Re: [Asterisk-Users] DIAX

[Dan]

Hi Bilal,
Well, what u advise us to use if the bandwidth is about 22kbps (dial up
connection in very old countries)?
Another thing: u have idea if it is working on Microsoft Windows OS? As 
most
of clients here are using Microsoft and not linux.

You can take alook here:
http://www.voip-info.org/wiki-Codecs
to see a codec comparison.
You can try iLBC and Speex on DIAX.
Best regards,
Dan
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Re: [Asterisk-Users] Streaming Audio - Music On Hold Feature

[Justin Richards]

honestly I only did it for testing, and probably less than 30 minutes,
so I don't know how long it would stay active long term, but it worked
great during that test... definitely longer than 30 seconds..  I am
pretty sure I'm running straight v1.0, i have been reluctant to
upgrade because its working just how I want it to..


On Fri, 14 Jan 2005 08:04:57 -0700, Ken Godee <[EMAIL PROTECTED]> wrote:
> Justin Richards wrote:
> 
> > I have not used any M$ products, but it works with shoutcast like this:
> >
> > default => quietmp3:/var/lib/asterisk/mohmp3-empty,http://host.com:8000/
> >
> > basically, create an empty directory to point it to first, then the
> > url to the stream.
> >
> > If the microsoft stream can be played via url in winamp in MP3 format,
> > then it should work about the same.
> 
> Justin,
> 
> How are you keeping the mp3 stream open?
> 
> My mpg client connections are closing after about 30-105 secs.
> The moh/mpg processes remain running and moh works fine
> but they're just looping whatever has been
> previously streamed before connections dropped.
> 
> Is this not happening on your system?
> It is doing this on v1.0.3
>
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Re: [Asterisk-Users] Echo Training - how long

[Rich Adamson]

> I have echo training set on in my zapata.conf file for a X101P card:
> echocancel = yes
> echocancelwhenbridged = yes
> echotraining = yes
> 
> Now, I know that echo cancellation is a black art, but I am finding that
> at the beginning of a call bridged between a SIP channel and a Zap
> channel the voice quality is poor to abysmal for the first few seconds,
> but as the call progresses, esp after about 30 seconds, the call quality
> becomes very acceptable.
> 
> Should echo training take that long?
> Is it, in fact, echo training or some thing else?
> Has any one got any guidance on ET other than what is in the wiki, which
> I find to be very hard to follow?

Try echotraining=800 and see if that makes a difference.



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[Asterisk-Users] DIAX

[Bilal Ghayad]

Dear Dan;

Thanks alot for your kindly reply.

Well, what u advise us to use if the bandwidth is about 22kbps (dial up
connection in very old countries)?

Another thing: u have idea if it is working on Microsoft Windows OS? As most
of clients here are using Microsoft and not linux.

Regards
Bilal

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[Asterisk-Users] Echo Training - how long

[Howard Lowndes]

I have echo training set on in my zapata.conf file for a X101P card:
echocancel = yes
echocancelwhenbridged = yes
echotraining = yes

Now, I know that echo cancellation is a black art, but I am finding that
at the beginning of a call bridged between a SIP channel and a Zap
channel the voice quality is poor to abysmal for the first few seconds,
but as the call progresses, esp after about 30 seconds, the call quality
becomes very acceptable.

Should echo training take that long?
Is it, in fact, echo training or some thing else?
Has any one got any guidance on ET other than what is in the wiki, which
I find to be very hard to follow?


-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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Re: [Asterisk-Users] Re: Grandstream Bugetone 101 & mwi

[Paul Fielding]

Hahawell the MWI is the blinking blue LCD.  The message button
is "reserved for future use"  Hang in there.  There will soon to be some
upgrades and rumor has it that the conferencing feature will soon be
introduced so that conference button on the phone will soon be 
working.
The message button isn't reserved, it works fine, you simply need to 
correctly configure it.   It's job is to dial the voicemail box when 
pressed.   This works as designed.   It just doesn't blink.

Paul


David
On Fri, 14 Jan 2005 10:25:46 -0500, Stephen R. Besch wrote
Ronald Wiplinger wrote:
> I tried to use message waiting indicator, by "Subscribe for MWI" in the
> web menu of the phone.
>
> However, it does not light up / flash, even if a voice mail is waiting.
>
> Where is the switch to turn it to?
>
> bye
>
> Ronald
>
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I don't mean to be rude to everyone who responded to this question,
 but I think that everyone is answering the wrong question. The
point is that the message waiting indicator doesn't light up, at all,
 ever. All that happens when messages are waiting is that the
display blinks and the phone gives a stutter dialtone. That's it.
There is no light under the button - there should be, but there
isn't. The "blinking" phone designers should have put those stupid
blinking red leds - that only flash on boot up - under the message
button and flashed the display during boot up. But they didn't and
we're stuck with it. Such is life.
Stephen R. Besch
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-
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HK Tel: +852 3107-1333
HK Cell: +852 9166-1880
US Tel: +1 313 228-0906
-
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Re: [Asterisk-Users] Remote Voicemail Retrieval...

[Howard Lowndes]

On Sat, 2005-01-15 at 15:03, Philippe Daoust wrote:
> Hello list,
> 
> I want to listen to voicemails on my * box from a phone that is not 
> local to my pbx.  I.e., from my cellphone or my PSTN work line etc.  I'm 
> aware that I can forward VM to email or use a web interface but that is 
> not always practical.
> 
> Other than doing an IVR type arrangement or a phone number dedicated to 
> VM access is there a way to do this?  On my old POTS line I used to be 
> able to call my line and simply punch "*" during unavailable message 
> playback to go to the equivalent of voicemailmain().  Is there a way to 
> do this in *?

Set up voicemailmain in an extension that is part of the context used by
the dial in line and use a Background message so that you can capture
the DTMF for the extension.

> 
> Thanks!
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-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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Re: [Asterisk-Users] DIAX PC to Phone

[Dan]

Hi,
Is DIAX supported for G723 codec and can work on Windows OS?
It supports just: alaw, ulaw, gsm, ilbc and speex.
G723 is not very usual in the Asterisk world.
Best regards,
Dan
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Re: [Asterisk-Users] voice quality with asterisk

[Matt Riddell]

voip technocrat wrote:
hello list ,
Hello!
:-)
my set up is like this
ip device -->ser ---> asterisk(astcc) --> pstn gatewsy
my asterisk version is 1.0.2
Latest stable package is 1.03 with 1.04 to be released very shortly :-)
iam using the ser as registration and asterisk aa the 
prepaid one with the help of the astcc.
now my problem is the destination people 
i.e  the pstn line s are listening low voice 
You need to let us know what devices you are using.
and also the blurr sound quality along with 
the audio of the ip device at the destination side 
of the pstn lines 
As above.
iam useing the open g729 and ulaw and alaw as codecs
No such thing as open g729.  G729 is a patented codec.  If you are using 
it in a nation with a government, chances are that the patent is upheld. 
(People - don't flame on this, we all know everybody's opinions on the 
topics of codec patents)

so what may be problem 

iam not keeping any overheards on the asterisk box
by not ruuning other services
So there is nothing else running on the Asterisk box?
iam giveng bandwidth of more that 512 kbs to asterisk 
With Quality of service?  Are there any packet drops?
so any body can suggest any new things to done to
improve the sound quality
As above, let us know what devices you are using.  I would say that the 
PSTN GW is the likely culprit.  Does it have any settings for 
receive/transmit gain adjustment?

--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] voice quality in asterisk

[Matt Riddell]

VoIP technocrat wrote:
hello list ,
iam using a simple setup as shown below
ip device ---> ser --> asterisk (astcc) --->pstn gatewsy
:-)
Are you having some problems?  Is the voice quality good/bad?
What is your network load?
What is your PC load?
What PSTN Gateway are you using?
What codecs?
Snippet of config files?
What IP device?
Do you have QOS set up on your network?
--
Cheers,
Matt Riddell
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[Asterisk-Users] Asterisk@Home Install Problems

[Jonathan Curd]

I am trying to install [EMAIL PROTECTED] on a Dell 1650
server. The setup cd runs fine and completes with no
errors but when I try to connect to the web site
(http:myip/maint) to edit the configs nothing happens
just a page not found error. 

I'm not sure if there is a problem with my dual on
board nics but when I check ifconfig I see the dhcp
address info that was assigned for eth0 and then
underneath some info about a loop back with the
address 127.0.0.1. (I can ping the address assined
from my other machie but I cannot see the address as
leased in my dhcp server) I can successfully change
the dhcp address to my own config for eth0 using
netconfig but the loop back stuff remains when I use
ifconfig to check the ip. Also I can't ever see the ip
or config the ip on the second ethernet port. In
addition the use of hostnae -i shows an ip of
127.0.0.1. If I use only the ethernet jack 1 and leave
the second jack unplugged same result as above, but if
I use only jack 2and leave jac 1 unplugged I see just
the loopback info and nothing about eth0 with the
defined ip and netconfig does nothing.

Any help getting this up and runnign would be greatly
appreciated.

-Jonathan

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[Asterisk-Users] voice quality with asterisk

[voip technocrat]

hello list ,

my set up is like this


ip device -->ser ---> asterisk(astcc) --> pstn gatewsy

my asterisk version is 1.0.2

iam using the ser as registration and asterisk aa the 

prepaid one with the help of the astcc.

now my problem is the destination people 


i.e  the pstn line s are listening low voice 

and also the blurr sound quality along with 

the audio of the ip device at the destination side 

of the pstn line s 

iam useing the open g729 and ulaw and alaw as codecs

so what may be problem 

iam not keeping any overheards on the asterisk box

by not ruuning other services

iam giveng bandwidth of more that 512 kbs to asterisk 

box

so any body can suggest any new things to done to
improve the sound quality

with regards


Yahoo! India Matrimony: Find your life partner online
Go to: http://yahoo.shaadi.com/india-matrimony
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[Asterisk-Users] voice quality in asterisk

[voip technocrat]

hello list ,

iam using a simple setup as shown below

ip device ---> ser --> asterisk (astcc) --->pstn gatewsy


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Go to: http://yahoo.shaadi.com/india-matrimony
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Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

[Matt Riddell]

Eric Bishop wrote:
I logged a support issue with HP and their response was that it's not
their server that is the problem and if other cards show interrupts
(which they do) there's nothing more they can do
And you told them that this is the only server the cards don't work in?
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] TDM400 answers the line all the time!

[Matt Riddell]

Michael George wrote:
On Tue, Jan 18, 2005 at 04:16:08AM -0600, Justin Carlson wrote:
no i was using line 1 for testing /w fxs module and i never changed it
back
Also, could you show us the contents of your [routing] context in 
extensions.conf?

--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Remote Voicemail Retrieval...

[timebandit001]

> I want to listen to voicemails on my * box from a phone that is not
> local to my pbx.  I.e., from my cellphone or my PSTN work line etc.  I'm
> aware that I can forward VM to email or use a web interface but that is
> not always practical.
> 
> Other than doing an IVR type arrangement or a phone number dedicated to
> VM access is there a way to do this?  On my old POTS line I used to be
> able to call my line and simply punch "*" during unavailable message
> playback to go to the equivalent of voicemailmain().  Is there a way to
> do this in *?

You can include the voicemail extension in your incomig-line context

That way, while you are in the main menu, you could punch 8500 (or
whatever extension is you voicemail)

At least, that's the way I did it

Hope that help
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Re: [Asterisk-Users] Remote Voicemail Retrieval...

[Randy]

In the same context as your Voicemail() dialplan execution add:

exten => a,1,VoicemailMain() ; you can optionally pass it a vm box number in 
the ()
exten => a,2,Playback(goodbye)
exten => a,3,Hangup

Now when you dial in, while your greeting is playing hit '*' and it will
prompt you for a mbox number and password (or just a password if you
pass in a mbox number)

Randy

On Fri, Jan 14, 2005 at 11:03:36PM -0500, Philippe Daoust wrote:
> Hello list,
> 
> I want to listen to voicemails on my * box from a phone that is not 
> local to my pbx.  I.e., from my cellphone or my PSTN work line etc.  I'm 
> aware that I can forward VM to email or use a web interface but that is 
> not always practical.
> 
> Other than doing an IVR type arrangement or a phone number dedicated to 
> VM access is there a way to do this?  On my old POTS line I used to be 
> able to call my line and simply punch "*" during unavailable message 
> playback to go to the equivalent of voicemailmain().  Is there a way to 
> do this in *?
> 
> Thanks!
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Re: [Asterisk-Users] Remote Voicemail Retrieval...

[Steven Critchfield]

On Fri, 2005-01-14 at 23:03 -0500, Philippe Daoust wrote:
> Hello list,
> 
> I want to listen to voicemails on my * box from a phone that is not 
> local to my pbx.  I.e., from my cellphone or my PSTN work line etc.  I'm 
> aware that I can forward VM to email or use a web interface but that is 
> not always practical.
> 
> Other than doing an IVR type arrangement or a phone number dedicated to 
> VM access is there a way to do this?  On my old POTS line I used to be 
> able to call my line and simply punch "*" during unavailable message 
> playback to go to the equivalent of voicemailmain().  Is there a way to 
> do this in *?

Hmm doesn't that example you mention sound an awful lot like an IVR
setup? Essentially, once you answer the line, just make sure there is an
extension that will send you to voicemailmain. 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] REGISTER Problems With Realtime

[Michael Shuler]

It was a very misleading error.  I had the DB name spelled wrong in my
/etc/odbc.ini... You would think it would give a more intuitive error than
that.



Michael Shuler, C.E.O.
BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP)
682 High Point Lane
East Peoria, IL 61611
Office: (217) 585-0357
Cell: (309) 657-6365
Fax: (309) 213-3500
E-Mail: [EMAIL PROTECTED]
Customer Service: (877) 976-0711 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Matthew Boehm
> Sent: Friday, January 14, 2005 1:05 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] REGISTER Problems With Realtime
> 
> 
> I have no idea how ODBC converts a prepared statement to 
> MySQL when only up
> until 4.1 did mysql support them. Have you tried using 
> res_config_mysql
> inside asterisk-addons?
> 
> -Matthew
> 
> - Original Message - 
> From: "Michael Shuler" <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> 
> Sent: Friday, January 14, 2005 12:24 PM
> Subject: RE: [Asterisk-Users] REGISTER Problems With Realtime
> 
> 
> > MySQL but its using the ODBC driver.
> >
> > 
> >
> > Michael Shuler, C.E.O.
> > BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP)
> > 682 High Point Lane
> > East Peoria, IL 61611
> > Office: (217) 585-0357
> > Cell: (309) 657-6365
> > Fax: (309) 213-3500
> > E-Mail: [EMAIL PROTECTED]
> > Customer Service: (877) 976-0711
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf Of
> > > Matthew Boehm
> > > Sent: Friday, January 14, 2005 9:56 AM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] REGISTER Problems With Realtime
> > >
> > >
> > > Its trying to prepare a statement. What database are you using?
> > >
> > > -Matthew
> > >
> > > - Original Message - 
> > > From: "Michael Shuler" <[EMAIL PROTECTED]>
> > > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> > > 
> > > Sent: Thursday, January 13, 2005 7:05 PM
> > > Subject: [Asterisk-Users] REGISTER Problems With Realtime
> > >
> > >
> > > Why is the SELECT statement below putting a "?" in for the
> > > username?  I am
> > > using today's CVS.
> > >
> > > Jan 13 18:48:41 WARNING[7570]: res_config_odbc.c:105
> > > realtime_odbc: SQL
> > > Execute error!
> > > [SELECT * FROM sip_buddies WHERE name = ?]
> > >
> > >
> > >
> > > Full dump:
> > >
> > > Sip read:
> > > REGISTER sip:198.88.216.85 SIP/2.0
> > > Via: SIP/2.0/UDP 
> 192.168.1.2:5060;branch=z9hG4bK-ffmzndpfrao2;rport
> > > From: "Mike's Peoria Snom"
> > > ;tag=ljql3vjulo
> > > To: "Mike's Peoria Snom" 
> > > Call-ID: [EMAIL PROTECTED]
> > > CSeq: 106321 REGISTER
> > > Max-Forwards: 70
> > > Contact: 
> ;q=1.0
> > > User-Agent: snom200-3.56m
> > > P-NAT-Refresh: 15
> > > Supported: gruu
> > > Allow-Events: dialog
> > > X-Real-IP: 192.168.1.2
> > > WWW-Contact: 
> > > WWW-Contact: 
> > > Expires: 60
> > > Content-Length: 0
> > >
> > >
> > > 17 headers, 0 lines
> > > Using latest request as basis request
> > > Sending to 192.168.1.2 : 5060 (NAT)
> > > Jan 13 18:48:41 WARNING[7570]: res_config_odbc.c:105
> > > realtime_odbc: SQL
> > > Execute error!
> > > [SELECT * FROM sip_buddies WHERE name = ?]
> > >
> > > Transmitting (NAT):
> > > SIP/2.0 403 Forbidden
> > > Via: SIP/2.0/UDP
> > > 192.168.1.2:5060;branch=z9hG4bK-ffmzndpfrao2;received=206.222.
> > > 58.98;rport=32
> > > 777
> > > From: "Mike's Peoria Snom"
> > > ;tag=ljql3vjulo
> > > To: "Mike's Peoria Snom"
> > > ;tag=as01156a34
> > > Call-ID: [EMAIL PROTECTED]
> > > CSeq: 106321 REGISTER
> > > User-Agent: Asterisk PBX
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > > Contact: 
> > > Content-Length: 0
> > >
> > >
> > >  to 206.222.58.98:32777
> > > Jan 13 18:48:41 NOTICE[7570]: chan_sip.c:8036 handle_request:
> > > Registration
> > > from '"Mike's Peoria Snom"
> > > ' failed
> > > for '206.222.58.98'
> > > Scheduling destruction of call
> > > '[EMAIL PROTECTED]' in
> > > 15000 ms
> > >
> > > 
> > >
> > > Mike
> > >
> > > ___
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> > > http://lists.digium.com/mailman/listinfo/asterisk-users
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> > >
> >
> > ___
> > Asterisk-Users ma

[Asterisk-Users] Getting started with Asterisk

[E. Wong]

I am interested in learning Asterisk and have DSL (1 static IP) and a
single POTS line at home.  I have an Ethernet LAN running behind a
Linksys router using NAT.  My question is only about the hardware
needed at this point.  The software configuration I will read about
and learn.  So what hardware do I need to do the following?
1) Route incoming calls from the existing POTS line to 8 regular
phones. (Work no differently than without Asterisk)
2) Sign up with a VOIP provider, get a few numbers and route each VOIP
number to a subset of above phones. (e.g., One VoIP number rings 2 or
3 of the 8 phones, first phone to pick up gets the call, or even
better, all 2 or 3 phones can hear the call.  Is this possible?  How
well do fax machines work over VoIP?)
3) Outgoing calls will use VoIP provider if available, POTS line if not.
4) 911 will always use POTS line.
5) I want to use regular phones, not IP or SIP phones.

I should ask first, is what I want to do possible?  I've read the
documents and manuals, but it is still all rather confusing.  This is
the hardware I think I would need in a Linux box running Asterisk,
please correct if I am wrong:
  1) One FXO module to connect Asterisk Linux box with POTS line
  2) A FXS module for each regular phone in my house
  3) Some module for connecting to the VOIP provider (Is this just the
Ethernet NIC?)

Can someone please provide me with the appropriate hardware product
models for what I need?  What is the lowest cost way to do this?  If
things cost too much, it may be too expensive a hobby to indulge in,
at least until I win the lottery.  =)

And some general miscellaneous questions I have about telephony/Asterisk:
1) With the above setup, would I be able to just add SIP phones onto
the Ethernet LAN in the future?
2) With just one POTS line, I wouldn't be able to have multiple POTS
phone numbers, is that correct?
3) How do phone calls come over a T1 line?  I thought T1 is for data. 
Are those strictly for VoIP calls?  Is this where a TE410P is used?

Thanks for your help.
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Re: [Asterisk-Users] Polycom IP 500 Dial Issues

[Andrew Thrift]

As has been mentioned earlier this is to do with the DIGITMAP that is 
configured in the phone1.cfg or whatever you have called it. 

Mine looks like this:   
[129]xxT|0[34679]xxx|[1-4]xxx|02[1-9]xxT|111|12x|195x|*8|8xx|0[58]0[08]xxT

I will explain it bit by bit so you understand it.
NOTE: The Pipe | symbol is the seperator!
[129]xxT Allows users to dial 1,2 or 9 followed by 2 digits, the 
phone will then wait for a second (this is what the T does) and then 
dial.  It waits incase the number being dialed is actually longer.

0[34679]xxx In New Zealand our toll numbers start with 0 
followed by and area number 3,4,6,7 or 9 this is what this allows, so if 
someone dials 06-355- the polycom will dial immediately.

[1-4]xxx   All extensions are 4 digits long and start with 
numbers in the range of 1-4 followed by 3 digits, this allows these to 
be dialled.

02[1-9]xxT  Mobile phones in NZ start with 02 followed by either 
1,5,7 or 9 but I just included the whole range from 1-9.  The timer (T) 
is used as the digits can either be 6 long for contract mobiles, or 7 
for pre-pay mobiles.

111Is the emergency services number in NZ and dials straight 
away

12xThe telco's service number all start with 12 followed by 
a single digit e..g 7

195x Telco numbers for testing lines, e.g. 1957 repeats your 
phone number back to you.

*8Allows staff to pickup calls using asterisk *8 feature
8xx  Allows local numbers
0[58]0[08]xxT   Allows toll free numbers such as 
0508-000999 or 0800-000999 to be dialed, the timer allows to longer 
combinations.


I hope this has explained it to you more, if not download the admin 
guide as I did and figure it out yourself.

Best of luck,

Andrew
Andrei (MPI) wrote:
Greg Boehnlein wrote:
Hello,
I have a mixture of Polycom SP IP 500 and 300 phones. I have 
been reading through the administration manual to try and solve 
this problem, but I do not seem to be able to find the answers to 
my question. I figured I would ask here and see if anyone has some 
suggestions.

The problem is kind of annoying. After dialing 4 digits, the phone 
seems to pause and miss the 5th digit, often requiring the user to 
re-dial the 5th digit several times.

I'm not sure if this is some sort of "Early Dial" feauture trying 
to match on a 4 figit extension, but I would like any help that 
people can provide.


It's because of the dialplan configuration in phone MAC-config file. 
Easiest way is to get default conf file that came with your firmware 
and replace it, modifying settings as needed.

Andrei
  

Andrei,
I am using the default file that came with the firmware version 
1.3.1. I've only made very slight modifications to it, most of which 
is specific to the phonexxx.cfg file specifically for registration, 
MWI and line issues. I'll dig deeper into this tonight when I get a 
chance.

 

You may want to request firmware 1.3.4 from your Polycom dealer. 
Usually they would send it to you with no problem. (You probably know 
that Polycom same as Cisco is not allowing direct downloads from their 
website).
I am using 1.3.4 with no issues for a couple of months now.

Andrei (MPI)
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Re: [Asterisk-Users] Asterisk and Voice Pulse Open Access

[Randy]

Chris,

I do not have VoicePulse Open Access, but I do have an incoming number through
VoicePulse Connect.  You might want to give the following a try unless you get
a repsonse back from someone who uses Open Access specifically.  (I found the
access1.voicepulse.com address from another posting.)

Edit sip.conf and extensions.conf as follows, editing the 2165551212 to
match your assigned phone number from VoicePulse, as well as filling in your
userid and password.

To have the extension go to another context than default, you must specify it
as the context in the general section in sip.conf - I was unable to get the
normal peer matching to work for voicepulse, at the moment I suspect its due
to inconsistent rev mappings for their ip's.  If you do not have an extension
that matches your number, it will defer to 's'.

sip.conf

; in your [general] section add:
register => userid:[EMAIL PROTECTED]

extensions.conf

; add an extension matching your phone number to your default context (or the
; context specified in sip.conf)
exten => 2165551212,1,Answer
exten => 2165551212,2,Wait,1
exten => 2165551212,3,Playback(vm-goodbye)
exten => 2165551212,4,Hangup

Hope this works for you - it does for me with VoicePulse Connect.

Randy

On Fri, Jan 14, 2005 at 10:19:17PM -0500, Chris Wallace wrote:
> 
>Has  any  messed  with  getting Asterisk to work using the Voice Pulse
>Open Access plan?  I currently have 2 numbers with Voice Pulse, 1 is a
>number  that  is  assigned to their hardware device (Sipura SPA-2000),
>the  other  is a Open Access number that uses SIP from any device (you
>must  authenticate  with  them).   I  want  to be able to use the Open
>Access number on my Asterisk server here at home with no FXO cards.  I
>have  having  a hard time getting this to work; I have only been using
>Asterisk for about a week now.  I have managed to get Asterisk working
>with  a plain phone line going into an XP100.  This list is an awesome
>tool, any help would be appreciated!!!
> 
> 
>Chris

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[Asterisk-Users] Remote Voicemail Retrieval...

[Philippe Daoust]

Hello list,
I want to listen to voicemails on my * box from a phone that is not 
local to my pbx.  I.e., from my cellphone or my PSTN work line etc.  I'm 
aware that I can forward VM to email or use a web interface but that is 
not always practical.

Other than doing an IVR type arrangement or a phone number dedicated to 
VM access is there a way to do this?  On my old POTS line I used to be 
able to call my line and simply punch "*" during unavailable message 
playback to go to the equivalent of voicemailmain().  Is there a way to 
do this in *?

Thanks!
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[Asterisk-Users] Asterisk and Voice Pulse Open Access

[Chris Wallace]

Has any messed with getting Asterisk to work using the Voice
Pulse Open Access plan?  I currently have 2 numbers with Voice Pulse, 1 is
a number that is assigned to their hardware device (Sipura SPA-2000), the other
is a Open Access number that uses SIP from any device (you must authenticate
with them).  I want to be able to use the Open Access number on my
Asterisk server here at home with no FXO cards.  I have having a hard time
getting this to work; I have only been using Asterisk for about a week
now.  I have managed to get Asterisk working with a plain phone line going
into an XP100.  This list is an awesome tool, any help would be
appreciated!!!

 

Chris

 






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Re: [Asterisk-Users] TDM400 answers the line all the time!

[Michael George]

On Tue, Jan 18, 2005 at 04:16:08AM -0600, Justin Carlson wrote:
> no i was using line 1 for testing /w fxs module and i never changed it
> back

does changing it back make a difference?

> On Fri, 2005-01-14 at 07:43 -0500, Michael George wrote:
> > On Mon, Jan 17, 2005 at 08:12:24AM -0600, Justin Carlson wrote:
> > > hi all,
> > > 
> > >   We have a TDM400 card with 4 wfo modules.  now the modules load fine
> > > and when i start asterisk with on phone line connected it just starts
> > > spewing these messages:
> > >-- Starting simple switch on 'Zap/4-1'
> > > Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2
> > > (Ring/Answered)...
> > > Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2
> > > (Ring/Answered)...
> > > Jan 13 12:59:49 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2
> > > (Ring/Answered)...
> > > Jan 13 12:59:49 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2
> > > (Ring/Answered)...
> > > Jan 13 12:59:51 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2
> > > (Ring/Answered)...
> > > 
> > > but no one is calling.  i have plugged in a analog phone and dialed out
> > > on this line before i used it for *.  any help would be great.
> > > 
> > > zapata.conf 
> > > [trunkgroups]
> > > [channels]
> > > language=en
> > > context=routing
> > > group=1
> > > immediate=no
> > > signalling=fxs_ks
> > > channel => 1-4
> > > 
> > > zaptel.conf
> > > fxsks=2-4
> > > loadzone = us
> > 
> > Is there a reason you have "fxsks=2-4" in zaptel.conf rather than 1-4?
> > 
> 
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-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] iaxComm 0.99pre11 binaries posted to Sourceforge

[Michael Van Donselaar]

On Sat, 15 Jan 2005 12:41:42 +1100, Howard Lowndes <[EMAIL PROTECTED]> wrote:

>On Sat, 2005-01-15 at 12:27, Michael Van Donselaar wrote:
>> iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol.
>> It is distributed as part of Steve Kann's iaxclient library.
>> 
>> I've just posted new Windows, Linux and Mac OSX binaries to sourceforge.
>> 
>> The Windows binary was compiled on WinXP.
>> The Linux binary was compiled on RedHat 9.
>
>...and when I try to run this on FC2 it complains:
># ./iaxcomm
>Error wxWindows Fatal Error : Couldn't Initialize IAX Client .

This means that the iaxclient library couldn't initialize.  Most always due to
inability to initialize audio, or trying to run on a system that already has
asterisk running.

>WTF is wxWindows?

An insidious plot to drive people to www.google.com

>
>
>> The OSX binary was compiled by Andreas Wrede on 10.3 and was tested on 10.4
>> (Tiger) beta.
>> 
>> These builds are from iaxclient CVS of 8 JAN 2005.
>> 
>> http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-0.99pre11.zip
>> http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-mac-0.99pre11.zip
>> http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-lin-0.99pre11.tar



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Re: [Asterisk-Users] Spandsp....And garble incoming fax

[Steve Underwood]

Andrew Kohlsmith wrote:
On January 14, 2005 11:09 am, Matthew Boehm wrote:
 

check out my bug post, I have yet to recieve a successful fax using rxfax.
and I'm using newest versions of everything.
   

That's likely your problem.  :-)  I don't feel like registering Yet Another 
Account just to see a bug report, but unless you're using

spandsp-0.0.6
libtiff-3.5.7
asterisk from CVS (that's what I use anyway)
 

If you have the source for spandsp-0.0.6 can you send it to me, please. 
I'm only up to 0.0.2pre7 :-)

Steve
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Re: [Asterisk-Users] iaxComm 0.99pre11 binaries posted to Sourceforge

[Howard Lowndes]

On Sat, 2005-01-15 at 12:27, Michael Van Donselaar wrote:
> iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol.
> It is distributed as part of Steve Kann's iaxclient library.
> 
> I've just posted new Windows, Linux and Mac OSX binaries to sourceforge.
> 
> The Windows binary was compiled on WinXP.
> The Linux binary was compiled on RedHat 9.

...and when I try to run this on FC2 it complains:
# ./iaxcomm
Error wxWindows Fatal Error : Couldn't Initialize IAX Client .

WTF is wxWindows?


> The OSX binary was compiled by Andreas Wrede on 10.3 and was tested on 10.4
> (Tiger) beta.
> 
> These builds are from iaxclient CVS of 8 JAN 2005.
> 
> http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-0.99pre11.zip
> http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-mac-0.99pre11.zip
> http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-lin-0.99pre11.tar
-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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

[Eric Bishop]

It's most definately something to do with the G4 series both DL360 and
DL380. Most G3 series owners are reporting it working OK.


On Fri, 14 Jan 2005 20:50:40 +1100, Adam Goryachev
<[EMAIL PROTECTED]> wrote:
> On Fri, 2005-01-14 at 09:23 +, Steve Hanselman wrote:
> > Has anyone also logged a support call with Digium, it has to be either the
> > card, Linux or the Zaptel drivers.
> 
> You missed the obvious "or the HP Compaq DL380 G4 server"
> 
> Regards,
> Adam
> 
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Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

[Eric Bishop]

I logged a support issue with HP and their response was that it's not
their server that is the problem and if other cards show interrupts
(which they do) there's nothing more they can do



On Fri, 14 Jan 2005 16:30:25 +1000, Joshua McAdam <[EMAIL PROTECTED]> wrote:
> Has anyone logged a support issue with HP on this one?
> 
> I still haven't been able to get it working so far,
> So I'm going to log a support issue here in australia to see what HP can do
> about this and was wondering if anyone else has.
> 
> Josh
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Alexander
> Lopez
> Sent: Monday, 10 January 2005 4:22 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
> 
> Make sure you has a span defined for each port on the TE410P. With out
> signaling it would not take interrupts.
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Karl H.
> Putz
> Sent: Monday, January 10, 2005 12:38 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4
> server
> 
> I have been having this exact problem with a Tatung dual EMT-64 server
> as
> well.
> 
> I have been trying to get a TE410P running and all looks great, driver
> loads, runs ztcfg OK, etc. but no interrupts are ever processed.
> 
> One additional piece of info that I have not seen in this thread is that
> I
> am able to successfully start and run a T100P card in this system.  In
> the
> same PCI slot, wct1xxp driver built from the same CVS HEAD version as
> the
> wct4xxp.
> 
> Just hoping this might shed some light on the problem for any Digium
> folks
> monitoring the forum.
> 
> Karl Putz
> 
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Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

[Eric Bishop]

Hi Peter,

Basically they told me that they have several people complaining of
the problem with G4 series servers and they their hardware engineers
are going to order some of these servers and look into it. Currenly
the only "solution" they have is to use a different motherboard.


On Fri, 14 Jan 2005 16:10:10 +1030, Peter Childs
<[EMAIL PROTECTED]> wrote:
> 
>  Gday Eric.
> 
>  Re: TE410P w/DL380 G4 no interrupts.
> 
>  Did you ever get a resolution to this issue.
> 
>  I have something similar in the newest version of the NEC rack mounted
> servers where
>  the older versions of the servers worked fine.
> 
>  3 days going and I'm going to pour some petrol on this box fairly shortly,
> and I think I
>  will feel much better :)
> 
>  Cheers,
>   Peter
> 
>
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RE: [Asterisk-Users] Packet8 DTA310 and Asterisk

[Olson, Dana]

Okay, I've got success!

What I did was I made the change that you said, but I disabled the secret in 
sip.conf. I then made the SIP changes on the DTA, and then I upgraded the 
firmware to 12.34. I was then able to call the 8006 extension from another 
extension, and I was able to call back that extension from the phone on the 
DTA. I also tested dialing long distance, and that worked fine. Woo, DTA for 
$29US (eBay purchase)!

Thanks.
__
Dana Olson



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olson, Dana
Sent: Friday, January 14, 2005 8:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Packet8 DTA310 and Asterisk


Erik, thanks for your replies.

I tried both ways, and I'm still getting the same messages in the console. Do 
you get these or similar in your console? Do you know what firmware they are 
using by any chance?
__
Dana Olson


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Erik
Espinoza
Sent: Friday, January 14, 2005 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Packet8 DTA310 and Asterisk


Whoops, I meant auth=plain for a packet8 dta.

Erik


On Fri, 14 Jan 2005 16:52:06 -0800, Erik Espinoza
<[EMAIL PROTECTED]> wrote:
> Under your sip.conf change to this:
> 
> [8006]
> type=friend
> host=dynamic
> auth=md5
> secret=1234
> dtmfmode=rfc2833
> context=sip
> callerid=8006
> [EMAIL PROTECTED]
> 
> The key is auth=md5
> 
> I have set a few of my buddies who use to have packet8 on my asterisk
> box just fine.
> 
> Erik
> 
> On Fri, 14 Jan 2005 19:08:52 -0500, Olson, Dana <[EMAIL PROTECTED]> wrote:
> > I have my DTA310 getting an IP via DHCP. I loaded the unlocked firmware 
> > version (Application Code Version: DTA version 1.0 US (8x8 00)) onto it 
> > via TFTP, so I could access the SIP configuration.
> >
> > Under the SIP config, I put the IP of my * system, the 5060 port, and for 
> > Domain Name, I put default (is that right?). I checked off the Send 
> > Registration Request box. Dial Plan I left at the default, 1xx|x.T 
> > (is that right?), and Transport is set to UDP. For Line 1, I have it set as 
> > follows: Phone Number=8006, CallerID Name=8006, Port=5060, AEC On=Off (no 
> > idea what this is), Username=8006, Password=1234.
> >
> > In the OOB Signalling page, where I set the RFC2833 options, I haven't 
> > changed anything from the defaults. Same goes for the VLAN pages.
> >
> > The CODECS page currently has G711U, G711A, and G729 selected, all three 
> > with Silence Suppression turned off.
> >
> > I have other VoIP phones (soft and hard) working.
> >
> > >From extensions.conf:
> > [sip]
> > exten => 8006,1,Answer
> > exten => 8006,2,Wait(1)
> > exten => 8006,3,Dial(SIP/8006,20)
> > exten => 8006,4,Voicemail(u8006)
> > exten => 8006,5,Hangup
> > exten => 8006,103,Voicemail(b8006)
> > exten => 8006,104,Hangup
> >
> > >From sip.conf:
> > [8006]
> > type=friend
> > host=dynamic
> > secret=1234
> > dtmfmode=rfc2833
> > context=sip
> > callerid=8006
> > [EMAIL PROTECTED]
> >
> > >From voicemail.conf:
> > [default]
> > 8006 => ,Packet8,[EMAIL PROTECTED]
> >
> > Okay, so when I apply those settings and restart the unit, I get a bunch of 
> > these messages in the * console:
> >
> > Jan 14 18:03:42 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to 
> > authenticate user 8006;tag=t2a7aca1ef6g41a7 for 
> > SUBSCRIBE
> >
> > And these messages continue once or twice every second until I reset the 
> > phone or unplug it. During this time, if I pick up the phone, I hear a 
> > dialtone. If I hang up and pick up again, the dialtone is still there. 
> > However, if I try dialing another * extension, say 8005, it doesn't do 
> > anything. If I hang up and pick up again after trying this, the dialtone is 
> > gone. If I try the same thing but instead dial 8005#, the same thing 
> > happens. If I wait for a few seconds/minute after hanging up, I'll get the 
> > dialtone back. If I try to dial a long disatance, the same thing happens. 
> > If I try without the 1 for long distance, the same thing happens. If I try 
> > with a 9 in front of 1 and then the number (this works for our other 
> > phones) then the same thing happens. However, now I'm getting more messages 
> > in the * console mixed in with the original ones:
> >
> > Jan 14 19:04:24 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to 
> > authenticate user 8006;tag=t2c3b0a372ad3g for 
> > SUBSCRIBE
> >
> > Maybe the dialplan is set wrong in the web config of the DTA310? I don't 
> > know what I would set it to though.
> >
> > If anyone can assist, then that would be appreciated.
> > __
> > Dana Olson
> >
> > Disclaimer: The information transmitted in this message is intended only 
> > for the person or entity to which it is addressed and may contain 
> > confidential and/or privileged material.  Any review, retransmission, 
> > dissemina

[Asterisk-Users] IAX on multiple ports

[nik martin]

Is it possible to listen on more than one port within a single instance 
of *?  I have an engineer in Iraq that we need voice comms with, but the 
gov't limits traffic to ports 80,443, 25, and 110.  Can I set up IAX to 
listen on port 80 AND the regular IAX port?

Or will I have to set up some weird TCP tunnel?
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[Asterisk-Users] iaxComm 0.99pre11 binaries posted to Sourceforge

[Michael Van Donselaar]

iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol.
It is distributed as part of Steve Kann's iaxclient library.

I've just posted new Windows, Linux and Mac OSX binaries to sourceforge.

The Windows binary was compiled on WinXP.
The Linux binary was compiled on RedHat 9.
The OSX binary was compiled by Andreas Wrede on 10.3 and was tested on 10.4
(Tiger) beta.

These builds are from iaxclient CVS of 8 JAN 2005.

http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-0.99pre11.zip
http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-mac-0.99pre11.zip
http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-lin-0.99pre11.tar



--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 1/12/2005

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RE: [Asterisk-Users] Packet8 DTA310 and Asterisk

[Olson, Dana]

Erik, thanks for your replies.

I tried both ways, and I'm still getting the same messages in the console. Do 
you get these or similar in your console? Do you know what firmware they are 
using by any chance?
__
Dana Olson


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Erik
Espinoza
Sent: Friday, January 14, 2005 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Packet8 DTA310 and Asterisk


Whoops, I meant auth=plain for a packet8 dta.

Erik


On Fri, 14 Jan 2005 16:52:06 -0800, Erik Espinoza
<[EMAIL PROTECTED]> wrote:
> Under your sip.conf change to this:
> 
> [8006]
> type=friend
> host=dynamic
> auth=md5
> secret=1234
> dtmfmode=rfc2833
> context=sip
> callerid=8006
> [EMAIL PROTECTED]
> 
> The key is auth=md5
> 
> I have set a few of my buddies who use to have packet8 on my asterisk
> box just fine.
> 
> Erik
> 
> On Fri, 14 Jan 2005 19:08:52 -0500, Olson, Dana <[EMAIL PROTECTED]> wrote:
> > I have my DTA310 getting an IP via DHCP. I loaded the unlocked firmware 
> > version (Application Code Version: DTA version 1.0 US (8x8 00)) onto it 
> > via TFTP, so I could access the SIP configuration.
> >
> > Under the SIP config, I put the IP of my * system, the 5060 port, and for 
> > Domain Name, I put default (is that right?). I checked off the Send 
> > Registration Request box. Dial Plan I left at the default, 1xx|x.T 
> > (is that right?), and Transport is set to UDP. For Line 1, I have it set as 
> > follows: Phone Number=8006, CallerID Name=8006, Port=5060, AEC On=Off (no 
> > idea what this is), Username=8006, Password=1234.
> >
> > In the OOB Signalling page, where I set the RFC2833 options, I haven't 
> > changed anything from the defaults. Same goes for the VLAN pages.
> >
> > The CODECS page currently has G711U, G711A, and G729 selected, all three 
> > with Silence Suppression turned off.
> >
> > I have other VoIP phones (soft and hard) working.
> >
> > >From extensions.conf:
> > [sip]
> > exten => 8006,1,Answer
> > exten => 8006,2,Wait(1)
> > exten => 8006,3,Dial(SIP/8006,20)
> > exten => 8006,4,Voicemail(u8006)
> > exten => 8006,5,Hangup
> > exten => 8006,103,Voicemail(b8006)
> > exten => 8006,104,Hangup
> >
> > >From sip.conf:
> > [8006]
> > type=friend
> > host=dynamic
> > secret=1234
> > dtmfmode=rfc2833
> > context=sip
> > callerid=8006
> > [EMAIL PROTECTED]
> >
> > >From voicemail.conf:
> > [default]
> > 8006 => ,Packet8,[EMAIL PROTECTED]
> >
> > Okay, so when I apply those settings and restart the unit, I get a bunch of 
> > these messages in the * console:
> >
> > Jan 14 18:03:42 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to 
> > authenticate user 8006;tag=t2a7aca1ef6g41a7 for 
> > SUBSCRIBE
> >
> > And these messages continue once or twice every second until I reset the 
> > phone or unplug it. During this time, if I pick up the phone, I hear a 
> > dialtone. If I hang up and pick up again, the dialtone is still there. 
> > However, if I try dialing another * extension, say 8005, it doesn't do 
> > anything. If I hang up and pick up again after trying this, the dialtone is 
> > gone. If I try the same thing but instead dial 8005#, the same thing 
> > happens. If I wait for a few seconds/minute after hanging up, I'll get the 
> > dialtone back. If I try to dial a long disatance, the same thing happens. 
> > If I try without the 1 for long distance, the same thing happens. If I try 
> > with a 9 in front of 1 and then the number (this works for our other 
> > phones) then the same thing happens. However, now I'm getting more messages 
> > in the * console mixed in with the original ones:
> >
> > Jan 14 19:04:24 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to 
> > authenticate user 8006;tag=t2c3b0a372ad3g for 
> > SUBSCRIBE
> >
> > Maybe the dialplan is set wrong in the web config of the DTA310? I don't 
> > know what I would set it to though.
> >
> > If anyone can assist, then that would be appreciated.
> > __
> > Dana Olson
> >
> > Disclaimer: The information transmitted in this message is intended only 
> > for the person or entity to which it is addressed and may contain 
> > confidential and/or privileged material.  Any review, retransmission, 
> > dissemination, or other use of or taking of any action in reliance upon 
> > this information by persons or entities other than the intended recipient 
> > is prohibited.  If you received this message in error, please contact the 
> > sender and delete the material from any system.
> >
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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Re: [Asterisk-Users] Softphone for Linux recommendation

[William Suffill]

Yes iaxcomm is an IAX softphone. I know Xten is working on improving
their linux support for their SIP based shoftphones.
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Re: [Asterisk-Users] Packet8 DTA310 and Asterisk

[Erik Espinoza]

Whoops, I meant auth=plain for a packet8 dta.

Erik


On Fri, 14 Jan 2005 16:52:06 -0800, Erik Espinoza
<[EMAIL PROTECTED]> wrote:
> Under your sip.conf change to this:
> 
> [8006]
> type=friend
> host=dynamic
> auth=md5
> secret=1234
> dtmfmode=rfc2833
> context=sip
> callerid=8006
> [EMAIL PROTECTED]
> 
> The key is auth=md5
> 
> I have set a few of my buddies who use to have packet8 on my asterisk
> box just fine.
> 
> Erik
> 
> On Fri, 14 Jan 2005 19:08:52 -0500, Olson, Dana <[EMAIL PROTECTED]> wrote:
> > I have my DTA310 getting an IP via DHCP. I loaded the unlocked firmware 
> > version (Application Code Version: DTA version 1.0 US (8x8 00)) onto it 
> > via TFTP, so I could access the SIP configuration.
> >
> > Under the SIP config, I put the IP of my * system, the 5060 port, and for 
> > Domain Name, I put default (is that right?). I checked off the Send 
> > Registration Request box. Dial Plan I left at the default, 1xx|x.T 
> > (is that right?), and Transport is set to UDP. For Line 1, I have it set as 
> > follows: Phone Number=8006, CallerID Name=8006, Port=5060, AEC On=Off (no 
> > idea what this is), Username=8006, Password=1234.
> >
> > In the OOB Signalling page, where I set the RFC2833 options, I haven't 
> > changed anything from the defaults. Same goes for the VLAN pages.
> >
> > The CODECS page currently has G711U, G711A, and G729 selected, all three 
> > with Silence Suppression turned off.
> >
> > I have other VoIP phones (soft and hard) working.
> >
> > >From extensions.conf:
> > [sip]
> > exten => 8006,1,Answer
> > exten => 8006,2,Wait(1)
> > exten => 8006,3,Dial(SIP/8006,20)
> > exten => 8006,4,Voicemail(u8006)
> > exten => 8006,5,Hangup
> > exten => 8006,103,Voicemail(b8006)
> > exten => 8006,104,Hangup
> >
> > >From sip.conf:
> > [8006]
> > type=friend
> > host=dynamic
> > secret=1234
> > dtmfmode=rfc2833
> > context=sip
> > callerid=8006
> > [EMAIL PROTECTED]
> >
> > >From voicemail.conf:
> > [default]
> > 8006 => ,Packet8,[EMAIL PROTECTED]
> >
> > Okay, so when I apply those settings and restart the unit, I get a bunch of 
> > these messages in the * console:
> >
> > Jan 14 18:03:42 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to 
> > authenticate user 8006;tag=t2a7aca1ef6g41a7 for 
> > SUBSCRIBE
> >
> > And these messages continue once or twice every second until I reset the 
> > phone or unplug it. During this time, if I pick up the phone, I hear a 
> > dialtone. If I hang up and pick up again, the dialtone is still there. 
> > However, if I try dialing another * extension, say 8005, it doesn't do 
> > anything. If I hang up and pick up again after trying this, the dialtone is 
> > gone. If I try the same thing but instead dial 8005#, the same thing 
> > happens. If I wait for a few seconds/minute after hanging up, I'll get the 
> > dialtone back. If I try to dial a long disatance, the same thing happens. 
> > If I try without the 1 for long distance, the same thing happens. If I try 
> > with a 9 in front of 1 and then the number (this works for our other 
> > phones) then the same thing happens. However, now I'm getting more messages 
> > in the * console mixed in with the original ones:
> >
> > Jan 14 19:04:24 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to 
> > authenticate user 8006;tag=t2c3b0a372ad3g for 
> > SUBSCRIBE
> >
> > Maybe the dialplan is set wrong in the web config of the DTA310? I don't 
> > know what I would set it to though.
> >
> > If anyone can assist, then that would be appreciated.
> > __
> > Dana Olson
> >
> > Disclaimer: The information transmitted in this message is intended only 
> > for the person or entity to which it is addressed and may contain 
> > confidential and/or privileged material.  Any review, retransmission, 
> > dissemination, or other use of or taking of any action in reliance upon 
> > this information by persons or entities other than the intended recipient 
> > is prohibited.  If you received this message in error, please contact the 
> > sender and delete the material from any system.
> >
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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Re: [Asterisk-Users] Packet8 DTA310 and Asterisk

[Erik Espinoza]

Under your sip.conf change to this:

[8006]
type=friend
host=dynamic
auth=md5
secret=1234
dtmfmode=rfc2833
context=sip
callerid=8006
[EMAIL PROTECTED]

The key is auth=md5

I have set a few of my buddies who use to have packet8 on my asterisk
box just fine.

Erik

On Fri, 14 Jan 2005 19:08:52 -0500, Olson, Dana <[EMAIL PROTECTED]> wrote:
> I have my DTA310 getting an IP via DHCP. I loaded the unlocked firmware 
> version (Application Code Version: DTA version 1.0 US (8x8 00)) onto it 
> via TFTP, so I could access the SIP configuration.
> 
> Under the SIP config, I put the IP of my * system, the 5060 port, and for 
> Domain Name, I put default (is that right?). I checked off the Send 
> Registration Request box. Dial Plan I left at the default, 1xx|x.T 
> (is that right?), and Transport is set to UDP. For Line 1, I have it set as 
> follows: Phone Number=8006, CallerID Name=8006, Port=5060, AEC On=Off (no 
> idea what this is), Username=8006, Password=1234.
> 
> In the OOB Signalling page, where I set the RFC2833 options, I haven't 
> changed anything from the defaults. Same goes for the VLAN pages.
> 
> The CODECS page currently has G711U, G711A, and G729 selected, all three with 
> Silence Suppression turned off.
> 
> I have other VoIP phones (soft and hard) working.
> 
> >From extensions.conf:
> [sip]
> exten => 8006,1,Answer
> exten => 8006,2,Wait(1)
> exten => 8006,3,Dial(SIP/8006,20)
> exten => 8006,4,Voicemail(u8006)
> exten => 8006,5,Hangup
> exten => 8006,103,Voicemail(b8006)
> exten => 8006,104,Hangup
> 
> >From sip.conf:
> [8006]
> type=friend
> host=dynamic
> secret=1234
> dtmfmode=rfc2833
> context=sip
> callerid=8006
> [EMAIL PROTECTED]
> 
> >From voicemail.conf:
> [default]
> 8006 => ,Packet8,[EMAIL PROTECTED]
> 
> Okay, so when I apply those settings and restart the unit, I get a bunch of 
> these messages in the * console:
> 
> Jan 14 18:03:42 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to 
> authenticate user 8006;tag=t2a7aca1ef6g41a7 for 
> SUBSCRIBE
> 
> And these messages continue once or twice every second until I reset the 
> phone or unplug it. During this time, if I pick up the phone, I hear a 
> dialtone. If I hang up and pick up again, the dialtone is still there. 
> However, if I try dialing another * extension, say 8005, it doesn't do 
> anything. If I hang up and pick up again after trying this, the dialtone is 
> gone. If I try the same thing but instead dial 8005#, the same thing happens. 
> If I wait for a few seconds/minute after hanging up, I'll get the dialtone 
> back. If I try to dial a long disatance, the same thing happens. If I try 
> without the 1 for long distance, the same thing happens. If I try with a 9 in 
> front of 1 and then the number (this works for our other phones) then the 
> same thing happens. However, now I'm getting more messages in the * console 
> mixed in with the original ones:
> 
> Jan 14 19:04:24 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to 
> authenticate user 8006;tag=t2c3b0a372ad3g for SUBSCRIBE
> 
> Maybe the dialplan is set wrong in the web config of the DTA310? I don't know 
> what I would set it to though.
> 
> If anyone can assist, then that would be appreciated.
> __
> Dana Olson
> 
> Disclaimer: The information transmitted in this message is intended only for 
> the person or entity to which it is addressed and may contain confidential 
> and/or privileged material.  Any review, retransmission, dissemination, or 
> other use of or taking of any action in reliance upon this information by 
> persons or entities other than the intended recipient is prohibited.  If you 
> received this message in error, please contact the sender and delete the 
> material from any system.
> 
> ___
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> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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RE: [Asterisk-Users] ULaw not negotiating

[Paul Rodan]

According to the provider's techs, Asterisk isn't properly following the SIP
guidelines about codec negotiation. 

>From my understanding of the convo, Asterisk is expecting multiple codec
capabilities to be in the Capabilities string, in the first header sent
over, if it doesn't find a match, it reports no compatible codecs and drops
the call. But the SIP standard calls more for something like 1 codec per
header, the remote server first sends what its preferred codec is (in this
case g729a), and then a "true SIP device" is supposed to respond and say
that the codec isn't acceptable (mine only accepts G711U), then the remote
system sends over another header with the next preferred codec (in this case
would have been G711ULaw), and so on until there is a match (mine would
accept G711ULaw and proceed with the call). Asterisk is accepting the
information in the first header as the ONLY codecs the remote system
accepts, and doesn't understanding sending a SIP response to the remote
system so that it will then send the next available codec, if any.

Our system has to send a response and say g729 isn't acceptable and to
please send the next available. Their techs are writing a patch for
chan_sip.c now to correct the issue and make sip more compatible with
standards. 

This is what I was told, I also may have misinterpreted the whole thing, I'm
still learning, but it sounds quite feasible. This patch should add that
functionality and make the system work great. 




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Friday, January 14, 2005 6:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ULaw not negotiating

Paul Rodan wrote:
> Capabilities: us - 0x4(ULAW), peer - audio=0x100(G729A)/video=0x0(EMPTY),
> combined - 0x0(EMPTY)
> 
> Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
> 0x1(G723)
> 
> Jan 14 17:29:55 WARNING[81922]: chan_sip.c:2820 process_sdp: No compatible
> codecs!
> 
>  
> 
> What throws me off is the description format of G729. They said they used
to
> be sending in ULaw and G729, but then I had him turn off G729 all
together.
> But this sip debug doesn't confirm that, I see g729 mentioned several
times.
> But I do see ULaw mentioned in there as well as well as 0x4, under
> Capabilities. So why isn't Asterisk accepting it?

They lie.  They are sending G729 only.
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Re: [Asterisk-Users] Routing incoming calls to various extensions.

[Don Dawson]

Try
stratagy = leastrecent


- Original Message -
From: "dean collins" <[EMAIL PROTECTED]>
To: "Denis Voitenko" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Friday, January 14, 2005 5:16 PM
Subject: RE: [Asterisk-Users] Routing incoming calls to various extensions.


Do a search on ACD and agents, this is certainly achievable.


Cheers,
Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Denis
Voitenko
Sent: Friday, January 14, 2005 5:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Routing incoming calls to various extensions.

I am setting up * to accept incoming calls and route them to our reps.
What I'd like to do route the call to the rep who has been idle the
most, thus distributing the load among the reps. I can't seem to find
this functionality. Can someone point me in the right direction?

Script Head
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[Asterisk-Users] Packet8 DTA310 and Asterisk

[Olson, Dana]

I have my DTA310 getting an IP via DHCP. I loaded the unlocked firmware version 
(Application Code Version: DTA version 1.0 US (8x8 00)) onto it via TFTP, 
so I could access the SIP configuration.

Under the SIP config, I put the IP of my * system, the 5060 port, and for 
Domain Name, I put default (is that right?). I checked off the Send 
Registration Request box. Dial Plan I left at the default, 1xx|x.T (is 
that right?), and Transport is set to UDP. For Line 1, I have it set as 
follows: Phone Number=8006, CallerID Name=8006, Port=5060, AEC On=Off (no idea 
what this is), Username=8006, Password=1234.

In the OOB Signalling page, where I set the RFC2833 options, I haven't changed 
anything from the defaults. Same goes for the VLAN pages.

The CODECS page currently has G711U, G711A, and G729 selected, all three with 
Silence Suppression turned off.

I have other VoIP phones (soft and hard) working.

>From extensions.conf:
[sip]
exten => 8006,1,Answer
exten => 8006,2,Wait(1)
exten => 8006,3,Dial(SIP/8006,20)
exten => 8006,4,Voicemail(u8006)
exten => 8006,5,Hangup
exten => 8006,103,Voicemail(b8006)
exten => 8006,104,Hangup

>From sip.conf:
[8006]
type=friend
host=dynamic
secret=1234
dtmfmode=rfc2833
context=sip
callerid=8006
[EMAIL PROTECTED]

>From voicemail.conf:
[default]
8006 => ,Packet8,[EMAIL PROTECTED]


Okay, so when I apply those settings and restart the unit, I get a bunch of 
these messages in the * console:

Jan 14 18:03:42 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to 
authenticate user 8006;tag=t2a7aca1ef6g41a7 for SUBSCRIBE

And these messages continue once or twice every second until I reset the phone 
or unplug it. During this time, if I pick up the phone, I hear a dialtone. If I 
hang up and pick up again, the dialtone is still there. However, if I try 
dialing another * extension, say 8005, it doesn't do anything. If I hang up and 
pick up again after trying this, the dialtone is gone. If I try the same thing 
but instead dial 8005#, the same thing happens. If I wait for a few 
seconds/minute after hanging up, I'll get the dialtone back. If I try to dial a 
long disatance, the same thing happens. If I try without the 1 for long 
distance, the same thing happens. If I try with a 9 in front of 1 and then the 
number (this works for our other phones) then the same thing happens. However, 
now I'm getting more messages in the * console mixed in with the original ones:

Jan 14 19:04:24 NOTICE[17688]: chan_sip.c:7446 handle_request: Failed to 
authenticate user 8006;tag=t2c3b0a372ad3g for SUBSCRIBE


Maybe the dialplan is set wrong in the web config of the DTA310? I don't know 
what I would set it to though.

If anyone can assist, then that would be appreciated.
__
Dana Olson

Disclaimer: The information transmitted in this message is intended only for 
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Re: [Asterisk-Users] Softphone for Linux recommendation

[Howard Lowndes]

On Sat, 2005-01-15 at 07:09, Adam Fineberg wrote:
> Howard Lowndes wrote:
> 
> >Can anyone _recommend_ a downloadable OSS softphone that _works_ under
> >Linux and is compatible with Asterisk.
> >
> >So far I have tried kphone and linphone and had problems with both, and
> >I am still waiting to hear back from the X-Lite beta folks.
> >
> >  
> >
> 
> How about iaxcomm?
> 
> http://iaxclient.sourceforge.net/iaxcomm/

I should have added SIP reqd.  I assume this only does IAX2 but I will
look at it.

I have almost got sflphone compiled only I have hit a missing file in
one of the library compiles along the way.

> 
> Adam
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-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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RE: [Asterisk-Users] Routing incoming calls to various extensions.

[dean collins]

Do a search on ACD and agents, this is certainly achievable.


Cheers,
Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Denis
Voitenko
Sent: Friday, January 14, 2005 5:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Routing incoming calls to various extensions.

I am setting up * to accept incoming calls and route them to our reps.
What I'd like to do route the call to the rep who has been idle the
most, thus distributing the load among the reps. I can't seem to find
this functionality. Can someone point me in the right direction?

Script Head
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[Asterisk-Users] Problems with a combination of AVM B1 and HFC-S on Kernel 2.6.x

[Uwe Betz]

Hello List!
Several users of a german VoIP-Forum experience the following similar 
problems when using CAPI with an active ISDN-Card AVM B1 PCI connected 
to the PSTN and a HFC-S in NT-Mode used as an "internal" S0-Bus to 
connect ISDN-Phones.

The problem is, that when making a phone call from an ISDN-Phone 
connected to the HFC-S-Card that is meant to be routed to the PSTN 
through the AVM B1, the voice quality is very bad. It sounds as if 
someone has a really bad, bad cold.

What makes us wander is, that if a call is made from a SIP-client 
through the AVM B1 to the PSTN, everything ist fine. Calls between 
SIP-clients and ISDN-Phhones connected to the HFC-CArd are also ok.

In addition one user found that if one monitors/records a phonecall made 
from an ISDN-Phone to the PSTN through the AVM B1, the recorded sound is 
also fine, only the "live" sound is that bad and we have no more ideas 
what to try to figure out where the problem is.

What we heard, but we still have to verify this is, that on a Linx with 
Kernel 2.4 everything seems to work. But this is still unverified. The 
Kernel is 2.6.8 in my case the actual updated Kernel of a SuSE 9.2 
Distribution.

We were also testing different Motherboards and made sure that there are 
no IRQ-Conflicts between HFC and AVM card, etc. The Test-Systems had no 
load and are running smoothly.

Any ideas?
Thanks,
Jui
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Re: [Asterisk-Users] Problems with loading TE110 module

[Asterisk List]

I encountered the same problem today.  'lspci -nvv' showed that the
subsystem ID of the TE110p changed from 79fe to 79fa or 797e. 
Powering off/on the machine restored the subsystem ID to 79fe and the
wcte11xp module could then load.  I already emailed digium support for
this.


On Mon, 20 Dec 2004 11:32:15 +0100, Tamas J <[EMAIL PROTECTED]> wrote:
> 
> 
> Hello!
> 
> I discovered, that I'm unable to load ther kernel module under 2.4.28.
> Before that I had 2.4.26 and tryed to upgrade the kernel to 2.4.28.
> After restart (reboot - soft restart) I can't load the module. When I
> go back to 2.4.26 and try to load the module, I'm getting the same
> problem.
> Only turning off/on helps. What can I do? It's very annoying because
> the box is in hosting and not easy to just restart.
> 
> The wct1xxp worked fine in the same box (with restarts also).
> 
> Any idea, hint?
> 
> Kind regards,
>Tamas
> 
> modprobe wcte11xp
> /lib/modules/2.4.28-magic/misc/wcte11xp.o: init_module: No such device
> Hint: insmod errors can be caused by incorrect module parameters, including 
> invalid IO or IRQ parameters.
>  You may find more information in syslog or the output from dmesg
> /lib/modules/2.4.28-magic/misc/wcte11xp.o: insmod 
> /lib/modules/2.4.28-magic/misc/wcte11xp.o failed
> /lib/modules/2.4.28-magic/misc/wcte11xp.o: insmod wcte11xp failed
> 
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Re: [Asterisk-Users] ULaw not negotiating

[Eric Wieling aka ManxPower]

Paul Rodan wrote:
Capabilities: us - 0x4(ULAW), peer - audio=0x100(G729A)/video=0x0(EMPTY),
combined - 0x0(EMPTY)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Jan 14 17:29:55 WARNING[81922]: chan_sip.c:2820 process_sdp: No compatible
codecs!
 

What throws me off is the description format of G729. They said they used to
be sending in ULaw and G729, but then I had him turn off G729 all together.
But this sip debug doesn't confirm that, I see g729 mentioned several times.
But I do see ULaw mentioned in there as well as well as 0x4, under
Capabilities. So why isn't Asterisk accepting it?
They lie.  They are sending G729 only.
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[Asterisk-Users] Routing incoming calls to various extensions.

[Denis Voitenko]

I am setting up * to accept incoming calls and route them to our reps.
What I'd like to do route the call to the rep who has been idle the
most, thus distributing the load among the reps. I can't seem to find
this functionality. Can someone point me in the right direction?

Script Head
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[Asterisk-Users] ULaw not negotiating

[Paul Rodan]

Ok,

 

My provider is sending a call to me via ULaw but Asterisk
isn’t picking up on this, I’ve only allowed ulaw, I disallow=all
and then allow=ulaw in my sip.conf and that’s the only thing I allow, but
when my provider sends me the requests, I get an error about No Compatible
Codecs:

 

 

17 headers, 8 lines

Using latest request as basis request

Sending to 67.19.245.213 : 5060 (non-NAT)

Found RTP audio format 18

Found RTP audio format 101

Peer audio RTP is at port 38.114.20.207:28442

Found description format G729

Found description format telephone-event

Capabilities: us - 0x4(ULAW), peer -
audio=0x100(G729A)/video=0x0(EMPTY), combined - 0x0(EMPTY)

Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723),
combined - 0x1(G723)

Jan 14 17:29:55 WARNING[81922]: chan_sip.c:2820 process_sdp:
No compatible codecs!

 

What throws me off is the description format of G729. They
said they used to be sending in ULaw and G729, but then I had him turn off G729
all together. But this sip debug doesn’t confirm that, I see g729
mentioned several times. But I do see ULaw mentioned in there as well as well
as 0x4, under Capabilities. So why isn’t Asterisk accepting it?

 

Any help with this debug would be immensely helpful!

 

 

I also have BroadVoice which sends me calls ULaw and works
fine, so I called that number and captured the inbound with sip debug and saw
this:

 

14 headers, 9 lines

Using latest request as basis request

Sending to 147.135.4.128 : 5060 (non-NAT)

Found RTP audio format 0

Found RTP audio format 101

Peer audio RTP is at port 147.135.4.128:14664

Found description format PCMU

Found description format telephone-event

Capabilities: us - 0x4(ULAW), peer -
audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)

Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723),
combined - 0x1(G723)

Found peer 'broadvoice'

Looking for 551212 in incoming

list_route: hop: 

Transmitting (no NAT):

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 147.135.4.128:5060;branch=jkjk245kjelkjelkj2435sadflkj435.1sr

From: "TEST PHONE";tag=SD50vt601-662260634-1105742161664

To: "Deon Rodden";tag=as59299ec2

Call-ID: SD50vt601-8b297b5d0b4543648439f18f9eba5903-js19002

CSeq: 967783297 INVITE

User-Agent: CSCO/7

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: 

Content-Length: 0

Answering with preferred capability 0x4(ULAW)

Reliably Transmitting (no NAT):

 

 

Best Regards,

Paul






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Re: [Asterisk-Users] Firefly repeats registering to * server

[timebandit001]

> Is the reregistering normal behaviour for an external client ?

Yes, IAX default behavior is to register every minutes or so, external
or internal

If I'm wrong, please someone correct me
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[Asterisk-Users] Having trouble with T405P and PPP: ZT_SPANCONFIG failed

[Ben Greear]

Hello!
I am trying to set up multi-link PPP using two T100P cards in one
machine, and 1 T405P card (the 4-port one) in another machine.  I have
previously been able to get PPP working between the two T100P cards
in separate machines
The 4-port card seems to be my problem currently.  I am trying to use the tor2
driver (from a fresh CVS download this morning).  When I load the driver (or 
run ztcfg)
I get this error:
ZT_SPANCONFIG failed on span 1:  No such device or address (6)
I assume it is a configuration issue on my part, but I don't see
what I'm doing wrong.
ztcfg -vvv gives:
Zaptel Configuration
==
SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: Clear channel (Default) (Slaves: 01 02 03 04 05 06 07 08 09 10 11 
12 13 14 15 16 17 18 19 20 21 22 23 24)
Channel 25: Clear channel (Default) (Slaves: 25 26 27 28 29 30 31 32 33 34 35 
36 37 38 39 40 41 42 43 44 45 46 47 48)
48 channels configured.
Please let me know if you have any ideas!
Thanks,
Ben
--
Ben Greear <[EMAIL PROTECTED]>
Candela Technologies Inc  http://www.candelatech.com
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[Asterisk-Users] DIAX PC to Phone

[Bilal Ghayad]

Dear Dan;

Thanks for your kindly email and reply.

Is DIAX supported for G723 codec and can work on Windows OS?

Regards
Bilal
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[Asterisk-Users] ASTCC

[Bilal Ghayad]

Dear Sebastian;

Thanks a lot for your kindly advise to use ASTCC.

But can u advise me the link for ASTCC to download it and wether it is open
source (to download the source and work on it?

Regards
Bilal

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[Asterisk-Users] Asterisk for voicemail -> C2611XM, 7940 & 7960 phones

[Stafford A. Rau]

Hello,

My group at work has a test voip deployment, using a Cisco 2611XM with
FXO and FXS modules, and a small number of Cisco 7940 and 7960 ip
phones. This is working today, with outgoing calls going over a pair of
POTS lines on the 2611XM.

I would like to add an asterisk server to the mix, just to add voicemail
(at least for starters).

The phones, router, and asterisk server are all on seperate private
internal networks.

I'm very new to all things voice, and have been digging in to the
various asterisk docs as fast as I can absorb them.

I could use push in the right direction, and some sample configs would
be even better.

Thanks,
--Stafford

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Re: [Asterisk-Users] I Don't Want Asterisk in the Media Path

[Daryll Strauss]

Look for "canreinvite=yes" to get Asterisk out of the RTP path. Since
SIP traffic is infrequent and low volume having Asterisk in that loop
shouldn't be a problem, it's the RTP traffic you really want going
point to point. Realize that Asterisk can't get out of the loop if you
use the t or T option to support transfers or if you're recording the
call.
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Re: [Asterisk-Users] Re: Budgetone and MWI

[C F]

The message button can be programmed to dial an extension that checks voicemail
exten => 160,1,Voicemailmain(${CALLERIDNUM})


On Fri, 14 Jan 2005 18:57:41 +0100, Aldo Bergamini <[EMAIL PROTECTED]> wrote:
> [EMAIL PROTECTED] is believed to have said:
> 
> >I don't mean to be rude to everyone who responded to this question, but
> >I think that everyone is answering the wrong question. The point is that
> >the message waiting indicator doesn't light up, at all, ever. All that
> >happens when messages are waiting is that the display blinks and the
> >phone gives a stutter dialtone. That's it. There is no light under the
> >button - there should be, but there isn't. The "blinking" phone
> >designers should have put those stupid blinking red leds - that only
> >flash on boot up - under the message button and flashed the display
> >during boot up. But they didn't and we're stuck with it. Such is life.
> >
> >Stephen R. Besch
> 
> I noticed this strange factoid as soon as I got MWI to work.
> 
> Does anybody know what then is the use for the message button?
> 
> Thanks,
> Aldo
> 
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Re: [Asterisk-Users] SS7 and Asterisk solution

[Steven Critchfield]

On Fri, 2005-01-14 at 14:38 -0600, Matthew Boehm wrote:
> Time for you to learn some maners and not to bitch at/insult people for
> something they don't understand. Time for you to stop telling people what to
> do. Please add those 2 to you ToDo list.

Before your whining gets too out of hand, you need to find anything
specific in my original reply that was a blatant insult.If you don't
like being called for being lazy, then don't act that way. 

The information was put before you to then do the minor amount of
homework necessary to understand. In this case, while Eric Wieling did
you a favor and explained a bit more, it should have been left as an
exercise for the reader as it involves legalities that while most of us
agree with the answer, we are not qualified to give the advice.(you
know, no law degree, not retained counsel and all that fun stuff) It
also is covered in the GPL faq. I'm sure a simple search of the archive
would show a few threads of this exact nature in less than 

> - Original Message - 
> From: "Steven Critchfield" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Friday, January 14, 2005 2:19 PM
> Subject: Re: [Asterisk-Users] SS7 and Asterisk solution
> 
> 
> > On Fri, 2005-01-14 at 14:09 -0600, Matthew Boehm wrote:
> > > So are you telling me that you cannot use other commercial products in
> > > conjunction with asterisk?
> >
> > Time for you to go learn about the GPL. Time to go learn about proper
> > trimming of an email. Time to learn how to use the archives for
> > information already given. Time to go learn a lot of things. Please add
> > those 3 to your ToDo list though.
> >
> > > - Original Message - 
> > > From: "izo" <[EMAIL PROTECTED]>
> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > 
> > > Sent: Friday, January 14, 2005 1:32 PM
> > > Subject: Re: [Asterisk-Users] SS7 and Asterisk solution
> > >
> > >
> > > > On Fri, 14 Jan 2005 10:01:52 -0600, Matthew Boehm wrote:
> > > > > Why does it have to be commercially licenced?
> > > >
> > > > Without it, the SS7 software would be linking to GPL software which
> > > > means they would have
> > > > to GPL the code too. So the only way to get commercial SS7 is to have
> > > > it with commercial
> > > > asterisk.
> > > >

-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] SS7 and Asterisk solution

[Eric Wieling aka ManxPower]

Matthew Boehm wrote:
Eric,
  Thank you for explaining this to me instead of being rude and bitching at
me about my lack of GPL understanding.
You caught me in an unusally good mood, that's all.
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Re: [Asterisk-Users] Help in E1-T1 encoding

[Peter Svensson]

On Fri, 14 Jan 2005, Alejandro G wrote:

> Sorry for the delay. I had to reconfigure all again. I do I inbound call to
> asterisk and the result log is this (hope this is usefull):

> < Protocol Discriminator: Q.931 (8)  len=47
> < Call Ref: len= 2 (reference 1280/0x500) (Originator)
> < Message type: SETUP (5)
> < [04 03 80 90 a2]
> < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
> capability: Speech (0)
> <  Ext: 1  Trans mode/rate: 64kbps, circuit-mode
> (16)
> <  Ext: 1  User information layer 1: u-Law (34)
> < [18 03 a9 83 81]

Ok, the incoming call is signalled as being u-Law from the originating
device to Asterisk. Can you describe the set up (E1/T1, expected codecs on
the various legs) for this call? Also, sum up the problem again.

Peter


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[Asterisk-Users] Re: T100P with NEC C2400 IPX switch

[Shields, Larry]

Title: Re: T100P with NEC C2400 IPX switch





Jerry,


I have made this work on a 2400 IPX and * 1.0 with the T100P setup as a
PRI.  The PRI has to be programmed so that the IPX is set as the CPE
side and * as the NETWORK side. 


--LJ



"Jerry Geis" <[EMAIL PROTECTED]> wrote in message
news:<[EMAIL PROTECTED]>...
> I am looking at interface the T100P with a NEC C2400 IPX.
> Has this been done before by anyone???
> 
> quick search did not bring anything up.
> 
> Thanks,
> 
> Jerry
> 
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Re: [Asterisk-Users] SS7 and Asterisk solution

[Matthew Boehm]

Eric,
  Thank you for explaining this to me instead of being rude and bitching at
me about my lack of GPL understanding.

Sincerely,
Matthew
- Original Message - 
From: "Eric Wieling aka ManxPower" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Friday, January 14, 2005 2:26 PM
Subject: Re: [Asterisk-Users] SS7 and Asterisk solution


> Matthew Boehm wrote:
> > So are you telling me that you cannot use other commercial products in
> > conjunction with asterisk?
>
> You cannot distribute a closed source add-on (except AGI) for Asterisk
> without a commercial license for Asterisk.  This is just standard GPL
> stuff, not Asterisk sprcific.
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Re: [Asterisk-Users] SS7 and Asterisk solution

[Matthew Boehm]

Time for you to learn some maners and not to bitch at/insult people for
something they don't understand. Time for you to stop telling people what to
do. Please add those 2 to you ToDo list.

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

Sent: Friday, January 14, 2005 2:19 PM
Subject: Re: [Asterisk-Users] SS7 and Asterisk solution


> On Fri, 2005-01-14 at 14:09 -0600, Matthew Boehm wrote:
> > So are you telling me that you cannot use other commercial products in
> > conjunction with asterisk?
>
> Time for you to go learn about the GPL. Time to go learn about proper
> trimming of an email. Time to learn how to use the archives for
> information already given. Time to go learn a lot of things. Please add
> those 3 to your ToDo list though.
>
> > - Original Message - 
> > From: "izo" <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> > Sent: Friday, January 14, 2005 1:32 PM
> > Subject: Re: [Asterisk-Users] SS7 and Asterisk solution
> >
> >
> > > On Fri, 14 Jan 2005 10:01:52 -0600, Matthew Boehm wrote:
> > > > Why does it have to be commercially licenced?
> > >
> > > Without it, the SS7 software would be linking to GPL software which
> > > means they would have
> > > to GPL the code too. So the only way to get commercial SS7 is to have
> > > it with commercial
> > > asterisk.
> > >
> -- 
> Steven Critchfield <[EMAIL PROTECTED]>
>
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Re: [Asterisk-Users] SS7 and Asterisk solution

[Eric Wieling aka ManxPower]

Matthew Boehm wrote:
So are you telling me that you cannot use other commercial products in
conjunction with asterisk?
You cannot distribute a closed source add-on (except AGI) for Asterisk 
without a commercial license for Asterisk.  This is just standard GPL 
stuff, not Asterisk sprcific.
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[Asterisk-Users] Strange CRCX

[Leonardo Tramontina]

Sirs,
 
I have the following situration:
1) AudioCodes Stretto 2000 media gateway running MGCP
2) E1 Digium card at a PC with Asterisk
3) My application running as Call Agent (CA) from Stretto 2000
 
 
| My app |--| Stretto 2000 |--| E1 card + Asterisk |
 
 
As my application is the CA of Stretto 2000, everything it sends
(RSIPs, acks, etc.) my app answers. And everything I send, Stretto
answers.
 
At extensions.conf (Asterisk) I have the following number of my
softphone in another PC:
exten => 123456,1,Dial(MGCP/aaln/[EMAIL PROTECTED])
 
When my app sends CRCX to Stretto, it answers with a "200
 OK, with session description", but the channel on the
E1 card is not occuped and the softphone doesn't ring. I was observing
the Asterisk debug screen, but nothing happens. It seems the
connection doesn't leave the Stretto 2000. Is it possible the media
gateway answer to the CA that created a connection, but doesn't make
the call?? Or is this a parameter at media gateway that I must set
(something like change "loopback" mode to "normal" mode?)??
Does anyone has already faced some situation like this?? I don't know
what could be happening...
The create connection I send is:
 
 
CRCX  TS/trunk#0/[EMAIL PROTECTED] MGCP 1.0
C: 987654321
L: p:20
M: sendonly
 
a=dialed:111222
a=called:333444
a=dialing:123456
 
 
 
The last 3 lines I set with "the called number, the number to which
the call was delivered and the calling number" (RFC 3435). Other SDP
parameters different from "a=", Stretto 2000 doesn't allow me to set.
 
 
Thanks in advance,
Leonardo
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[Asterisk-Users] SIP Registration problem, 403 forbidden

[Brian Chrystal]

trying to set up and configure a polycom soundpoint ip 500 phone,  when trying 
to get it to register with sip, i get the following message


Sip read:
REGISTER sip:67.110.252.13:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58
From: "138polycom" ;tag=B8D9FA39-9D85A6AC
To: 
CSeq: 1 REGISTER
Call-ID: [EMAIL PROTECTED]
Contact: ;methods="INVITE, ACK, BYE, 
CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0
Max-Forwards: 70
Expires: 300
Content-Length: 0


11 headers, 0 lines
Using latest request as basis request
Sending to 67.110.253.129 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58
From: "138polycom" ;tag=B8D9FA39-9D85A6AC
To: ;tag=as62b71d67
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 67.110.253.129:5060
Jan 14 11:44:49 NOTICE[3257]: chan_sip.c:8007 handle_request: Registration from 
'' failed for '67.110.253.129'
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Destroying call '[EMAIL PROTECTED]'


i'm kinda new to this stuff, so if you need to see any cfg files, let me know 
and i'll put them up,  thanks

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Re: [Asterisk-Users] SS7 and Asterisk solution

[Steven Critchfield]

On Fri, 2005-01-14 at 14:09 -0600, Matthew Boehm wrote:
> So are you telling me that you cannot use other commercial products in
> conjunction with asterisk?

Time for you to go learn about the GPL. Time to go learn about proper
trimming of an email. Time to learn how to use the archives for
information already given. Time to go learn a lot of things. Please add
those 3 to your ToDo list though.

> - Original Message - 
> From: "izo" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Friday, January 14, 2005 1:32 PM
> Subject: Re: [Asterisk-Users] SS7 and Asterisk solution
> 
> 
> > On Fri, 14 Jan 2005 10:01:52 -0600, Matthew Boehm wrote:
> > > Why does it have to be commercially licenced?
> >
> > Without it, the SS7 software would be linking to GPL software which
> > means they would have
> > to GPL the code too. So the only way to get commercial SS7 is to have
> > it with commercial
> > asterisk.
> >
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] SIP Registration problem, 403 forbidden

[Brian Chrystal]

trying to set up and configure a polycom soundpoint ip 500 phone,  when trying 
to get it to register with sip, i get the following message


Sip read:
REGISTER sip:67.110.252.13:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58
From: "138polycom" ;tag=B8D9FA39-9D85A6AC
To: 
CSeq: 1 REGISTER
Call-ID: [EMAIL PROTECTED]
Contact: ;methods="INVITE, ACK, BYE, 
CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0
Max-Forwards: 70
Expires: 300
Content-Length: 0


11 headers, 0 lines
Using latest request as basis request
Sending to 67.110.253.129 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58
From: "138polycom" ;tag=B8D9FA39-9D85A6AC
To: ;tag=as62b71d67
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 67.110.253.129:5060
Jan 14 11:44:49 NOTICE[3257]: chan_sip.c:8007 handle_request: Registration from 
'' failed for '67.110.253.129'
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Destroying call '[EMAIL PROTECTED]'


i'm kinda new to this stuff, so if you need to see any cfg files, let me know 
and i'll put them up,  thanks
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Re: [Asterisk-Users] Softphone for Linux recommendation

[Adam Fineberg]

Howard Lowndes wrote:
Can anyone _recommend_ a downloadable OSS softphone that _works_ under
Linux and is compatible with Asterisk.
So far I have tried kphone and linphone and had problems with both, and
I am still waiting to hear back from the X-Lite beta folks.
 

How about iaxcomm?
http://iaxclient.sourceforge.net/iaxcomm/
Adam
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RE: [Asterisk-Users] Call Parking

[Brian West]

Use valetparking :P

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kyle Hagan
> Sent: Friday, January 14, 2005 1:39 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Call Parking
> 
> I am using a manager app to do redirects, when I redirect to 700, for
> parking, the person on the other end hears the number its parked on.
> 
> How do I stop this?
> 
> Kyle
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Re: [Asterisk-Users] context wide variable scope

[Steven Critchfield]

On Fri, 2005-01-14 at 14:13 -0500, Jeremy Hinton wrote:
>   Maybe i missed this somewhere, but is it possible to define a variable 
> with a scope of the current context? I know i can define a system wide 
> variable, and i can define one that is valid for the duration of the 
> channel, but is it possible to define a variable that comes into scope 
> for every channel that comes into a context? I don't think so, but i 
> wanted to make sure.

You can if you use the DB functions, kind of. If you could better
describe an example of your problem, there may be better/other solutions
for you to use.  
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] SS7 and Asterisk solution

[Matthew Boehm]

So are you telling me that you cannot use other commercial products in
conjunction with asterisk?

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

Sent: Friday, January 14, 2005 1:32 PM
Subject: Re: [Asterisk-Users] SS7 and Asterisk solution


> On Fri, 14 Jan 2005 10:01:52 -0600, Matthew Boehm wrote:
> > Why does it have to be commercially licenced?
>
> Without it, the SS7 software would be linking to GPL software which
> means they would have
> to GPL the code too. So the only way to get commercial SS7 is to have
> it with commercial
> asterisk.
>
> m.
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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

[Bruno Hertz]

On Fri, 2005-01-14 at 16:27 -0200, Denis GalvÃo - iSolve wrote:
> Em Sex 14 Jan 2005 16:11, Dan escreveu:

> I dont have problems when calling PSTN extensions, and calling VoceMail,  
> EchoTest, etc. The problem is related with the conversation between two 
> DIAX Softphones.

With * in the middle or direct calls? I had problems with
iaxcomm -> * -> firefly communication when * attempted a transfer.
Huge latencies (10 sec or so). Might be a bug in the iaxclient library,
just don't know.

Regards, Bruno.


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Re: [Asterisk-Users] Softphone for Linux recommendation

[Bruno Hertz]

On Sat, 2005-01-15 at 05:37 +1100, Howard Lowndes wrote:

> Can anyone _recommend_ a downloadable OSS softphone that _works_ under
> Linux and is compatible with Asterisk.
> 
> So far I have tried kphone and linphone and had problems with both, and
> I am still waiting to hear back from the X-Lite beta folks.
> 

XLite isn't exactly OSS, isn't it? :)

I tried linphone, iaxcomm, gnomemeeting, and SJPhone. Pros and cons:

(1) linphone

Audio OK, but it doesn't send media when somebody calls me. Buggy for
me.

(2) iaxcomm

Generally good, sometimes crackly audio. When people call me with
firefly and * attempts a transfer, we experience huge audio delays in
the 10 sec range. Could be firefly or iaxcomm bug. Maybe both, as they
both use the iaxclient library. But as long as * sticks to notransfer
quite usable.

(3) gnomemeeting

Fairly good, but currently supports only h323. SIP support is underway.
Since I couldn't yet find a simple way to register with * without a
gatekeeper and have people call me, I'm currently not using it.

(3) SJPhone (for linux)

Not OSS, but for me best audio and latencies. Definitive con: there
seems to be no dial pad (i.e. dtmf interaction during call not
possible).

Especially, the OSS phone situation is not really bright, at least for
me.

Regards, Bruno.


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Re: [Asterisk-Users] Realtime / sip.conf

[Muhammad Rizwan Khan]

Brain:

I am still hanging with the same problem, although i tried this:
# iptables -t nat -A PREROUTING -p udp -i eth0 --dport 0 -j REDIRECT
--to-port 5060
from:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20grandstream%20budgetone#comments

But still have same problems?
Any how are you able to make call now?


On Fri, 2005-01-14 at 23:39, Brian S. Adelson wrote:
> Thank you everyone for your help.  It looks like my problem was
> related to:
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0003332
> 
> I patched and all is well now.
> 
> -brian
> 
> 
> On Fri, 14 Jan 2005 at 12:57 Brian S. Adelson ([EMAIL PROTECTED]) wrote:
> 
> > Muhammad,
> >   Could you possible share you configuration and what version of
> >   asterisk you are running (I am using the head version from today)
> >   Maybe this will shed some light on the problem that both of us are
> >   having.
> > 
> >   -Brian
> > 
> > 
> > On Fri, 14 Jan 2005 at 22:08 Muhammad Rizwan Khan ([EMAIL PROTECTED]) wrote:
> > 
> > > I also have setting in extconfig.conf file, and i am able to register
> > > users. The only difference is that i am using odbc instead of mysql in
> > > settings.
> > > The problem i am facing is that whenever i call (using Xlite) from one
> > > extension (e.g 12345) to another (e.g. 123456), my dialler shows me
> > > error 484: address incomplete. On the other hand whenever i call from
> > > one extension (e.g 123245) to the same (e.g 12345) it grings properly.
> > > Any idea what can be the problem here.
> > > 
> > > Thanks
> > > 
> > > On Fri, 2005-01-14 at 21:46, Brian S. Adelson wrote:
> > > > I do not recieve any debug messages for sip when extconfig is setup to
> > > > use sipfriends.  Here is my extconfig:
> > > > 
> > > > [settings]
> > > > 
> > > > realtime_ext => mysql,asterisk,extensions_table 
> > > > voicemail => mysql,asterisk,voicemail_table 
> > > > sipfriends => mysql,asterisk,sip_extensions 
> > > > 
> > > > As you can see, there is not much to it.  But when I do have
> > > > "sipfriends" enabled, then I am not able to register any phones etc.  
> > > > 
> > > > -Brian
> > > > 
> > > > 
> > > > On Fri, 14 Jan 2005 at 10:40 Matthew Boehm ([EMAIL PROTECTED]) wrote:
> > > > 
> > > > > If you only get debug messages when you use console then you don't 
> > > > > have
> > > > > something setup right in extconfig
> > > > > 
> > > > > -Matthew
> > > > > - Original Message - 
> > > > > From: "Brian S. Adelson" <[EMAIL PROTECTED]>
> > > > > To: "Matthew Boehm" <[EMAIL PROTECTED]>
> > > > > Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > > > 
> > > > > Sent: Friday, January 14, 2005 10:16 AM
> > > > > Subject: Re: [Asterisk-Users] Realtime / sip.conf
> > > > > 
> > > > > 
> > > > > > Sorry, thought I mentioned that.
> > > > > >
> > > > > > In the debug, I do not see it attemping to query the mysql database.
> > > > > > It only makes this attempt when i try to pull information via the
> > > > > > console:
> > > > > >
> > > > > >
> > > > > > *CLI> realtime load sipfriends name 155
> > > > > >Column Name  Column Value
> > > > > >      
> > > > > >   uniqueid  1
> > > > > >   name  155
> > > > > >   callerid  X-Line Phone
> > > > > >canreinvite  N
> > > > > >context  from-internal
> > > > > >   dtmfmode  rfc2833
> > > > > >   host  dynamic
> > > > > >mailbox  155
> > > > > >nat  no
> > > > > >   port  5060
> > > > > > secret  155
> > > > > >   type  friend
> > > > > >   username  155
> > > > > > regseconds  0
> > > > > >
> > > > > >
> > > > > > Jan 14 23:10:57 DEBUG[11834]: MySQL RealTime: Retrieve SQL: SELECT 
> > > > > > * FROM
> > > > > sip_extensions WHERE name = '155'
> > > > > > Jan 14 23:10:57 DEBUG[11834]: MySQL RealTime: Everything is fine.
> > > > > >
> > > > > >
> > > > > >
> > > > > > On Fri, 14 Jan 2005 at 10:12 Matthew Boehm ([EMAIL PROTECTED]) 
> > > > > > wrote:
> > > > > >
> > > > > > > What's in your debug?
> > > > > > >
> > > > > > > -Matthew
> > > > > > > - Original Message - 
> > > > > > > From: "Brian S. Adelson" <[EMAIL PROTECTED]>
> > > > > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > > > > > 
> > > > > > > Sent: Friday, January 14, 2005 9:47 AM
> > > > > > > Subject: [Asterisk-Users] Realtime / sip.conf
> > > > > > >
> > > > > > >
> > > > > > > >
> > > > > > > > I am currently in the process of testing out realtime support 
> > > > > > > > for
> > > > > > > > sip.conf.  I have followed all of the directions that are 
> > > > > > > > listed in
> > > > > > > > the Wiki, but for some reason this does not work.
> > > > > > > >
> > > > > > > 

[Asterisk-Users] app_conference compile?

[Matt Hess]

Has anybody compiled app_conference as of late?
I've already asked on the app_conference devel list but as I'm rather in 
a hurry my thinking is somebody here has both run into and found a way 
to get this compiled and running.

Using stable asterisk and the most recent app_conference from it's cvs 
on sourceforge..

begin:vcard
fn:Matt Hess
n:Hess;Matt
org:LiveWireNet
adr;dom:;;4577 Pecos St;Denver;CO;80211
email;internet:[EMAIL PROTECTED]
title:Senior Network Engineer
tel;work:303-458-5667
tel;fax:303-458-5725
x-mozilla-html:FALSE
url:http://www.livewirenet.com/
version:2.1
end:vcard

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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

[Denis Galvão - iSolve]

I tried IaxComm in two Linux boxes. Everything work fine, with USB Phones + 
IaxComm.

So, the problem should be related to Windows OS!?

Wich version of Windows are you using Dan!?

Denis.


Em Sex 14 Jan 2005 17:16, Denis Galvão - iSolve escreveu:
> Same problem with jitterbuffer=no
>
> I tried IaxComm, same problem of DIAX.
>
> This is related with iaxclient...
>
> Denis.
>
> Em Sex 14 Jan 2005 17:03, Dan escreveu:
> > Hi,
> >
> > \> Em Sex 14 Jan 2005 16:43, Dan escreveu:
> > >> > I dont have problems when calling PSTN extensions, and calling
> > >> > VoceMail, EchoTest, etc. The problem is related with the
> > >> > conversation between two DIAX Softphones.
> > >>
> > >> Between 2 DIAX phone and the delay is in one direction only??
> > >
> > > Yes. One direction only... Just who make the call get the delay.
> >
> > Then try
> > jitterbuffer=no
> > in iax.conf
> > to see if it solves this issue.
> >
> > BR,
> > Dan
>
> ___
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-- 
D e n i s   G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1206B
CEP: 80530-000 - Curitiba - PR
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Re: [Asterisk-Users] Setting channel display in SIP

[Eric Wieling aka ManxPower]

Howard Lowndes wrote:
I have actually got a bit more cunning that this by using sipgetheader()
and sipaddheader().
The default user name is "asterisk", hard coded in chan_sip.c, so what I
did was to use sipgetheader() to get the From: header, then I cut() it
at the ":" character and the "@" character and checked the string
between these two characters.  If the string was "asterisk" then I did
sipaddheader(From: ${PIECE_BEFORE}:[EMAIL PROTECTED]).
OK, so it adds a second From: header, but as it gets added after the
original it doesn't seem to matter because it works and
"replacement-string" is what gets displayed on the phone, which is what
I want.  I also don't see that the tag= in the header makes any
difference either.
Can anyone see any probs I am likely to encounter using this?
Well if you like killing a fly with a with a tactical nuke, then no 
there's nothing wrong with how you are doing it.  The rest of us just 
use the Caller*ID functions of Asterisk to accomplish it.
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[Asterisk-Users] Call Parking

[Kyle Hagan]

I am using a manager app to do redirects, when I redirect to 700, for 
parking, the person on the other end hears the number its parked on.

How do I stop this?
Kyle
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RE: [Asterisk-Users] PrePaid Applications

[Sebastian Atala]

Try with ASTCC is free.


Sebastian


-Mensaje original-
De: Bilal Ghayad [mailto:[EMAIL PROTECTED] 
Enviado el: Martes, 14 de Enero de 2003 14:56
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] PrePaid Applications

Hi;

Is the Prepaid Applications that we can use it with Asterisk are Free or we
have to pay?

Regards
Bilal

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Re: [Asterisk-Users] SS7 and Asterisk solution

[izo]

On Fri, 14 Jan 2005 10:01:52 -0600, Matthew Boehm wrote:
> Why does it have to be commercially licenced?

Without it, the SS7 software would be linking to GPL software which
means they would have
to GPL the code too. So the only way to get commercial SS7 is to have
it with commercial
asterisk.

m.
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[Asterisk-Users] context wide variable scope

[Jeremy Hinton]

	Maybe i missed this somewhere, but is it possible to define a variable 
with a scope of the current context? I know i can define a system wide 
variable, and i can define one that is valid for the duration of the 
channel, but is it possible to define a variable that comes into scope 
for every channel that comes into a context? I don't think so, but i 
wanted to make sure.

- jeremy
--
Jeremy Hinton A little nonsense
Senior Network Manager   now and then
Continental VisiNet Broadband   is relished by
[EMAIL PROTECTED]the wisest men
757 873 4500
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Re: [Asterisk-Users] PC to Phone

[Dan]

Hi,
- Original Message - 
From: "Bilal Ghayad" <[EMAIL PROTECTED]>
To: 
Sent: Tuesday, January 14, 2003 8:58 PM
Subject: [Asterisk-Users] PC to Phone


Can some one advise me an PC to Phone client software to be used under
Windows OS at the client side, to be communicated with Asterisk PBX?
Have you tried DIAX?
It is a full featured IAX software phone, distributed as a freeware:
http://www.laser.com/dante
http://www.geocities.com/tdanro
There is an online help file too if you want to learn more about it.
Best regards,
Dan
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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

[Denis Galvão - iSolve]

Same problem with jitterbuffer=no

I tried IaxComm, same problem of DIAX.

This is related with iaxclient...

Denis.

Em Sex 14 Jan 2005 17:03, Dan escreveu:
> Hi,
>
> \> Em Sex 14 Jan 2005 16:43, Dan escreveu:
> >> > I dont have problems when calling PSTN extensions, and calling
> >> > VoceMail, EchoTest, etc. The problem is related with the
> >> > conversation between two DIAX Softphones.
> >>
> >> Between 2 DIAX phone and the delay is in one direction only??
> >
> > Yes. One direction only... Just who make the call get the delay.
>
> Then try
> jitterbuffer=no
> in iax.conf
> to see if it solves this issue.
>
> BR,
> Dan
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RE: Re: [Asterisk-Users] Bristuff 0.20RC3 loses connectivity after short line interruption?

[Remco Barende]

Weird, I haven't actually tried that, that may be part of my problem too.
If i disconnect the line from the NT1 bristuff will not reconnect on every 
occasion. Disconnecting the cord between the Nt1 and the HFC-S card makes 
it lose connectivity occasionally. But I guess either way, the modules 
should restore the line connection.

Does anybody know if any of the Junghanns developers follow this list or 
should we e-mail them a bug report? (I briefly looked through the tarball 
but didn't see an e-mail address but I didn't browse the sources).

Groetjes,
Remco
P.S. Michiel : Do you have any experience connecting KPN PRI to *? I need 
to do that soon.

On Fri, 14 Jan 2005, Michiel van Baak wrote:
We have 2 HFC-S cards.
We have a very simular problem here.
restarting * means no more outgoing calls.
I first have to unload the modules, load them again and start asterisk.
plugging/unplugging cables from the cards dont give any problems here.
I fixed the asterisk reloading thing by altering the /etc/init.d/asterisk 
script.

OT: I have the kernel oops messages when unloading the module like described 
on voip-info.org

Michiel van Baak
Terrazur

- Originele Bericht -
Van: Remco Barende
Aan: Asterisk Users Mailing List - Non-Commercial Discussion Datum: 
Thursday, 13 January 2005, 18:39 Onderwerp: Re: [Asterisk-Users] Bristuff 
0.20RC3 loses connectivity after short line interruption?

Sorry I forgot to mention that. Its just a cheap ass HFC-S single BRI 
card (manufactured by E-Tech). I googled around and I know it can take 
some time to recover for the NT1 but I think this doesnt apply for s0.

Even after waiting for 10 minutes I do not get any connectivity but 
unloading and reloading the modules seems to solve the problem instantly.

I could even have a script do it as a really ugly way to solve it but I 
dont think there is any way for a script to know if the ISDN connection 
is lost or not.

Remco
On Thu, 13 Jan 2005, George Konstantoulakis wrote:
Same thing here,
I am using bristuff0.20-RC2b with an octoBRI card.
It only happens with DDI lines. With normal ISDN lines I dont have a 
problem.
Which card are you using ?

Remco Barende wrote:
I installed bristuff0.20-RC3 (which includes * 1.0.3 stable)
It works fine until I disconnect the phone jack for the ISDN line. 
Even after plug it back in asterisk still reports that it could not 
create a zap channel when I try to dial out and the line gives an 
engaged tone when I try to dial.

Re-starting asterisk doesnt solve this, I have to stop asterisk, 
unload the modules, reload the modules and start asterisk again.

I assume this is a bug, not a feature (should I e-mail it to 
Junghanns directly??)?

I know the telco here in holland and I will lose the line for a 
short period every once in a while and its annoying when the line 
doesnt come back up.

Or did I forget some setting to recover from such a situation?
Thanks!
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[Asterisk-Users] OT: zaptel kernel mod

[Matthew Boehm]

I don't have any card specific modules loaded. I don't have chan_zap loaded.
So how can the zaptel kernel module show usage? I just rebooted this server
due to defunct asterisk process.

How can I find out what 4 things are using zaptel?

[EMAIL PROTECTED] localhost]# lsmod
Module  Size  Used byNot tainted
parport_pc 19204   1  (autoclean)
lp  9188   0  (autoclean)
parport39072   1  (autoclean) [parport_pc lp]
autofs 13684   0  (autoclean) (unused)
acenic241092   1
e100   62340   1
zaptel182304   4
ext3   73376   2
jbd56336   2  [ext3]
megaraid   31212   3
aic7xxx   142548   0  (unused)
sd_mod 13452   6
scsi_mod  110488   3  [megaraid aic7xxx sd_mod]

Thanks,
Matthew

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[Asterisk-Users] Help in E1-T1 encoding

[Alejandro G]

Peter,

Sorry for the delay. I had to reconfigure all again. I do I inbound call to
asterisk and the result log is this (hope this is usefull):


Alejandro


Enabled EXTENSIVE debugging on span 1
*CLI> T203 counter expired, sending RR and scheduling T203 again
Sending Receiver Ready (27)

> [ 00 01 01 37 ]

> Supervisory frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000EA: 1
> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> N(R): 027 P/F: 1
> 0 bytes of data
-- Restarting T203 counter

< [ 00 01 01 37 ]

< Supervisory frame:
< SAPI: 00  C/R: 0 EA: 0
<  TEI: 000EA: 1
< Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
< N(R): 027 P/F: 1
< 0 bytes of data
-- ACKing all packets from 26 to (but not including) 27
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
-- Got RR response to our frame
-- Restarting T203 counter

< [ 02 01 36 36 08 02 05 00 05 04 03 80 90 a2 18 03 a9 83 81 6c 0c 00 80 31
31 34 37 38 35 33 38 37 37 70 0c 80 30 31 31 35 35 34 34 35 34 30 39 7d 02
91 81 ]

< Informational frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
< N(S): 027   0: 0
< N(R): 027   P: 0
< 47 bytes of data
-- ACKing all packets from 26 to (but not including) 27
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
< Protocol Discriminator: Q.931 (8)  len=47
< Call Ref: len= 2 (reference 1280/0x500) (Originator)
< Message type: SETUP (5)
< [04 03 80 90 a2]
< Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
<  Ext: 1  Trans mode/rate: 64kbps, circuit-mode
(16)
<  Ext: 1  User information layer 1: u-Law (34)
< [18 03 a9 83 81]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
 [ 02 01 01 38 ]

> Supervisory frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> N(R): 028 P/F: 0
> 0 bytes of data
-- Restarting T203 counter
-- Restarting T203 counter

> [ 00 01 36 38 08 02 85 00 02 18 03 a9 83 81 ]

> Informational frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000EA: 1
> N(S): 027   0: 0
> N(R): 028   P: 0
> 10 bytes of data
-- Restarting T203 counter
Stopping T_203 timer
Starting T_200 timer
> Protocol Discriminator: Q.931 (8)  len=10
> Call Ref: len= 2 (reference 1280/0x500) (Terminator)
> Message type: CALL PROCEEDING (2)
> [18 03 a9 83 81]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
>ChanSel: Reserved
>   Ext: 1  Coding: 0   Number Specified   Channel Type:
3
>   Ext: 1  Channel: 1 ]
-- Accepting call from '1147853877' to '01155335410' on channel 0/1,
span 1
-- Executing Ringing("Zap/1-1", "") in new stack

> [ 00 01 38 38 08 02 85 00 01 1e 02 81 88 ]

> Informational frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000EA: 1
> N(S): 028   0: 0
> N(R): 028   P: 0
> 9 bytes of data
T_200 timer already going (2)
> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 1280/0x500) (Terminator)
> Message type: ALERTING (1)
> [1e 02 81 88]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0:
0   Location: Private network serving the local user (1)
>   Ext: 1  Progress Description: Inband
information or appropriate pattern now available. (8) ]
-- Executing Wait("Zap/1-1", "4") in new stack

< [ 00 01 01 38 ]

< Supervisory frame:
< SAPI: 00  C/R: 0 EA: 0
<  TEI: 000EA: 1
< Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
< N(R): 028 P/F: 0
< 0 bytes of data
-- ACKing all packets from 26 to (but not including) 28
-- ACKing packet 27, new txqueue is 28 (-1 means empty)
-- Something left to transmit (28), restarting T200 counter

< [ 00 01 01 3a ]

< Supervisory frame:
< SAPI: 00  C/R: 0 EA: 0
<  TEI: 000EA: 1
< Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
< N(R): 029 P/F: 0
< 0 bytes of data
-- ACKing all packets from 27 to (but not including) 29
-- ACKing packet 28, new txqueue is -1 (-1 means empty)
-- Since there was nothing left, stopping T200 counter
-- Nothing left, starting T203 counter
-- Restarting T203 counter
 

Re: [Asterisk-Users] PRI concentrator

[Matthew Boehm]

> Yes,  it is called a Lucent MAX TNT or Cisco AS 5400 with DS-3 input

Perhaps you meant 5300? We have one of those and yes, it works fine. But
it costs about $20K for one.

> If you really want RJ-45 connections for your PRIs (ick) then you can

What else would you use to make a PRI cable other than RJ45?

-Matthew

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Re: [Asterisk-Users] REGISTER Problems With Realtime

[Matthew Boehm]

I have no idea how ODBC converts a prepared statement to MySQL when only up
until 4.1 did mysql support them. Have you tried using res_config_mysql
inside asterisk-addons?

-Matthew

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

Sent: Friday, January 14, 2005 12:24 PM
Subject: RE: [Asterisk-Users] REGISTER Problems With Realtime


> MySQL but its using the ODBC driver.
>
> 
>
> Michael Shuler, C.E.O.
> BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP)
> 682 High Point Lane
> East Peoria, IL 61611
> Office: (217) 585-0357
> Cell: (309) 657-6365
> Fax: (309) 213-3500
> E-Mail: [EMAIL PROTECTED]
> Customer Service: (877) 976-0711
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Matthew Boehm
> > Sent: Friday, January 14, 2005 9:56 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] REGISTER Problems With Realtime
> >
> >
> > Its trying to prepare a statement. What database are you using?
> >
> > -Matthew
> >
> > - Original Message - 
> > From: "Michael Shuler" <[EMAIL PROTECTED]>
> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> > 
> > Sent: Thursday, January 13, 2005 7:05 PM
> > Subject: [Asterisk-Users] REGISTER Problems With Realtime
> >
> >
> > Why is the SELECT statement below putting a "?" in for the
> > username?  I am
> > using today's CVS.
> >
> > Jan 13 18:48:41 WARNING[7570]: res_config_odbc.c:105
> > realtime_odbc: SQL
> > Execute error!
> > [SELECT * FROM sip_buddies WHERE name = ?]
> >
> >
> >
> > Full dump:
> >
> > Sip read:
> > REGISTER sip:198.88.216.85 SIP/2.0
> > Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK-ffmzndpfrao2;rport
> > From: "Mike's Peoria Snom"
> > ;tag=ljql3vjulo
> > To: "Mike's Peoria Snom" 
> > Call-ID: [EMAIL PROTECTED]
> > CSeq: 106321 REGISTER
> > Max-Forwards: 70
> > Contact: ;q=1.0
> > User-Agent: snom200-3.56m
> > P-NAT-Refresh: 15
> > Supported: gruu
> > Allow-Events: dialog
> > X-Real-IP: 192.168.1.2
> > WWW-Contact: 
> > WWW-Contact: 
> > Expires: 60
> > Content-Length: 0
> >
> >
> > 17 headers, 0 lines
> > Using latest request as basis request
> > Sending to 192.168.1.2 : 5060 (NAT)
> > Jan 13 18:48:41 WARNING[7570]: res_config_odbc.c:105
> > realtime_odbc: SQL
> > Execute error!
> > [SELECT * FROM sip_buddies WHERE name = ?]
> >
> > Transmitting (NAT):
> > SIP/2.0 403 Forbidden
> > Via: SIP/2.0/UDP
> > 192.168.1.2:5060;branch=z9hG4bK-ffmzndpfrao2;received=206.222.
> > 58.98;rport=32
> > 777
> > From: "Mike's Peoria Snom"
> > ;tag=ljql3vjulo
> > To: "Mike's Peoria Snom"
> > ;tag=as01156a34
> > Call-ID: [EMAIL PROTECTED]
> > CSeq: 106321 REGISTER
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > Contact: 
> > Content-Length: 0
> >
> >
> >  to 206.222.58.98:32777
> > Jan 13 18:48:41 NOTICE[7570]: chan_sip.c:8036 handle_request:
> > Registration
> > from '"Mike's Peoria Snom"
> > ' failed
> > for '206.222.58.98'
> > Scheduling destruction of call
> > '[EMAIL PROTECTED]' in
> > 15000 ms
> >
> > 
> >
> > Mike
> >
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

[Dan]

Hi,
\> Em Sex 14 Jan 2005 16:43, Dan escreveu:
> I dont have problems when calling PSTN extensions, and calling
> VoceMail, EchoTest, etc. The problem is related with the conversation
> between two DIAX Softphones.
Between 2 DIAX phone and the delay is in one direction only??
Yes. One direction only... Just who make the call get the delay.
Then try
jitterbuffer=no
in iax.conf
to see if it solves this issue.
BR,
Dan
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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

[Denis Galvão - iSolve]

Em Sex 14 Jan 2005 16:43, Dan escreveu:
> > I dont have problems when calling PSTN extensions, and calling
> > VoceMail, EchoTest, etc. The problem is related with the conversation
> > between two DIAX Softphones.
>
> Between 2 DIAX phone and the delay is in one direction only??

Yes. One direction only... Just who make the call get the delay.

> The phones are connected to the same PBX?

Yes they are in the same Asterisk.

> The problem is the same independent of the codec used?

Yes. I tried out all of the codecs available.

Driving me nuts...

Denis.


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Re: [Asterisk-Users] I Don't Want Asterisk in the Media Path

[Eric Wieling aka ManxPower]

Dhennys Pestana wrote:
I'm trying to find a way to connect two (or more) extensions directly without
being kept in the middle during the conversation but it won't happen.
Asterisk will always stay in the SIP signaling path.  It can get out of 
the RTP path (only way to really see this is using something like 
tcpdump since sip show channels shows the signaling not the RTP path). 
Asterisk CANNOT get out of the RTP path if you are using the "t" or "T" 
option to dial (maybe other options too) or if the codec for the two 
legs of the call are different.

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[Asterisk-Users] PrePaid Applications

[Bilal Ghayad]

Hi;

Is the Prepaid Applications that we can use it with Asterisk are Free or we
have to pay?

Regards
Bilal

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[Asterisk-Users] PC to Phone

[Bilal Ghayad]

Hi;

Can some one advise me an PC to Phone client software to be used under
Windows OS at the client side, to be communicated with Asterisk PBX?

Regards
Bilal

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Re: [Asterisk-Users] Cisco VIP30

[Ryan Laginski]

Hi,
I've never used those instructions, this is my skinny.conf, and I was
able to connect 3 12 sp+ (apparently the exact same as a VIP30 minus
the extra buttons) with different firmwares.


;
; Skinny Configuration for Asterisk
;
[general]
port = 2000 ; Port to bind to, default tcp/2000
bindaddr = 172.16.0.5   ; Address to bind to IMPORTANT, must be the ip
of asterisk, not 0.0.0.0
dateFormat = M-D-Y  ; M,D,Y in any order (5 chars max)
keepAlive = 120
callwaiting=no
allow = all
; disallow =


; Typical config for 12SP+
[somename]
device=SEP00D0BA847E6B ; change this to be your MAC of the phone, make
sure to keep SEP infront
version=P002F202; so the firrmware on this phone is 2.02, if
it was 2.04, put P002F204
callwaiting=no
context=home
callerid = "SomeName",<01>;
line => 402 ; Dial(Skinny/[EMAIL PROTECTED])


Next, plug in a phone, and press **
You should be able to program the ip address. For the TFTP server, put
in your asterisk machine.

If you need it to be dhcp:
Use the option "Next-Server"  with asterisk's ip in  your dhcpd.conf.
I also found by confirguring the phone manually, let it connect, then
change it back to dhcp, it will remember the asterisk ip.

Let me know if you have any problems.
-ry

On Fri, 14 Jan 2005 13:13:08 +0100, Pawel Jaskorzynski @ Sokolka
<[EMAIL PROTECTED]> wrote:
> Hi,
> I am trying to run Cisco VIP30 phone with Asterisk. Here:
> http://www.voip-info.org/wiki-Configuring+Cisco+12SP+phones+with+Asterisk
> I have found some info on the setup- does anybody have the right firmware
> for the phone, namely the P002L2J2.bin file? Or, maybe, any hints as to
> where it could be found?
> 
> I would greatly appreciate any help.
> 
> Greetings,
> Pawel
> 
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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

[Dan]

I have modified the CallMe feature for DIAX to provide an Echo test.
Just use it with 0.9.9g and see the result. To pass the explanation or to
end the echo test just press '#'. You can still leave me  a message after
that.
I got the echo test. The result was fine, just a very SHORT delay, but
nothing like my problem.
I dont have problems when calling PSTN extensions, and calling VoceMail,
EchoTest, etc. The problem is related with the conversation between two
DIAX Softphones.
Between 2 DIAX phone and the delay is in one direction only??
The phones are connected to the same PBX?
The problem is the same independent of the codec used?
BR,
Dan
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Re: [Asterisk-Users] Realtime / sip.conf

[Brian S. Adelson]

Thank you everyone for your help.  It looks like my problem was
related to:

http://bugs.digium.com/bug_view_page.php?bug_id=0003332

I patched and all is well now.

-brian


On Fri, 14 Jan 2005 at 12:57 Brian S. Adelson ([EMAIL PROTECTED]) wrote:

> Muhammad,
>   Could you possible share you configuration and what version of
>   asterisk you are running (I am using the head version from today)
>   Maybe this will shed some light on the problem that both of us are
>   having.
> 
>   -Brian
> 
> 
> On Fri, 14 Jan 2005 at 22:08 Muhammad Rizwan Khan ([EMAIL PROTECTED]) wrote:
> 
> > I also have setting in extconfig.conf file, and i am able to register
> > users. The only difference is that i am using odbc instead of mysql in
> > settings.
> > The problem i am facing is that whenever i call (using Xlite) from one
> > extension (e.g 12345) to another (e.g. 123456), my dialler shows me
> > error 484: address incomplete. On the other hand whenever i call from
> > one extension (e.g 123245) to the same (e.g 12345) it grings properly.
> > Any idea what can be the problem here.
> > 
> > Thanks
> > 
> > On Fri, 2005-01-14 at 21:46, Brian S. Adelson wrote:
> > > I do not recieve any debug messages for sip when extconfig is setup to
> > > use sipfriends.  Here is my extconfig:
> > > 
> > > [settings]
> > > 
> > > realtime_ext => mysql,asterisk,extensions_table 
> > > voicemail => mysql,asterisk,voicemail_table 
> > > sipfriends => mysql,asterisk,sip_extensions 
> > > 
> > > As you can see, there is not much to it.  But when I do have
> > > "sipfriends" enabled, then I am not able to register any phones etc.  
> > > 
> > > -Brian
> > > 
> > > 
> > > On Fri, 14 Jan 2005 at 10:40 Matthew Boehm ([EMAIL PROTECTED]) wrote:
> > > 
> > > > If you only get debug messages when you use console then you don't have
> > > > something setup right in extconfig
> > > > 
> > > > -Matthew
> > > > - Original Message - 
> > > > From: "Brian S. Adelson" <[EMAIL PROTECTED]>
> > > > To: "Matthew Boehm" <[EMAIL PROTECTED]>
> > > > Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > > 
> > > > Sent: Friday, January 14, 2005 10:16 AM
> > > > Subject: Re: [Asterisk-Users] Realtime / sip.conf
> > > > 
> > > > 
> > > > > Sorry, thought I mentioned that.
> > > > >
> > > > > In the debug, I do not see it attemping to query the mysql database.
> > > > > It only makes this attempt when i try to pull information via the
> > > > > console:
> > > > >
> > > > >
> > > > > *CLI> realtime load sipfriends name 155
> > > > >Column Name  Column Value
> > > > >      
> > > > >   uniqueid  1
> > > > >   name  155
> > > > >   callerid  X-Line Phone
> > > > >canreinvite  N
> > > > >context  from-internal
> > > > >   dtmfmode  rfc2833
> > > > >   host  dynamic
> > > > >mailbox  155
> > > > >nat  no
> > > > >   port  5060
> > > > > secret  155
> > > > >   type  friend
> > > > >   username  155
> > > > > regseconds  0
> > > > >
> > > > >
> > > > > Jan 14 23:10:57 DEBUG[11834]: MySQL RealTime: Retrieve SQL: SELECT * 
> > > > > FROM
> > > > sip_extensions WHERE name = '155'
> > > > > Jan 14 23:10:57 DEBUG[11834]: MySQL RealTime: Everything is fine.
> > > > >
> > > > >
> > > > >
> > > > > On Fri, 14 Jan 2005 at 10:12 Matthew Boehm ([EMAIL PROTECTED]) wrote:
> > > > >
> > > > > > What's in your debug?
> > > > > >
> > > > > > -Matthew
> > > > > > - Original Message - 
> > > > > > From: "Brian S. Adelson" <[EMAIL PROTECTED]>
> > > > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > > > > 
> > > > > > Sent: Friday, January 14, 2005 9:47 AM
> > > > > > Subject: [Asterisk-Users] Realtime / sip.conf
> > > > > >
> > > > > >
> > > > > > >
> > > > > > > I am currently in the process of testing out realtime support for
> > > > > > > sip.conf.  I have followed all of the directions that are listed 
> > > > > > > in
> > > > > > > the Wiki, but for some reason this does not work.
> > > > > > >
> > > > > > > When utilizing a flat file, I am able to register endpoints 
> > > > > > > without
> > > > > > > any problems, and calls can proceed.  One interesting side effect 
> > > > > > > that
> > > > > > > I have noticed is that when I am using realtime for sip, I am 
> > > > > > > unable
> > > > > > > to see any debug messages on the console (sip debug). By just
> > > > > > > commenting out the sipfriends line in extconfig.conf the problem 
> > > > > > > goes
> > > > > > > away.
> > > > > > >
> > > > > > > I do have the system utilizing realtime for Voicemail and 
> > > > > > > Extensions,
> > > > > > > and I do not have any problems.  Has anyone seen this prob

[Asterisk-Users] Softphone for Linux recommendation

[Howard Lowndes]

Can anyone _recommend_ a downloadable OSS softphone that _works_ under
Linux and is compatible with Asterisk.

So far I have tried kphone and linphone and had problems with both, and
I am still waiting to hear back from the X-Lite beta folks.

-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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[Asterisk-Users] Radius on *

[Tenorio, Leandro]

 
I'm currently trying to use a Radius server for acct and auth, cause
much of our systems are using it.
Anyone has an asterisk server working with Radius Auth and Acct? 
 
Tkx, LTenorio
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[Asterisk-Users] T100P with NEC C2400 IPX switch

[Jerry Geis]

I am looking at interface the T100P with a NEC C2400 IPX.
Has this been done before by anyone???
quick search did not bring anything up.
Thanks,
Jerry
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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

[Denis Galvão - iSolve]

Em Sex 14 Jan 2005 16:11, Dan escreveu:
> I have modified the CallMe feature for DIAX to provide an Echo test.
> Just use it with 0.9.9g and see the result. To pass the explanation or to
> end the echo test just press '#'. You can still leave me  a message after
> that.

I got the echo test. The result was fine, just a very SHORT delay, but 
nothing like my problem.

I dont have problems when calling PSTN extensions, and calling VoceMail,  
EchoTest, etc. The problem is related with the conversation between two 
DIAX Softphones.

Thanks for your help.

Denis.
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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

[Dan]

DIAX and iaxComm both use the iaxclient library, which has been in flux 
lately:
lots of new features added.  If your iaxComm is version 0.99pre4 or later, 
then
I *think* it might be using a more bleeding-edge version of the iaxclient
library.

Some of the recent library work has been on the jitterbuffer code, which is 
much
more likely to produce the kind of results you describe than portaudio 
latency
tuning.

When:
Jon -> call -> Fred
Fred listen Jon without problems, but Jon listen Fred with 10 seconds of
delay.
When:
Fred -> call -> Jon
Jon listen Fred wihtout problems, but Fred listen Jon with 10 seconds of
delay
With Firefly Softphone(IAX2) I dont get this problem, everything works
great.
For the people with this problem, please try to use another version of
the diax.dll which can be downloaded from:
http://www.laser.com/dante/diax/wiax_jb_old.zip
or
http://www.geocities.com/tdanro/diax/wiax_jb_old.zip
This use an older version of the jitter buffer.
Please send me your feedback.
Best regards,
Dan

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