 
> 192.168.0.106

Try puting a 
permit=0.0.0.0/0.0.0.0
In the sip.conf for your two phones.


BTW: your extensions.conf looks silly, you'll only be able to call test3 from 
test3.
Busy most of the time ;-)

Stefan Märkle

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Re: [Asterisk-Users] Quad BRI card

[Kevin P. Fleming]

stoffell wrote:

> Aside from being available.. What driver does it use?
> Will it be needing bristuff ? (that wouldn't work I guess)

The Digium B410P will use the mISDN stack and chan_misdn for Asterisk.

> Or will the near future integrate BRI ( and hfc?) drivers in asterisk?
> And thus, making bristuff obsolete? (wich means, BRI users will be
> able to use cvs easily..)

No, that will not happen, unless the authors of those drivers want to
disclaim them for inclusion into Zaptel and Asterisk.

> Just to make clear I'm very curious on this card. And yes I'm in europe ;)

As another poster mentioned, the B410P card is definitely targeted at
the non-US market... not because the card would not work here, but
because there is very little availability of BRI lines in the US at all.
Most telcos don't even know what they are if you ask :-)
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Re: [Asterisk-Users] SIP debugging

[Kevin P. Fleming]

Klaus Darilion wrote:

> Is Asterisk not able of handling multiple early dialogs with pedantic=yes?

Asterisk is not capable of handling multiple dialogs in response to an
outbound INVITE at all. The code is not prepared for requests that it
sends to be forked by a proxy.

The next major version of chan_sip (to be worked on during the next
development cycle) will probably be able to handle this, but today, it's
not expected to work properly.
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Re: [Asterisk-Users] Slackware 10.2

[Dave Fullerton]

Fernando Lujan wrote:

Hi guys, I'm trying to use asterisk with my slackware 10.2 box.
Kernel 2.6.13 from the testing...

The udevd are not creating the /dev/zap devices.

Someone already have success installing asterisk over slackware?


Thanks in advance.
Fernando Lujan


I also use asterisk on slackware with the 2.6 kernel. You need to make 
sure you have the udev rules and permissions files from zaptel installed 
and also make sure that rc.hotplug is set executable (or at the very 
least that "/sbin/hotplug" is set in /proc/sys/kernel/hotplug).


I also have slackbuild scripts I made up for both asterisk and zaptel if 
anyone wants them.


-Dave
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[Asterisk-Users] Default dialplan??

[Aaron Paxson]

Hey all!
 
I've got my Asterisk box tied into my PBX.  
Currently, if a call comes into my PBX, and can't find the extension, it 
forwards it through my Asterisk trunk to Asterisk.
 
This works great!
 
Is there a special dialplan function (or common 
usage pattern) that can do the same thing in Asterisk?  i.e. If it can't 
find the extension, send it out Zap/g1?
 
My dialplan works with patterns, but patterns isn't 
what I need here.  Is anyone doing anything like this?
 
Thanks!
~~Aaron
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Re: [Asterisk-Users] just softphone

[Ralph Liebessohn]

On 5/18/06, Benchev <[EMAIL PROTECTED]> wrote:
> I'm trying to start with Asterisk, but I could not put 2 softphones to> talk. The asterisk server rejects the connections always when I dial.>> May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 
192.168.0.106>> What is necessary to put it to work?> There is no need to configure external lines.>> extensions.conf>> [internal1]> exten => 311000,1,Dial(SIP/teste1)
>> [internal2]> exten => 312000,1,Dial(SIP/teste2)>> [internal3]> exten => 313000,1,Dial(SIP/teste3)> [teste1]>>> sip.conf>> [teste1]
> type=friend> username=teste1> secret=123>> qualify=yes> nat=no> host=dynamic> canreinvite=no> context=internal>> [teste2]> type=friend
> username=teste2> secret=123>> qualify=yes> nat=no> host=dynamic> canreinvite=no> context=internal2>> [teste3]> type=friend> username=teste3
> secret=123>> qualify=yes> nat=no> host=dynamic> canreinvite=no> context=internal3Debug/verbose is too short, butprobably your peers cannot "meet" in
a mutual context.Try:extensions.conf[default]include => internal[internal]exten => 311000,1,Dial(SIP/teste1)exten => 311000,2,Hangup ; Hangup is goodexten => 312000,1,Dial(SIP/teste2)
exten => 312000,2,Hangupexten => 313000,1,Dial(SIP/teste3)exten => 313000,2,HangupPut context=internal or default in all your sip "friends".Hope that would do.Benchev
Benchev,thanks for the attention.But didn't solve the problem. I think it is something with access.I set debug and verbose to 10 and got this.extensions.conf ( I've changed internal by from-sip)
[default]include => from-sipinclude => demo[from-sip]exten => 9222,1,Dial(SIP/9222,25)exten => 9222,2,Hangupexten => 9223,1,Dial(SIP/9223,25)exten => 9223,2,Hangup
exten => 31200,1,Dial(SIP/312000,25)exten => 31200,2,Hangupsip.conf[general]context=defaultport=5060    bindaddr=0.0.0.0    
;srvlookup=yes[9222]type=friendcallerid = "Nome - 9222" <9222>username=9222secret=9222host= dynamiccontext=from-sipdtmfmode=rfc2833nat=yescanreinvite=nocontext=internal
[9223]type=friendcallerid = "Nome - 9223" <9223>username=9223secret=9223host= dynamiccontext=from-sipdtmfmode=rfc2833nat=yescanreinvite=nocontext=internal
[312000]type=friendusername=312000secret=312000host=dynamiccontext=from-sipdtmfmode=rfc2833nat=yescanreinvite=nocontext=internalWhen I try to connect From local machine the softphone only call itself:
May 18 10:07:25 DEBUG[2218]: Allocating new SIP call for [EMAIL PROTECTED].1.73May 18 10:07:25 VERBOSE[2218]: -- Registered SIP '9222' at 
192.168.1.73 port 5061 expires 1800May 18 10:07:25 VERBOSE[2218]: -- Saved useragent "X-Lite release 1105d" for peer 9222May 18 10:07:40 DEBUG[2218]: Auto destroying call '
[EMAIL PROTECTED]'When I call From network I got the error Call ended: unknown:May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subcl
ass: NEWMay 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW   Timestamp: 3ms  SCall: 07747  DCall: 0 [192.168.0.106:4569
]May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW   Timestamp: 3ms  SCall: 07747  DCall: 0 [192.168.0.106:4569
]   VERSION : 2May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW   Timestamp: 3ms  SCall: 07747  DCall: 0 [
192.168.0.106:4569]   VERSION : 2   CALLING NUMBER  : 312000May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subcl   VERSION : 2   CALLING NUMBER  : 312000
   CALLING NAME    : 312000May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW   Timestamp: 3ms  SCall: 07747  DCall: 0 [
192.168.0.106:4569]   VERSION : 2   CALLING NUMBER  : 312000   CALLING NAME    : 312000   FORMAT  : 2May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subcl
ass: NEW   Timestamp: 3ms  SCall: 07747  DCall: 0 [192.168.0.106:4569]   VERSION : 2   CALLING NUMBER  : 312000   CALLING NAME    : 312000   FORMAT  : 2
   CAPABILITY  : 1550May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW   Timestamp: 3ms  SCall: 07747  DCall: 0 [
192.168.0.106:4569]   VERSION : 2   CALLING NUMBER  : 312000   CALLING NAME    : 312000   FORMAT  : 2   CAPABILITY  : 1550   USERNAME    : 312000   Timestamp: 3ms  SCall: 07747  DCall: 0 [
192.168.0.106:4569]May 18 10:10:14 VERBOSE[2213]: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW   CALLING NAME    : 312000   FORMAT  : 2
   CAPABILITY  : 1550   USERNAME    : 312000   CALLED NUMBER   : 9222   DNID    : 9222Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT   Timestamp: 00010ms  SCall: 1  DCall: 07747 [
19

[Asterisk-Users] Asterisk - SPA-3000, 407 error

[Neil Cherry]

I recently lost my setup (bad drive) and I'm now trying to get my setup 
back. I have Asterisk setup to a BT100, a Cisco 7960 (7.2 SIP) and an 
SPA-3000. I can call the phone extension, I can call from the phone on 
the SPA to other extensions and I can call out to the PSTN. What I can't 
do is to call from the PSTN to through the SPA to Asterisk. It rings 
twice then I get fast busy. I have the SPA wait for the caller ID info 
(the reason it rings twice). What I've got is as follows:


Code:

registry=>pstn:[EMAIL PROTECTED]:5061

[pstn_in]
  username  = pstn
  secret= pstn
  type  = user
  host  = spa.uucp
  port  = 5061
  context   = from-pstn
  mailbox   = 2202
  nat   = never
  dtmfmode  = rfc2833
  canreinvite   = yes
  qualify   = yes
  insecure  = very
  disallow  = all; need disallow before we can allow
  allow = ulaw
;
[pstn_out]
  username  = pstn
  secret= pstn
  type  = peer
  host  = spa.uucp
  port  = 5061
  context   = to-pstn
  from_user   = pstn
  nat   = never
  dtmfmode  = rfc2833
  canreinvite   = yes
  qualify   = yes
  insecure  = yes
  disallow  = all; need disallow before we can allow
  allow = ulaw
;


My dial plan on the SPA looks like this:

(S0<:[EMAIL PROTECTED]>)

What I see on the Asterisk console is :

May 17 23:36:29 NOTICE[11306]: chan_sip.c:10326 handle_request_invite: 
Failed to authenticate user xxx 
;tag=blahblahblah


In sniffer I traces I see a 407 (Proxy Authentication Required). I've 
also noticed a 401 (Unauthorized) for the phone on the SPA. What am I 
configuring incorrectly.


I'm using Asterisk 1.2.7, previously I was using 1.2.0.


--
Linux Home Automation Neil Cherry   [EMAIL PROTECTED]
http://www.linuxha.com/ Main site
http://linuxha.blogspot.com/My HA Blog
http://home.comcast.net/~ncherry/   Backup site
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Re: [Asterisk-Users] soekris hadware

[Michael Graves]

You will have problems. Physically the card won't fit into the Soekris
case. There isn't sufficient power to provide ring signalling to FXS
ports, which is why the Digium TDM400 card has a dedicated power
connector. 

Of course the Net4801 will only transcode a couple of calls at a
time...which means that you should really stick with G.711 coding
calls.

Then there is the old problem that most of the small FXO adapters/cards
are not very good. Search the mailing list archives and you will find
numerous threads discussing porblems with various FXO cards. 

The Sangoma A200 card would be your best shot as it seems to be able to
cope with various motherboards. Otherwise, you should consider a
freestanding multi-port FXO/FXS adapter like those from Audio Codes or
Mediatrix.

Since you want a more fully featured installation based upon AAH you
may be better served by more powerful hardware.

Michael


On Thu, 18 May 2006 10:31:29 +0200, Jonathan Gonzalez wrote:

>
>Hi Michael,
>
>your document is very good. In fact this was one of the first i read.
>I googled looking for soekris and asterisk and you appreared.
>
>Anyway, your document do not cover the same setup i have. You point
>the problem of the digium cards don't using it, while i have or i
>think i require one of this card with a soekris.
>
>I'm thinking buy only de SBC and look for another chassis where all
>equipment fit fine. I say you don't cover, or this i think, because i
>need to use FXO<->PBX<->FXS/SIP and you don't use this setup.
>
>You call forward all calls over an ip-to-pstn account while i want to
>receive incoming calls via pstn and call from internal sip phones to
>sip phones and to pstn trhu asterisk (of course, paying for the call,
>like a normal call).
>
>Another difference is that you use CF with the SBC, while i want to
>use the mini-hard disk option and make a complete installation of
>[EMAIL PROTECTED]
>
>It seems the product is quite stable and i couldn't see this about
>astlinux in any place. I would appreciate your thougts about astlinux
>and some recomendations will be welcomed.
>
>Thanks for you tu answer and for your magnificent document.
>
>Kind regards,
>
>Jonathan GF
>
>On 5/18/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>> Also...an article I wrote earlier this year
>>
>> http://www.tomsnetworking.com/2006/01/13/how_to_asterisk_pbx/
>>
>> Michael Graves
>> Sr Product Specialist
>> Pixel Power Inc
>> [EMAIL PROTECTED]
>> o(713) 861-4005
>> o(800) 905-6412
>> f(713) 864-8668
>> c(713) 201-1262
>>
>>
>>
>> >  Original Message 
>> > Subject: Re: [Asterisk-Users] soekris hadware
>> > From: "olivier.taylor" <[EMAIL PROTECTED]>
>> > Date: Wed, May 17, 2006 11:12 am
>> > To: Asterisk Users Mailing List - Non-Commercial Discussion
>> > 
>> >
>> >more kindly :
>> >
>> >  http://www.astlinux.org/
>> >
>> >  Olivier
>> >
>> >  Christopher Snell a écrit : Google and voip-info.org will have answers to 
>> > all of your questions.
>> >
>> >
>> > On 5/17/06, Jonathan Gonzalez < [EMAIL PROTECTED]> wrote: Hi group,
>> >
>> >  i'm brand new and i would like to ask about soekris hardware. I read
>> >  along the web but i have some doubts that i think can be solved here.
>> >  My question are the following:
>> >
>> >  [...]
>> >
>> >
>> >
>> >  ___
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>> >  http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> > -
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>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>>
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>>
>
>
>-- 
>si secretum tibi sit, tege illud, vel revela
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>

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

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245


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Re: [Asterisk-Users] Default dialplan??

[Cosmin Prund]

I'll give this one a try, but don't trust me, test it yourself :-)

Of course Asterisk can do it! All you need to do is set up a rule for 
matching ALL extensions in the PBX in it's own separate context and 
include that context into your "normal" context. In the following 
example, asterisk will try matching all extensions in context Normal 
(all extensions defined on *) and, if no match was found, start 
searching the context "secondary_pbx". In my sample this secondary 
context will match any 3-digit number and send it to the other PBX. 
Should work...


[Normal]
include => secondary_pbx
exten => 101,1,Dial(sip/101)

[secondary_pbx]
exten => _XXX,Dial(Zap/g1)

Aaron Paxson wrote:

Hey all!
 
I've got my Asterisk box tied into my PBX.  Currently, if a call comes 
into my PBX, and can't find the extension, it forwards it through my 
Asterisk trunk to Asterisk.
 
This works great!
 
Is there a special dialplan function (or common usage pattern) that 
can do the same thing in Asterisk?  i.e. If it can't find the 
extension, send it out Zap/g1?
 
My dialplan works with patterns, but patterns isn't what I need here.  
Is anyone doing anything like this?
 
Thanks!

~~Aaron


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Re: [Asterisk-Users] Re: Bristuffed Asterisk: Hangup problems

[stoffell]

On 5/17/06, Marcel van der Boom <[EMAIL PROTECTED]> wrote:

We had the exact same problem. It started happening for us starting at
the 'k' release of bristuff (i mailed a msg on it in february i think
to junghanns).


Marcel, thanks.  This does seem to work indeed! I just tested this on
our bristuff 0.3.0-pre1p, works perfect. Thanks a lot!

I will also forward your mail to junghanns support.

cheers!
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[Asterisk-Users] Home asterisk system with single PSTN Line

[Barrass Kevin]

Hi

Im new to asterisk and want to setup a small system at home to play
with.

Can anyone advise a good card I can use so the asterisk box Im building
can act as a gateway to PSTN using my single home analogue phone line.

Kind Regards

Kev
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[Asterisk-Users] SIP re-invite and billing

[Mojo Jojo]

I know this may sound like a stupid question but I will put on my flame 
retardant suit and ask anyway.


Is there any way to use/allow SIP reinvite and still track the length of the 
call?


I realize that the whole idea of reinvite is that it takes the proxy out of 
the media path which, from what I understand also kills the proxy's ability 
to track the start/end time of the call for billing purposes.


Are there any really smart guys out there with propeller hats that have come 
up with a way to get the best of both worlds?


Do we lose anything else using reinvite with Asterisk?

Thanks in advance for any help..

--Mojo


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Re: [Asterisk-Users] SNOM, g722 and 16 kHz audio

[Steve Underwood]

Philipp von Klitzing wrote:


Hi there,

I've been playing with a SNOM 360 and 190 trying to get them talk to each 

other using g722 with 16 kHz. However all I see in the SIP log codec 
negotiation is "g722/8000" which makes me believe that this is only a 8 
kHz link (and that's what it sounds like).


Anyone every managed to establish a 16 kHz wideband call between SNOM 
phones?


Cheers, Philipp
 

G.722 is always a 16000 samples/second codec. There is no 8000 
samples/second option for it.


Steve

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RE: [Asterisk-Users] Home asterisk system with single PSTN Line

[Cory Andrews]

Low end server with a free PCI slot, Asterisk @ Home, Digium TDM11B (1FXO /
1FXS)

Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice - 800.398.VoIP X3402
fax - 716.630.1548
e - [EMAIL PROTECTED]
m - 716.907.4059
aim - B2Cory

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Barrass Kevin
Sent: Thursday, May 18, 2006 9:53 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Home asterisk system with single PSTN Line


Hi

Im new to asterisk and want to setup a small system at home to play
with.

Can anyone advise a good card I can use so the asterisk box Im building
can act as a gateway to PSTN using my single home analogue phone line.

Kind Regards

Kev
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RE: [Asterisk-Users] Quad BRI card

[Guido Hecken]

> On Thursday 18 May 2006 03:35, Mark Coccimiglio wrote:
> >  Otherwise the Diva server cards
> > are a good option (extensive, but come highly recomended from most that
> > I hear).  Good luck and happy hunting.
> Ouch, you weren't joking. 1453 Euro!

What about the Gerdes Primux Cards. They can be used in NT and TE Mode.
Price ~ 670.- EURO
We have a 2S0 card running on a customer site with chan_capi-cm and all
looks good.
Have a look at http://www.primuxisdn.de

Perhaps it helps...

Regards 

Guido
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Re: [Asterisk-Users] SIP re-invite and billing

[Kevin P. Fleming]

Mojo Jojo wrote:

> Is there any way to use/allow SIP reinvite and still track the length of
> the call?

This is discussed nearly every week on this list, it's well covered on
the wiki, and various other places. Have you tried to research this
before asking here?

The simple answer: you are mistaken. SIP reinvites for the media path
have no effect on billing at all.
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Re: [Asterisk-Users] small form factor WAS soekris hadware

[Darrick Hartman]

Andrew Kohlsmith wrote:

On Thursday 18 May 2006 08:07, Darrick Hartman wrote:
  

Don't attempt to use a Digium or other FXO and FXS card on a soekris
board, especially if you plan on using a full-featured distribution like
CentOS (which is what AAH uses).  See more below...



I read below, but you don't explain why not to use an FXS/FXO card with a 
Sokeris board if I'm planning on using a full-featured distro.  You only say 
that the EPIA systems are "better."


Can you elaborate?
  

Yes.  There are several reasons, at least for his situation.

First, the case provided with the Soekris is too small to house a Digium 
TDM card.
Second, there is no easy way to provide the additional power needed for 
FXS modules.
Third, the board is really underpowered.  For about the same money you 
can get an EPIA PD 1, 256 or 512MB of ram, a compact case and have a 
system that is much more capable.


I've been using the EPIA PD as my firewall and office Asterisk system 
for about a year.  I currently have Slackware 10.2 installed with a 
Digium TDM11b card, Asterisk 1.2.7.1, iaxmodem and hylafax.  It also has 
OpenVPN running.  It just plain works.  I use the g729 codec for my voip 
calls.  ulaw internally, so the box does some transcoding. 


Darrick

--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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[Asterisk-Users] Applet to test VoIP quality

[Mindaugas Kezys]

Hello,

 

Does anybody know of free Java/ActiveX applet which could be
placed into web and configured to check ping/latency/jitter/ports to selected
server?

 

Something like http://www.testyourvoip.com

 

 



Regards/Pagarbiai,

Mindaugas Kezys

 



 






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RE: [Asterisk-Users] New To Asterisk - Advice needed

[Forrest Beck]

Give http://www.asteriskguru.com/tutorials
a try.

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Adams
Sent: Wednesday, May 17, 2006 5:19
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] New To
Asterisk - Advice needed



 

Hi
People, 

I’m
writing to get some advice on where to start when learning asterisk? I was
going to begin learning with AAH but I wanted to find out if there is a certain
build to avoid or if there is a Gui/front end that is better then another. I have
been working with dialogic cards for the past 5 years and with auto dialers but
I want to get into providing voip service, support and eventually help people
save money with their phone systems. At the moment it is strictly for education
but I really get a kick out of voip and telephone functions in general. 

Thanks
in advance 

 

- 
Mark
Adams 

 


 
  
   
  
 
 
  
  
  
 


 






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RE: [Asterisk-Users] Providers using Embedded Devices

[Douglas Garstang]

Egads. That'a shame, because from experience I can tell you that trying to make 
Asterisk work in a 'hosted' manner is really tricky. Asterisk wasn't designed 
with multiple companies in mind.

> -Original Message-
> From: Damon Estep [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, May 17, 2006 11:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Providers using Embedded Devices
> 
> 
> Most of the VoIP service providers I have encountered are moving in a
> different direction, with a goal of NOT having any customer premise
> equipment other than the SIP hard phones, soft phones, and ATAs, along
> with an IP access router with QoS.
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] 
> [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Douglas Garstang
> > Sent: Wednesday, May 17, 2006 10:55 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] Providers using Embedded Devices
> > 
> > Just curious...
> > 
> > Does anyone know if any companies using Asterisk on 
> embedded hardware
> (out
> > at the customer premisis), such as the Soekris Net4801, to provide
> VOIP
> > service?
> > 
> > Doug.
> > 
> > 
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Re: [Asterisk-Users] Quad BRI card

[Craig Guy]

From the picture on the web site it looks like it uses a cologne chipset. 

Any idea if these cards will be available in Australia?

Craig

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


Sent: Thursday, May 18, 2006 9:15 PM
Subject: Re: [Asterisk-Users] Quad BRI card



stoffell wrote:


Aside from being available.. What driver does it use?
Will it be needing bristuff ? (that wouldn't work I guess)


The Digium B410P will use the mISDN stack and chan_misdn for Asterisk.


Or will the near future integrate BRI ( and hfc?) drivers in asterisk?
And thus, making bristuff obsolete? (wich means, BRI users will be
able to use cvs easily..)


No, that will not happen, unless the authors of those drivers want to
disclaim them for inclusion into Zaptel and Asterisk.

Just to make clear I'm very curious on this card. And yes I'm in europe 
;)


As another poster mentioned, the B410P card is definitely targeted at
the non-US market... not because the card would not work here, but
because there is very little availability of BRI lines in the US at all.
Most telcos don't even know what they are if you ask :-)
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Re: [Asterisk-Users] Default dialplan??

[Aaron Paxson]

Thanks Cosmin!!

I didn't realize that the dialplans ran in sequential order.  I'll try that. 
thanks!


--Aaron

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


Sent: Thursday, May 18, 2006 9:44 AM
Subject: Re: [Asterisk-Users] Default dialplan??



I'll give this one a try, but don't trust me, test it yourself :-)

Of course Asterisk can do it! All you need to do is set up a rule for 
matching ALL extensions in the PBX in it's own separate context and 
include that context into your "normal" context. In the following example, 
asterisk will try matching all extensions in context Normal (all 
extensions defined on *) and, if no match was found, start searching the 
context "secondary_pbx". In my sample this secondary context will match 
any 3-digit number and send it to the other PBX. Should work...


[Normal]
include => secondary_pbx
exten => 101,1,Dial(sip/101)

[secondary_pbx]
exten => _XXX,Dial(Zap/g1)

Aaron Paxson wrote:

Hey all!
 I've got my Asterisk box tied into my PBX.  Currently, if a call comes 
into my PBX, and can't find the extension, it forwards it through my 
Asterisk trunk to Asterisk.

 This works great!
 Is there a special dialplan function (or common usage pattern) that can 
do the same thing in Asterisk?  i.e. If it can't find the extension, send 
it out Zap/g1?
 My dialplan works with patterns, but patterns isn't what I need here. 
Is anyone doing anything like this?

 Thanks!
~~Aaron


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Re: [Asterisk-Users] Quad BRI card

[Michiel van Baak]

On 22:32, Thu 18 May 06, Craig Guy wrote:
> >From the picture on the web site it looks like it uses a cologne chipset. 
> Any idea if these cards will be available in Australia?

Can't you just order them from the digium website?
Or is digium not shiping to Australia?

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [Asterisk-Users] just softphone

[Benchev]

On Thursday 18 May 2006 16:32, Ralph Liebessohn wrote:
> On 5/18/06, Benchev <[EMAIL PROTECTED]> wrote:
> > > I'm trying to start with Asterisk, but I could not put 2 softphones to
> > > talk. The asterisk server rejects the connections always when I dial.
> > >
> > > May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from
> >
> > Try:
> > extensions.conf
> > [default]
> > include => internal
> > [internal]
> > exten => 311000,1,Dial(SIP/teste1)
> > exten => 311000,2,Hangup ; Hangup is good
> >
> > exten => 312000,1,Dial(SIP/teste2)
> > exten => 312000,2,Hangup
> >
> > exten => 313000,1,Dial(SIP/teste3)
> > exten => 313000,2,Hangup
> >
> > Put context=internal or default in all your sip "friends".

> extensions.conf ( I've changed internal by from-sip)
>
> [default]
> include => from-sip
> include => demo
>
> [from-sip]
> exten => 9222,1,Dial(SIP/9222,25)
> exten => 9222,2,Hangup
>
> exten => 9223,1,Dial(SIP/9223,25)
> exten => 9223,2,Hangup
>
> exten => 31200,1,Dial(SIP/312000,25)
> exten => 31200,2,Hangup
>
>
> sip.conf
>
> [general]
> context=default
> port=5060
> bindaddr=0.0.0.0
> ;srvlookup=yes
>
>
> [9222]
> type=friend
> callerid = "Nome - 9222" <9222>
> username=9222
> secret=9222
> host= dynamic
> context=from-sip
> dtmfmode=rfc2833
> nat=yes
> canreinvite=nocontext=internal
Ralph,
> canreinvite=nocontext=internal
Is the above a copy/paste mistake?
Try context=from-sip or default as in you case that's
declared in extensions.conf. Delete context=internal.

On  the other hand, this thing X-light, does it require
Use Auth ID or something similar? Try to turn it on/off.

Benchev
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Re: [Asterisk-Users] Quad BRI card

[Kevin P. Fleming]

Craig Guy wrote:
>> From the picture on the web site it looks like it uses a cologne chipset. 
> Any idea if these cards will be available in Australia?

(Please to trim your replies and not reply in the middle of quoted text...)

The cards will be available through all normal Digium distribution
channels. Availability in specific countries will depend on
certification and approval processes in those countries, though. We will
even allow US users to buy them, although I have no idea what they will
do with them :-)
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[Asterisk-Users] Polycom 601 -- programming buttons.

[Ken D'Ambrosio]

Hi, all.  I want to have a button on my receptionist's 601 that, when
pressed, will forward her current call to a given extension.  Is there any
way to do that?  I've tried to RTFM, but I'm coming up empty.

Thanks,

-Ken D'Ambrosio

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Re: [Asterisk-Users] FW: NuFone Update: DIDs

[Wilson Pickett]

On 4/18/06, Wes Baehr <[EMAIL PROTECTED]> wrote:

Well this is disappointing. Time to find somebody else...
From: NuFone Operations
Sent: Tuesday, April 18, 2006 3:44 PM
Subject: NuFone Update: DIDs
Effective 3pm EST Today, April 18th, 2006 Telesthetic, the carrier
snip


I received an email from Nufone today regarding a change users need to
make to bring their DID back to life. I have no idea how long the
process will take, but it's an encouraging sign.
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Re: [Asterisk-Users] Default dialplan??

[Eric \"ManxPower\" Wieling]

Aaron Paxson wrote:

Thanks Cosmin!!

I didn't realize that the dialplans ran in sequential order.  I'll try 
that. thanks!


We originally had dialplans run in random order, but people found it too 
confusing.


--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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[Asterisk-Users] Polycom - missed calls dial back

[Bill Gibbs]

This is not necessarily Asterisk specific but if I have
Polycom 301/501 and 601s and want to dial a missed call back, how do I prepend
a 9 – can I do this via the polycom config?  I can’t find anything
in the docs.

 

Bill






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[Asterisk-Users] sip and other phones

[Emre BALCI]

I am trying to call phone that on meridian My asterisk
connected to meridian via E1.but failed
I think I have routing problem between sip and E1
because I am debuging sip call then I see call going
to [EMAIL PROTECTED]
I just want sip client able to call phones that on
meridian and phones that on meridian able to call sip
clients 
linux*CLI> pri show span 1
Primary D-channel: 16
Status: Provisioned, Up, Active
Switchtype: EuroISDN
Type: Network
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T313 Timer: 4000
N200 Counter: 3
linux*CLI> zap show status
Description  Alarms
IRQbpviol CRC4

Digium Wildcard TE110P T1/E1 Card 0  OK 0 
0  0


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RE: [Asterisk-Users] Polycom - missed calls dial back

[Watkins, Bradley]

I have not found a way to do this via the Polycom 
configs.  However, what I do is just ensure that the callerid of an inbound 
call is set so that the recorded number on the Polycoms is a valid callback 
number (i.e., prepend '9' or '91' depending on the inbound 
CallerID).
 
Regards,
- Brad


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Bill 
GibbsSent: Thursday, May 18, 2006 11:37 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] Polycom - missed calls dial back


This is not necessarily Asterisk 
specific but if I have Polycom 301/501 and 601s and want to dial a missed call 
back, how do I prepend a 9 – can I do this via the polycom config?  I can’t 
find anything in the docs.
 
BillThe contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. 
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Re: [Asterisk-Users] Quad BRI card

[Craig Guy]

Any device to legally connect to the PSTN in Australia must be approved by 
the regulatory body.  A process that usually costs at least $20,000 and only 
allows the permit holder to sell the product for conneciton to the pstn.  It 
is a very high barrier to entry for the Australian market.  There is a guy 
in Victoria who certified the Fritz! card and charges $400 each for them. 
Paralell imports are not allowed to be connected.


Some manufacturers do the right thing by certifying the card themselves 
(Eicon for example).  Other manufacturers such as AVM leave it to the 
distributor to certify the card for the local market.


The difference is that I can buy an Eicon card off eBay from the US or 
Europe and legally connect it to the PSTN in Australia as the card comes 
from the factory carrying the regulartory approval mark.  If i was to buy 
AVM, Digium or Sangoma from another country I'm out of luck cause it doesn't 
carry the approval sticker that the Australian distributor puts on it.


I can understand both points of view - as a customer I want a competitive 
market so I get value for money.  As a distributor I want an exclusive 
territory so I can jack up the prices to whatever the market will bear 
without being undercut by nasty competition.


Craig

- Original Message - 
From: "Michiel van Baak" <[EMAIL PROTECTED]>

To: 
Sent: Thursday, May 18, 2006 10:45 PM
Subject: Re: [Asterisk-Users] Quad BRI card



On 22:32, Thu 18 May 06, Craig Guy wrote:
>From the picture on the web site it looks like it uses a cologne 
>chipset.

Any idea if these cards will be available in Australia?


Can't you just order them from the digium website?
Or is digium not shiping to Australia?

--
Michiel van Baak
http://michiel.vanbaak.info
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"Why is it drug addicts and computer afficionados are both called users?"

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Re: [Asterisk-Users] Re: Bristuffed Asterisk: Hangup problems

[Marcel van der Boom]

On 18 mei 2006, at 14:05, Jeroen Zwarts wrote:

I tested the outbound and inbound dialling over BRI, and * hangs up  
when it

needs to!
I will have to test some more to see if this little patch doesn't  
break

anything else, but so far so good.


I've done a bit more testing and in our install the patch seems to  
cause an issue with the 'hangup' (h) extensions. We use this to  
convert incoming faxes to pdf and send them off through mail after  
the sending fax machine hangs up.  The hangup extension is never  
reached so that bit of our dialplan didnt work anymore.


Since both patched and unpatched dont work with that particular  
setup, there's no way (i know) to test out wether this is actually  
caused by the patch or not, but i thought i'd just mention it.


The 'regular' dialplan seems to work fine for us too though. No other  
issues were seen by us at least.


marcel
--
Marcel van der Boom
HS-Development BV   --   http://www.hsdev.com
So! webapplicatie framework  --   http://make-it-so.info




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[Asterisk-Users] OT: Aastra Powertouch 350 caller id

[Dan Elder]

Hi all, is anyone using the Aastra Powertouch 350 analog (adsi) phone with
asterisk? I cannot get the phone to display incoming caller id... I can see
the CID if I hook up a cheap caller id enabled phone... Only the PT350 is
having problems. Looking all over for docs on this phone, but what I've
found doesn't have any mention of a setting to display CID...

Thx as always

Dan

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Re: [Asterisk-Users] Default dialplan??

[Philippe Lindheimer]

Aaron,     There are probably plenty of ways to do this, off the top of my head, if you add a 'include => go-to-pbx' context within the context where your Asterisk patterns are, and there is no match, Asterisk will then begin to check the 'include' contexts in order. (It does not even look at them if it can find a match in the current context. So ... put such an include with a 'catch all' dial plan within 'go-to-pbx' that will handle any patterns that don't match and send them off to your alternate pbx.     philippe  From: "Aaron Paxson" <[EMAIL PROTECTED]>To: Date: Thu, 18 May 2006 09:26:50 -0400Subject: [Asterisk-Users] Default dialplan??Hey all!     I've got my Asterisk box tied into my PBX.  Currently, if a call comes into my PBX, and can't find the extension, it forwards it through my Asterisk trunk to Asterisk.     This works great!     Is there a special dialplan function (or common usage pattern) that can do the same thing in Asterisk?  i.e. If it can't find the extension, send it out Zap/g1?     My dialplan works with patterns, but patterns isn't what I need here.  Is anyone doing anything like this?     Thanks!  ~~Aaron
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Re: [Asterisk-Users] New To Asterisk - Advice needed

[Cosmin Prund]

I'm fairly new to Asterisk myself and I also started with AAH. 
Unfortunately I had to remove all configuration files generated by 
FreePBX (the GUI of AAH) and started over using http://voip-info.org as 
my guide. Configuration files generated with FreePBX make use of 
advanced functionality available in Asterisk and that in turn makes it 
hard (impossible?) to read for a newby. If you've got some experience 
with Linux and it's kind of configuration files you might be better of 
without AAH. On the other hand I'm in the process of re-installing my 
Asterisk on a fresh Centos 4.3 installation so I can't comment on how 
difficult it is for a newby to install everything from sources. Hope 
I'll be able to manage it :)


Mark Adams wrote:


Hi People,

I’m writing to get some advice on where to start when learning 
asterisk? I was going to begin learning with AAH but I wanted to find 
out if there is a certain build to avoid or if there is a Gui/front 
end that is better then another. I have been working with dialogic 
cards for the past 5 years and with auto dialers but I want to get 
into providing voip service, support and eventually help people save 
money with their phone systems. At the moment it is strictly for 
education but I really get a kick out of voip and telephone functions 
in general.


Thanks in advance

- Mark Adams




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Re: [Asterisk-Users] Re: Ringing indication not working as expected

[Dinesh Nair]

On 05/18/06 18:45 Sebastian Kayser said the following:

So although the Zap interface is used for both types of "external" calls
(snom -> POST, snom -> PSTN) the ringing indication to my snoms fails
for calls to the PSTN.


we've got the following:

E1 PRI --- Asterisk ---+--- FXS Gateway --- Analog Phones
   |
   +--- FXS Gateway --- Analog Phones
   |
   +--- FXS Gateway --- Analog Phones
   |
   +--- FXS Gateway --- Analog Phones

we see the same problem for /some/ of the phones on a single fxs gateway, 
but not for the /other/ phones on the /same/ gateway. i'd always thought 
that this may have been caused by a lost SIP 180 Ringing packet between 
asterisk and the fxs gateway, but perhaps now i may look inside asterisk 
itself to try to see if it's causing it somewhere.


--
Regards,   /\_/\   "All dogs go to heaven."
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[Asterisk-Users] Auto Dial Out Madness

[Jon Scottorn]

Hi All,

    I have been struggling with the auto dial out in asterisk.  I am trying to get a call to be auto dialed and play back a message once the line is answered.  So far I have been unsuccessful.  
Currently what happens is I have my .call file.  I mv it into /var/spool/asterisk/outgoing.  The call is initiated and that all works, my problem is that it does not wait for the line to be answered before playing the message back.  It immediately after dialing the number begins playing the message and is done playing it before the person even answers the phone.
Does anyone know what I can do to get this working.

I want to be able to launch a script from cron that will create the .call file and mv it into the outgoing directory.  That is all working currently.  Here is a sample of what my .call file looks like and what my extensions.conf looks like.

.call

Channel: Zap/3/1234567890
Callerid: 1234567890
MaxRetries: 200
RetryTime: 30
WaitTime: 45
Context: outboundmsg1
Extension: s
Priority: 1

extensions.conf - snip -
[outboundmsg1]
  exten => s,1,AbsoluteTimeout,40   
  exten => s,2,DigitTimeout,5
  exten => s,3,ResponseTimeout,10
  exten => s,4,Answer
  exten => s,5,Wait(1)
  exten => s,6,Playback(outboundmsgs/msg1)   ; "play outbound msg"
  exten => s,7,Background(outboundmsgs/how_to_ack) ; "Press 1 to replay or 2 to acknowledge receiving this message"
  exten => 1,1,Goto(s,5)   ; replay message
  exten => 2,1,Goto(msgack,s,1) ; acknowledge message

  exten => t,1,Playback(vm-goodbye)
  exten => t,2,Hangup

  exten => T,1,Hangup

Thanks in advance for any suggestions.




Jon Scottorn
Systems Administrator
The Possibility Forge, Inc.
http://www.possibilityforge.com
435.635.0591 x.1004





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Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P

[Remco Barende]

On Wed, 17 May 2006, Rodney G. McDuff wrote:


Hi All
Before I go out and buy a DELL PowerEdge 2850 has anyone had
problems (or any other useful experience) getting a TE411P to work with
it. I also have a legacy TE110P. Has anyone had problems with this combo.


I tried using a TE110P and a TE210P. The irq hit percentage on the Dell 
with zttest is quite disappointing, even after tweaking. It's just above 
the minimum requirement


Also the 2850 is *always* sharing IRQ's on every PCI slot, you need to buy 
a dual port ethernet adapter which will use only one irq to free up an IRQ 
on another slot. This just totally sucks and irq sharing in a box with 
only 3 pci slots is totally unnecessary


I'm now using the 2850 with one TE210P, system is in production use since 
the 1st of May and no lockups or apparent problems so far.


I would have expected better from the 2850 however and would not buy one 
again


Sangoma cards are said to be better but I preferred to support Digium and 
I don't want to mess around with additional drivers


Just my $0.02
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Re: [Asterisk-Users] unicall dialing problem

[acriollo]

Hi every body.I was intalled the last version of unicall, but the problem persist.This issue can be due to a grounding problem?regards.2006/5/16, acriollo <
[EMAIL PROTECTED]>:I will try ...Regards.
2006/5/16, Moises Silva <[EMAIL PROTECTED]>:

A recent version of Unicall has a small bug in tone generation (amissmatching use of spandsp library), please try to upgrade to latestspandsp and unicall libraries, im pretty much sure your problem willbe solved that way.
RegardsOn 5/16/06, acriollo <[EMAIL PROTECTED]> wrote:> Hi every body, im triying to resolv thi issue with unicall.
> The asterisk box drop calls some times inbound and outband with no aparently
> reason> When i triying to dial to the outside , the PBX just send a couple of digits> and after this the call waits for a while and then the line is released,> with no call on line.>

> this is the log for one call.> Can some body helpme please.> This is my zaptel and unicall file.>> unicall.conf> loglevel=255>  protocolclass=mfcr2> protocolvariant=mx,10,4
> protocolend=cpe> supertones=mx> group = 1> context=incoming> channel => 1-10>> zaptel.conf> span=1,1,0,cas,hdb3> cas=1-10:1101> dchan=16>

>> Regards and thanks in advanced>> May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2> UniCall/1 Make call> May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2
> UniCall/1 Making a new call with CRN 32792>  May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2> UniCall/1 0001  ->  [1/   1/Idle  /Idle ]> May 16 00:35:06 WARNING[1556]: chan_unicall.c:2672 handle_uc_event:
> Unicall/1 event Dialing> May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2> UniCall/1  <- 1101  [1/  40/Seize /Idle ]> May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2
> UniCall/1 0 on  ->  [2/  40/Group I   /Idle ]> May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2> UniCall/1 Channel gains> May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2
> UniCall/1 Channel switching> May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2> UniCall/1 Call control(6)> May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2
> UniCall/1 Drop call(cause=Normal Clearing [16])> May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2> UniCall/1 Clearing fwd> May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2
> UniCall/1 1001  ->  [2/  40/Group I   /DNIS ]> May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2> UniCall/1 0 off ->  [1/  40/Group I   /Idle ]
> May 16 00:35:43 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2> UniCall/1  <- 1001  [1/  40/Clear fwd B   /Idle ]> May 16 00:35:43 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2
> UniCall/1 Destroying call with CRN 32792> May 16 00:35:43 WARNING[1556]: chan_unicall.c:2672 handle_uc_event:> Unicall/1 event Release call> May 16 00:35:43 WARNING[1556]: chan_unicall.c:612 unicall_report: MFC/R2
> UniCall/1 Channel echo cancel>> ___> --Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] Home asterisk system with single PSTN Line

[Weidong Shao]

I am new to this also. But it is hard to figure out what can be done through a single PSTN line. maybe you can buy the SDK card and then setup IVR menu for have separate voice mailboxes. Weidong
On 5/18/06, Barrass Kevin <[EMAIL PROTECTED]> wrote:
HiIm new to asterisk and want to setup a small system at home to playwith.Can anyone advise a good card I can use so the asterisk box Im buildingcan act as a gateway to PSTN using my single home analogue phone line.
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[Asterisk-Users] Powertouch 350 CallID display continued

[Dan Elder]

Hi All, as a followup to my previous posting (re: not getting caller id to
display on a powertouch 350), I've found the following... 

If I hook the PT350 up to the PSTN line, caller id is diplayed properly...
If I connect it to our CAS Acess Bank II, the 350 will not display incoming
CID... If I hook up a cheap callerid enabled phone to the same port that the
PT350 was on (on the AB II), the CID IS displayed on the cheap phone..so it
seems that something is happening with the connection of the PT350 to
asterisk. Anyone used one of these handsets successfully with CID?

Thanks as always

Dan

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Re: [Asterisk-Users] Slackware 10.2

[Fernando Lujan]

Jonathan Feally wrote:
I believe I had to do the udev permissions file and also cause udevd 
to launch at bootup before modprobe'ing zaptel stuff. Check to make 
sure that udevd is launching automatically on bootup and that the udev 
rules and permissions are in place.



Thanks all. I have this working with the 2.4.31 kernel. I hope the next 
version have a best version of udev. :)


Fernando Lujan
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[Asterisk-Users] R2/MFC Configuration.

[Fernando Lujan]

I'm trying to put asterisk working with a proprietary pbx system.

I'm doing it using a T1 crossover cable. The pbx system uses the R2/MFC 
specification. And the don't inform if it uses cas, ccs, ami or hbd3.


My digium card is flashing a red light.

How can I put this working with the R2/MFC system?

Thanks.

Fernando Lujan
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Re: [Asterisk-Users] Auto Dial Out Madness

[Doug Lytle]

Jon Scottorn wrote:

Hi All,

I have been struggling with the auto dial out in asterisk.  I am 
trying to get a call to be auto dialed and play back a message once 
the line is answered.  So far I have been unsuccessful. 
Currently what happens is I have my .call file.  I mv it into 
/var/spool/asterisk/outgoing.  The call is initiated and that all 
works, my problem is that it does not wait for the line to be answered 
before playing the message back.  It immediately after dialing the 
number begins playing the message and is done playing it before the 
person even answers the phone.


You can't.  You'll need to setup a loop that asks the end user to press 
a key to play your recording.  I have mine loop 6 times with a timeout 
value of 6 seconds, before giving up.  Code follows:


[voice-mail-callback]

; 
; Set timeouts
; 

exten => s,1,Set(TIMEOUT(response)=6)
exten => s,2,Set(TIMEOUT(digit)=3)
exten => s,3,Wait(5)
exten => s,4,Set(COUNT=0)

; ***
; Play, you attention is required, press 1 to
; collect voice mail
; ***

exten => s,5,Background(attention-required)
exten => s,6,Background(press-1)
exten => s,7,Background(to-collect-voicemail)

; *
; If 1 is pressed, then play transfer and
; then jump to voice-mail context.
; *

exten => 1,1,Playback(pbx-transfer)
exten => 1,2,Goto(voice-mail,s,1)

; 
; Setup a variable to count the number of
; times the message has been played, when
; $COUNT reaches > 5, play you've taken
; to long to dial and hangup.
; 

exten => t,1,Set(COUNT=$[${COUNT} + 1])
exten => t,2,NoOP(${COUNT})
exten => t,3,GotoIf($[ ${COUNT} > 5 ]?103)
exten => t,4,Goto(voice-mail-callback,s,5)
exten => t,103,Playback(local/tolong-todial)
exten => t,104,Playback(goodbye)
exten => t,105,Hangup()

; 
; Invalid extension plays invalid choice and adds
; 1 to $COUNT
; 

exten => i,1,Playback(local/sorry-invalid-choice)
exten => i,2,Set(COUNT=$[${COUNT} + 1])
exten => i,3,NoOP(${COUNT})
exten => i,4,Goto(voice-mail-callback,s,5)

exten => h,1,NoOP(Hungup)

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RE: [Asterisk-Users] Plan to free myself from AAH

[Mimmus]

Colin was right!
I forgot that AMP dialplan makes intensive use of database keys as AMPUSER
and DEVICE: I understand its logic but migration is postponed after a GREAT
dialplan rewriting :-(


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Mimmus
> Sent: Thursday, May 18, 2006 9:08 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Plan to free myself from AAH
> 
> AMP dialplan is full of garbage and perpaphs is not fully 1.2 
> compatible but it is anyway an Asterisk, working dialplan!
> I already tried to copy config files and Asterisk starts 
> without warnings:
> gradually I will clean out them from fax, queues, devices, 
> ring groups, weather reports, etc 
> 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Colin 
> > Anderson
> > Sent: Wednesday, May 17, 2006 6:29 PM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: RE: [Asterisk-Users] Plan to free myself from AAH
> > 
> > What I did with AMP was take the best parts of it and 
> copy/paste to a 
> > clean extensions.conf, then add my modifications onto it. 
> Worked for 
> > me.
> > 
> > -Original Message-
> > From: Strom Carlson [mailto:[EMAIL PROTECTED]
> > Sent: Wednesday, May 17, 2006 10:21 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Plan to free myself from AAH
> > 
> > 
> > On 5/17/06, Mimmus <[EMAIL PROTECTED]> wrote:
> > 
> > > I was thinking to this plan:
> > > - install another server with Red Hat 4 U3
> > > - install PHP, MySQL and other usefuls stuffs
> > > - download latest version of Asterisk and third parts
> > applications I
> > > use
> > > - compile all
> > > - copy /etc/asterisk from old server to new, change only what is 
> > > needed
> > > - start and try
> > >
> > > Do you think is it OK?
> > 
> > I doubt it.  The problem I have with AAH / AMP / FreePBX is 
> that the 
> > configuration files are absolutely full of useless garbage and are 
> > really not at all suitable for moving to a standard 
> asterisk install.
> > 
> > Set up a new server from scratch and start learning how to 
> configure 
> > asterisk manually.  Rebuild everything one step at a time 
> so that the 
> > functionality remains as you'd like it to be, but that the actual 
> > configs aren't full of that FreePBX garbage :)
> > 
> > --
> > Strom Carlson
> 
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RE: [Asterisk-Users] Polycom - missed calls dial back

[Chad Osmond]

Prepend outgoing calls by using Dial(9{exten}) instead of dialing 9 to 
get out on your system...
Or, add a 9 to caller id.
 
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Bill 
GibbsSent: May 18, 2006 11:37 AMTo: Asterisk Users Mailing 
List - Non-Commercial DiscussionSubject: [Asterisk-Users] Polycom - 
missed calls dial back


This is not necessarily Asterisk 
specific but if I have Polycom 301/501 and 601s and want to dial a missed call 
back, how do I prepend a 9 – can I do this via the polycom config?  I can’t 
find anything in the docs.
 
Bill
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Re: [Asterisk-Users] SPA-1001 behind NAT -> Internet Asterisk box -- BOUNTY!

[Lachek Butalek]

Stupid not-quite-an-answer - if you're willing to pay money for a fix, why not buy an IAX2 compatible FXS to replace the SPA-1001 with? It will traverse NATs without a problem. I'm using a GNet VP168I (same as the PA168V) and it works fine even behind a NAT which is itself behind a corporate firewall...
Just a thought.On 5/17/06, Eric Lyons <[EMAIL PROTECTED]> wrote:
I'm still unable to get my SPA-1001 to work behind NAT with an Asterisk box out on the Internet.  It works fine to my local [EMAIL PROTECTED] box.I've tried... many things.I'm willing to pay a $250 bounty (PayPal preferred) to anyone who can help me get it working.  Any Sipura experts out there?
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[Asterisk-Users] router with qos and compatible with stun

[Laurent Schweizer]

Hello, 

I have a client that needs 20-26 simultaneous voip connections and I don't
want to relay all this traffic. So I m looking for a router with non
symmetric NAT for SDSL. (to use STUN)


Thanks


Laurent


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[Asterisk-Users] E&M and Dial tone

[Bart Fisher]

I'm a bit confused about how to handle this.

I have Asterisk sitting in the middle between a Qwest Long Distance T1 
(Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic 
D/240SC-T1 card.


The Qwest T1 originally was connected to the Dialogic card directly.  The 
signaling was set to E&M Wink Start because Dialogic used this as its 
default settings, so it just worked without fiddling.


Before Asterisk:

Incoming Qwest calls would wink the Dialogic card and then send DTMF to the 
Dialogic after it winked back.
Outgoing calls from Dialogic would come off-hook, wait for wink. At this 
point Qwest would send dial tone.  The Dialogic has

call supervision and "wait for dial tone" enabled.

With the Asterisk in the middle, Incoming from Qwest are directed to 
Dialogic and are answered correctly.
The problem is when the dialogic card wants to dial out, we only get the 
wink Asterisk - no dial tone.  The dialogic reports this as

a failure and hangs-up

If I remove the Dial Tone Detection option on the Dialogic and add pauses 
before the dial string, all works OK.


My question is how can I emulate the Qwest functionality and provide a dial 
tone after the wink?


TIA Bart

My zaptel.conf:

# Span 1: TE4/0/1 "TE410P (PCI) Card 0 Span 1" AMI/D4 RED
span=1,0,0,d4,ami
e&m=1-24

# Span 2: TE4/0/2 "TE410P (PCI) Card 0 Span 2" AMI/D4 RED
span=2,1,0,d4,ami
e&m=25-48

My zapata.conf

; Span 1: TE4/0/1 "TE410P (PCI) Card 0 Span 1"
; This T1 is attached to in-house VM System
;
signalling =>em_w
context=from-internal
group = 1
channel => 1-24

; Span 2: TE4/0/2 "TE410P (PCI) Card 0 Span 2"
; This T1 is attached to Qwest LD
;
signalling =>em_w
context=from-pstn
group = 2
channel => 25-48 




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Re: [Asterisk-Users] Polycom - missed calls dial back

[Andrew Kohlsmith]

On Thursday 18 May 2006 15:01, Chad Osmond wrote:
> Prepend outgoing calls by using Dial(9{exten}) instead of dialing 9 to
> get out on your system...
> Or, add a 9 to caller id.

Yuck.  While technically correct, both of those solutions suck major goat nad.

-A.
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[Asterisk-Users] monitoring sangoma cards via snmp

[Sangoma Techdesk]

When used in TDM mode, the sangoma cards will work under zaptel, so you will
need to perform SNMP at a higher level (i.e in Asterisk). 

David Yat Sin
Sangoma Technologies
(905) 474 1990 x119
(800) 388 2475 x199
MSN: [EMAIL PROTECTED]
Email: [EMAIL PROTECTED]
Wiki: http://sangoma.editme.com
 

> Message: 2
> Date: Fri, 12 May 2006 09:39:55 +0200 (CEST)
> From: <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] monitoring sangoma cards via snmp
> To: asterisk-users@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=iso-8859-1
> 
> Hello,
> 
> Digium does not provide snmp support to monitor their cards !
> 
> Anybody has tried Sangoma product A104 Quad T1/E1 or others ?
> 
> Regards
> harry
> 
> 
> 
> 
> 


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Re: [Asterisk-Users] SPA-1001 behind NAT -> Internet Asterisk box -- BOUNTY!

[Luki]

Stupid not-quite-an-answer - if you're willing to pay money for a fix, why
not buy an IAX2 compatible FXS to replace the SPA-1001 with? It will
traverse NATs without a problem.


Looks like it wasn't a NAT of configuration problem after all... the
SPA devices are quite nice, IMO. If there's a need, I guess Eric can
explain it further...

--Luki
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[Asterisk-Users] Re: Auto Dial Out Madness

[Bruno Wolff III]

On Thu, May 18, 2006 at 10:38:14 -0600,
  Jon Scottorn <[EMAIL PROTECTED]> wrote:
> Hi All,
> 
> I have been struggling with the auto dial out in asterisk.  I am
> trying to get a call to be auto dialed and play back a message once the
> line is answered.  So far I have been unsuccessful.  
> Currently what happens is I have my .call file.  I mv it
> into /var/spool/asterisk/outgoing.  The call is initiated and that all
> works, my problem is that it does not wait for the line to be answered
> before playing the message back.  It immediately after dialing the
> number begins playing the message and is done playing it before the
> person even answers the phone.
> Does anyone know what I can do to get this working.

You didn't provide information about your hardware or where you are located
(different telephone systems use different signalling).

I am having the same problem with a TDM400 and as best I can tell, I have
to wait for improved software to detect when the phone is answered. Digital
hardware should do what you want, and TDM400s may work in different
circumstances than mine.

Depending where you are you may see polarity reversal as a call progress
indicator. There are config settings for that in zapata.conf. There is
also a call progress setting that didn't work for me. Some people claim
that problems can be caused by the phone company not providing a signal
to you (sometimes called "forward disconnect") and that if you talk to them
they may be able to turn it on for you.

There is also an entry on voip-info.org where someone claims to have an
app named nv_linedetect that will do call progress based on inband audio.
The source hasn't been published and I haven't seen any recent references
for anyone using it.

It seems common for people to stick in a wait(2) to give people a chance to
answer the phone. That isn't a real solution, but might make things not
so bad.
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[Asterisk-Users] Pulling the mISDN number from an incoming call

[Wayne Gemmell]

Hi all

Which command do I use to pull the mISDN number from an incoming call.
-- 
Cheers
Wayne
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[Asterisk-Users] VoiceMail Groups

[Forrest Beck]

Has anyone seen good scripts or documentation on Voicemail
groups?  We are looking to have a system where you can send a voicemail to
multiple mailboxes.

 

 






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Re: [Asterisk-Users] R2/MFC Configuration.

[Moises Silva]

Fernando: There are few or no people that will give you an Answer with
that information. The list is usually for people that already have
tried something and is experimienting some kind of specific problem.
Your question seems like "ahhh it does not work, help me!". As far as
I can see you have 2 options. Please search in google for information
about how to configure, try something and then come back with a more
meaningfull question, or 2, hire some Asterisk consultant to make it
work.

Do you know Unicall?

please check http://www.voip-info.org/wiki/index.php?page=Asterisk+MFC+R2

Regards

On 5/18/06, Fernando Lujan <[EMAIL PROTECTED]> wrote:

I'm trying to put asterisk working with a proprietary pbx system.

I'm doing it using a T1 crossover cable. The pbx system uses the R2/MFC
specification. And the don't inform if it uses cas, ccs, ami or hbd3.

My digium card is flashing a red light.

How can I put this working with the R2/MFC system?

Thanks.

Fernando Lujan
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Re: [Asterisk-Users] unicall dialing problem

[Moises Silva]

If you have an extra E1 port, please avoid testing with the telco and
try using E1 crossover cable. If you speak spanish here is a document
I wrote about MFCR2, Asterisk and Avantel. May be usefull for other
telcos as well.

http://phpmexic.u33.0web-hosting.com/wordpress/misc/mfcr2-asterisk-unicall.pdf

Regards

On 5/18/06, acriollo <[EMAIL PROTECTED]> wrote:

Hi every body.
I was intalled the last version of unicall, but the problem persist.

This issue can be due to a grounding problem?

regards.

2006/5/16, acriollo < [EMAIL PROTECTED]>:
>
> I will try ...
>
> Regards.
>
>
> 2006/5/16, Moises Silva <[EMAIL PROTECTED]>:
>
> > A recent version of Unicall has a small bug in tone generation (a
> > missmatching use of spandsp library), please try to upgrade to latest
> > spandsp and unicall libraries, im pretty much sure your problem will
> > be solved that way.
> >
> > Regards
> >
> > On 5/16/06, acriollo <[EMAIL PROTECTED]> wrote:
> > > Hi every body, im triying to resolv thi issue with unicall.
> > > The asterisk box drop calls some times inbound and outband with no
aparently
> > > reason
> > > When i triying to dial to the outside , the PBX just send a couple of
digits
> > > and after this the call waits for a while and then the line is
released,
> > > with no call on line.
> > >
> > > this is the log for one call.
> > > Can some body helpme please.
> > > This is my zaptel and unicall file.
> > >
> > > unicall.conf
> > > loglevel=255
> > >  protocolclass=mfcr2
> > > protocolvariant=mx,10,4
> > > protocolend=cpe
> > > supertones=mx
> > > group = 1
> > > context=incoming
> > > channel => 1-10
> > >
> > > zaptel.conf
> > > span=1,1,0,cas,hdb3
> > > cas=1-10:1101
> > > dchan=16
> > >
> > >
> > > Regards and thanks in advanced
> > >
> > > May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 Make call
> > > May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 Making a new call with CRN 32792
> > >  May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 0001  ->  [1/   1/Idle  /Idle ]
> > > May 16 00:35:06 WARNING[1556]: chan_unicall.c:2672 handle_uc_event:
> > > Unicall/1 event Dialing
> > > May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1  <- 1101  [1/  40/Seize /Idle ]
> > > May 16 00:35:06 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 0 on  ->  [2/  40/Group I   /Idle ]
> > > May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 Channel gains
> > > May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 Channel switching
> > > May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 Call control(6)
> > > May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 Drop call(cause=Normal Clearing [16])
> > > May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 Clearing fwd
> > > May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 1001  ->  [2/  40/Group I   /DNIS ]
> > > May 16 00:35:42 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 0 off ->  [1/  40/Group I   /Idle ]
> > > May 16 00:35:43 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1  <- 1001  [1/  40/Clear fwd B   /Idle ]
> > > May 16 00:35:43 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 Destroying call with CRN 32792
> > > May 16 00:35:43 WARNING[1556]: chan_unicall.c:2672 handle_uc_event:
> > > Unicall/1 event Release call
> > > May 16 00:35:43 WARNING[1556]: chan_unicall.c:612 unicall_report:
MFC/R2
> > > UniCall/1 Channel echo cancel
> > >
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> > >
> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >
> > >
http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > >
> >
> >
> > --
> > "Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org";
> > ___
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> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>


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RE: [Asterisk-Users] just softphone

[Forrest Beck]

You need to either need to put all your
extensions in the same context or use a GoTo command.

 

Try this instead:

extensions.conf 

[internal1]
exten => 311000,1,Dial(SIP/teste1)
exten => 312000,1,Dial(SIP/teste2) 
exten => 313000,1,Dial(SIP/teste3)

 

sip.conf

[teste1]
type=friend
username=teste1
secret=123
qualify=yes
nat=no 
host=dynamic
canreinvite=no

context=internal1

[teste2]
type=friend
username=teste2 
secret=123
qualify=yes
nat=no
host=dynamic
canreinvite=no

context=internal1

[teste3]
type=friend
username=teste3
secret=123
qualify=yes
nat=no
host=dynamic
canreinvite=no

context=internal1

 

or..

 

http://www.asteriskguru.com/tutorials/goto.html

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ralph Liebessohn
Sent: Thursday, May 18, 2006 7:18
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] just
softphone



 





Hi,

I'm trying to start with Asterisk, but I could not put 2 softphones to talk.
The asterisk server rejects the connections always when I dial.










May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106

What is necessary to put it to work?
There is no need to configure external lines.

extensions.conf 

[internal1]
exten => 311000,1,Dial(SIP/teste1)

[internal2] 
exten => 312000,1,Dial(SIP/teste2) 

[internal3]
exten => 313000,1,Dial(SIP/teste3) 





[teste1]


sip.conf

[teste1]
type=friend
username=teste1
secret=123










qualify=yes
nat=no 
host=dynamic
canreinvite=no









context=internal

[teste2]
type=friend
username=teste2 
secret=123










qualify=yes
nat=no
host=dynamic
canreinvite=no









context=internal2

[teste3]
type=friend
username=teste3
secret=123










qualify=yes
nat=no
host=dynamic
canreinvite=no









context=internal3







-- 
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn 






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Re: [Asterisk-Users] Polycom - missed calls dial back

[Lacy Moore - Aspendora]

How do others handle this?  I'm still in the testing phase and didn't even think of this until now.
 
Do you require 9 for outgoing calls?  If not, do you require users to either hit send or # to make the call immediately?  What about long distance numbers?  Do you allow for 10 digit dialing everywhere and add the 1 to the area codes that require it in your dial plan?
 
On 5/18/06, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
On Thursday 18 May 2006 15:01, Chad Osmond wrote:> Prepend outgoing calls by using Dial(9{exten}) instead of dialing 9 to
> get out on your system...> Or, add a 9 to caller id.Yuck.  While technically correct, both of those solutions suck major goat nad.-A.___--Bandwidth and Colocation provided by 
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-- Lacy MooreAspendora, Inc. 
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