snacktime wrote:
permit to be used for their contributions.. They won't be happy unless
everyone else does things their way. They wouldn't be happy if asterisk
was BSD or MIT licensed either.
No that's not true. I myself would be perfectly happy with an MPL.
However, because Asterisk is ava
On 10/9/05, Florian Overkamp <[EMAIL PROTECTED]> wrote:
snacktime wrote:> permit to be used for their contributions.. They won't be happy unless> everyone else does things their way. They wouldn't be happy if asterisk> was BSD or MIT licensed either.
No that's not true. I myself would be perfectl
Hello
I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom
IP300 phones.
My config files look like this:
features.conf
pickupextn = *8
zapata.conf
context=frompstnisdn
group=1
callgroup=1
pickupgroup=1
I also edited sip.conf like this:
group=1
callgroup=1
pickupgroup=1
Shouldn't it be pickupexten = *8
instead of pickupextn = *8 ?
Regards
Guido Hecken
>
> Hello
>
> I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom
> IP300 phones.
>
> My config files look like this:
>
> features.conf
> pickupextn = *8
>
> zapata.conf
> context=fromps
Yes sadly a typo on my part. It is pickupexten in features.conf
Any other ideas?
Angus
- Original Message -
From: "Guido Hecken" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, October 09, 2005 12:54 PM
Subject: RE: [Asterisk-Users] *
Hello
did you noticed that
http://www.asterisk.org is just pointing
to a web cvs directory
best
regards
Thierry
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On Sun, Oct 09, 2005 at 12:32:12PM +0100, Angus Comber wrote:
> Hello
>
> I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom
> IP300 phones.
I figure you use bristuff. Are you aware of app_pickup that comes with
it?
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
htt
asterisk-users
who has implemented callback function?
thanks
how to do ?
oncemore
[EMAIL PROTECTED]
2005-10-09
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I am astounded by the total lack of integrity people have displayed here.
Digium gave you Asterisk, and yet you turn around and stab them in the back.
As this is the Asterisk Users mailing list and this product will cease to be
Asterisk the moment it is forked, I don't really want to see any more
> Interesting. In their meeting minutes
> (http://wiki.openpbx.org/tiki-index.php?page=Meeting+Minutes+10-5-2005)
> I see that a BKW was elected to the board. Is this Brian West?
LOL!!! And the truth comes out. Children throwing their toys because they
don't have enough power...
--
Cheers,
M
Tony Mountifield wrote:
> Yes, it looks like the main people behind it are bkw, anthm and moc.
> They will be a great loss to the Asterisk community if they go off and
> only do their own thing.
I'm not sure I agree with that. If your friend stabs you in the back, is it
really a great loss if the
On Sun, 9 Oct 2005 21:32, Angus Comber wrote:
Hi
I have Polycom 600s and 500s but I find that we need to dial *8 then send. If
we pickup then dial *8 the phone or Asterisk re-aranges it to 8*. Likewise
with *97 and *98 foes to 9*7 and 9*8.
This might help.
> Hello
>
> I have a Junghanns ISDN
Matt Riddell a écrit :
Tony Mountifield wrote:
Yes, it looks like the main people behind it are bkw, anthm and moc.
They will be a great loss to the Asterisk community if they go off and
only do their own thing.
I'm not sure I agree with that. If your friend stabs you in the back, is
On Sun, 9 Oct 2005 13:44, Rudolf Ladyzhenskii wrote:
Hi
Use Koolstart. It is incorrect for Aust as we use Loopstart but it does not
work so Koolstart works.
Sample
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-pstn
signalling=fxs_ls
*PLONK*
--
Cheers,
Matt Riddell
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http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
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Matt Riddell a écrit :
*PLONK*
I was only stating the obvious... sorry you don't like it.
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On Sun, October 9, 2005 15:31, Matt Riddell said:
> *PLONK*
>
> --
> Cheers,
>
> Matt Riddell
>
Is that the sound of you dropping out of this list? It can't be a reply to
the previous poster's e-mail, as that was in fact a completely correct
statement...
But back to the topic: I can see the reaso
Very funny discussion :)
Same thing arrives with a lot of Gpl softwares,
A few months ago, it was another Voip software wich forked (Sip Express
Router aka Ser), there were pro and cons, now the discussion is finished and
they all cooperate.
Olivier
-Message d'origine-
De : [EMAIL PROTEC
Thanks for all the
feedback! I have posted the latest fax2mail and mail2fax scripts on the
site www.generationd.com
For those new to the
scripts, they are a friendly interface to asterisk to make it easy to
send/receive faxes by email. They now include email confirmations of
fax submis
Matt Riddell <[EMAIL PROTECTED]> wrote:
>I am astounded by the total lack of integrity people have displayed here.
Isn't that a bit over the top? If you have a license that permits you
to do something, and then you do it, what is the issue?
Doug
--
Doug Meredith ([EMAIL PROTECTED])
SystemGuard
Hi All,
Just a quick question, but I could really use some help on this one.
I've got the CVS-Head of * installed and running, and am using a VoiceTronix
OpenSwitch12 to connect to 12 analouge lines. I've got callerid activated by
the Telco, and can get callerid using a std phone. However, usi
On Sun, October 9, 2005 15:39, Ian Bonham said:
> Hi All,
>
> Just a quick question, but I could really use some help on this one.
>
> I've got the CVS-Head of * installed and running, and am using a
> VoiceTronix
> OpenSwitch12 to connect to 12 analouge lines. I've got callerid activated
> by
> t
Hi all!
I'm running a SuSE Linux 9.3 on a little book pc which is based on a VIA CPU
and Chipset:
cat /proc/cpuinfo
processor : 0
vendor_id : CentaurHauls
cpu family : 6
model : 7
model name : VIA Samuel 2
stepping: 3
cpu MHz : 532.776
cache size
On Sunday 09 October 2005 09:08, Matt Riddell wrote:
> Digium gave you Asterisk, and yet you turn around and stab them in the
> back. As this is the Asterisk Users mailing list and this product will
> cease to be Asterisk the moment it is forked, I don't really want to see
> any more spamming from
Andrew Kohlsmith wrote:
> Actually Digium gave the core of Asterisk. There is a *lot* of code in there
> that didn't come FROM Digium, but rather that Digium has incorporated and
> made a part of Asterisk.
Of course!! Well understood and agreed.
> Further, I think the *vast* majority of the is
Thanks John.
I can't seem to see if just applying the Asterisk side of the fix will
correct things though. The card I'm using is a VoiceTronix OpenSwitch 12.
I'm using the vpb driver as opposed to the Digium drivers in this instance.
Any clues?
Thanks,
Ian
From: "John Crowhurst" <[EMAIL
I have the following setup:
Asterisk Server
Sip Software Phones
and a Wholesale connection
At the moment, i can receive and send calls through the wholesale connection to and from the SIP phones.
Now i want to collect the CallerID from the wholesale connection, and pass it on to the SIP phone
On Sunday 09 October 2005 11:36, Matt Riddell wrote:
[ issues with Asterisk development ]
> But why couldn't it have been brought into the public forum and discussed?
It has been. Over, and Over, and Over again. On here, on -dev and on IRC.
Many times.
[openpbx design changes]
> Hmmm I don't
> Tony Mountifield wrote:
> > Yes, it looks like the main people behind it are bkw, anthm and moc.
> > They will be a great loss to the Asterisk community if they go off and
> > only do their own thing.
>
> I'm not sure I agree with that. If your friend stabs you in the back, is
it
> really a g
Hi All,
I have aquired a Cisco uBR900 voip router and was
wondering if anyone had a working config for it and the asterisk
configs.
I have a reletive new verson of the CVS tree and
have oh-h.323 installed.
Best regards,
Todd Reese
___
--Ba
On Sat, Oct 08, 2005 at 10:43:28PM -0400, Paul wrote:
> Steve Underwood wrote:
> >>
> >It's not harder. It's just different. A number of things have similar
> >requirements. The ISDN4Linux folk have certain versions of their
> >software approved by the telecoms bodies in Europe. They need to tie
I need to have a
bash script trigger Asterisk to goto a particular extension (as if that
extension were dialed). I can't find any documentation for
this.
Does anyone know of
a way for a bash script to Dial an asterisk extension? (I'm trying to make
a bash file call the app_rxfax applicat
On Sat, 2005-10-08 at 14:57 +0200, [EMAIL PROTECTED] wrote:
> thanks for that, i knew already but it misses the actual version
Oh yes, that new version. All it introduces is a digital signature on
the firmeware, for use w/ the new bootrom and such that require
digitally signed applications. (thi
I set up my * server using its publc IP address. Now that i switch over to using the domain name, X-Lite can't log in. =With Domain Name (doesnt work)
Transmitting (NAT) to 85.250.206.46:6007: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.250.179.93;branch=z9hG4bK-d87543-8953821
Mike M wrote:
On Sat, Oct 08, 2005 at 10:43:28PM -0400, Paul wrote:
Steve Underwood wrote:
It's not harder. It's just different. A number of things have similar
requirements. The ISDN4Linux folk have certain versions of their
software approved by the telecoms bodies in Europe. They ne
I tried both
exten => *8,1,PickUP()
and
exten => *8,1,PickUp(1)
But got:
-- Accepting voice call from '7768385144' to '787367' on channel 0/1,
span 1
-- Executing Dial("Zap/1-1", "SIP/200&SIP/202|20") in new stack
-- Called 200
-- Called 202
-- SIP/202-f041 is ringing
-- S
On Sun, Oct 09, 2005 at 05:15:36PM +0200, [EMAIL PROTECTED] wrote:
>
> Hi all!
>
> I'm running a SuSE Linux 9.3 on a little book pc which is based on a VIA CPU
> and Chipset:
> cat /proc/cpuinfo
> processor : 0
> vendor_id : CentaurHauls
> cpu family : 6
> model : 7
On Sun, Oct 09, 2005 at 01:34:11PM -0400, Technical Support wrote:
> I need to have a bash script trigger Asterisk to goto a particular extension
> (as if that extension were dialed). I can't find any documentation for this.
>
> Does anyone know of a way for a bash script to Dial an asterisk exte
No that's not problem.
On my current configs I get:
Oct 9 20:43:18 NOTICE[2990]: chan_sip.c:7455 handle_request: Nothing to
pick up
every time I try *8
Why does the phone think there is nothing to pickup?
Angus
- Original Message -
From: "Alan Harrison" <[EMAIL PROTECTED]>
To:
Hi,
recently I have tested my asterisk and I discovered that asterisk transfer
call even I not use option t,T in cmd dial. So it is problem with right
charge for call transfer. Situation:
1) Somebody A calling to asterisk user B
2) Asterisk user transfer call to somebody C
3) A and B talking but I
Did you make sure in sip.conf that both the called extensions and the
extension trying to pick up are part of the same group?
In your case it means that both sip/200 and sip/202, as well as the
phone where you are dialing *8 from should all be the same group
numbers.
On 10/9/05, Angus Comber <[EMA
Reading the patents and the comments written here, I couldn't resist
and had to make this comment (ouch it's Sunday again):
We should patent dugy style, and when Sprint screws Verizon from the
back, sue them both for patent infringement.
On 10/7/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
>
> >
While at this subject I would like to post this question regarding ADSI.
If I use a channel bank (like the Adit 600) with a digium T1/E1 card
connected to asterisk, can I still get to use ADSI phones? with all
the fun stuff?
If the answer is yes, if I go ahead and configure the Adit 600 to use
a CM
Does anybody know if Avaya has a test SIP firmware available for 4620
and 4640 IP phones? The 46xx SIP image from their website is a combo
download with SIP for the 4602, and h323 for the the 4620 and 4640.
It looks like they demo'd a SIP image for the 4640 as far back as 2004:
http://www.sip.org
Look at setgroup checkgroup
On 10/4/05, Crystal Stream, Incorporated <[EMAIL PROTECTED]> wrote:
> Hello,
> I have a block of 25 DIDs and have 10 phones on the
> network. I want when a person tries to call out for *
> to pick a number for the CIDN and I want to make sure
> that the number isn't dup
I have app_rxfax and
app_txfax loading properly (they are in my modules.conf and they show as loading
without error when I start asterisk)
I create .call files
(in the outgoing directory) and associated tiff files but app_txfax never acts
on them! They just sit there forever. I checked t
>sip show settings
Our auth realm sip1.sippal.com Realm. auth: No
Anyone know how to get this to say Yes?
Dan
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Does Asterisk work with Avaya? If so, is there any documentation on it?
Thanks,
Michael
On 10/9/05, Andy Vega <[EMAIL PROTECTED]> wrote:
> Does anybody know if Avaya has a test SIP firmware available for 4620 and
> 4640 IP phones? The 46xx SIP image from their website is a combo download
> wit
Chris... thanks for the great reply
On 10/5/05, Chris Shaw <[EMAIL PROTECTED]> wrote:
> Michael,
>
> Doing an All-Network setup is completely doable but there are many factors
> to consider.
>
> First of all, I didn't see any mention of how many connections it takes
> before Asterisk starts hav
The asterisk box has been working for about 20 days now, just recently
when someone dials into the box the and press 1 to goto general
enquiries, there is a pause that can last from 10 seconds through
to 1 minute before it dials the extentions, I have looked at the
console and no errors are showi
The DS3 currently available from Sangoma doesn't do channelized voice, yet. You wouldn't be able to use it with Asterisk, yet.
On 10/8/05, Jason Walker <[EMAIL PROTECTED]> wrote:
Has anyone used the DS3 card from Sangoma with Asterisk?I have read many posts from users that the Sangoma cards have b
Can someone who is knowledgable in the traditional telco space please give me a
layman's explanation (or point me to an appropriate url) of LBO as per the
zaptel configuration file?
# The line build-out (or LBO) is an integer, from the following table:
# 0: 0 db (CSU) / 0-133 feet (DSX-1)
# 1:
http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci211613,00.html
- Original Message -
From: "Rod Bacon" <[EMAIL PROTECTED]>
To:
Sent: Sunday, October 09, 2005 6:42 PM
Subject: [Asterisk-Users] Zaptel Line Build Out
> Can someone who is knowledgable in the traditional telco
http://www.stellenboersen.de/stellenboersen/
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To UNSUBSCRIBE or u
STINK
- Original Message -
From:
Danny
To: asterisk-users@lists.digium.com
Sent: Sunday, October 09, 2005 7:21
PM
Subject: [Asterisk-Users] Link
http://www.stellenboersen.de/stellenboersen/
___--Bandwi
This was just a recent personal experience
Maybe I missed a thread on this:
We recently installed asterisk (CVS-HEAD) on a debian system using 2.6
kernel and the enhanced RTC for all timing.
Also a custom compiled kernel for the CPU on the box (P4).
We had a strange thing happen in that with
I will be out of the office starting 10/10/2005 and will not return until
15/10/2005.
Dear Sir / Mdm,
I'm currently on a Overseas Business Trip.
During this period of time, I have minimal access to internet and email
cccess. As such, I might not be able to reply to your queries promptly. I
apo
Hi all,
I was wondering if anyone knows of a Asterisk Voice
Mail USER MANUAL.
With a tree breakdown of the options?
With descriptions.
I suppose I could write one myself, but why re-invent
the wheel.
I did some googling, but could not find anything.
I find it strange
Maybe I need to be a little more specific.
I know what signal attenuation is. What I don't know, is how LBO (and
specifically the implementation of it as used in the zaptel hardware/software)
helps the situation.
My servers are co-located with my carrier, and my PRI circuits are run through
Channelized DS3 from Sangoma is on their near-term roadmap from what I
have been told.
Cory Andrews
Senior Partner
+++
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
+++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
fax - 716.630.1548
BJ Weschke wrote:
The DS3 curren
Koolstart - see attached
Rudolf Ladyzhenskii wrote:
Hi, all
I have installed TDM400 with 1 FXS and 1 FXP ports.
Now I am goig through documentation on how to configure it.
It mentions 3 protocols: Loopstart, Groundstart and Koolstart. Which one
do I use?
Can someone send me sample zaptel.con
http://www.adc.com/Library/Techpub/80348_1.pdf?refer=Library&C=Copper_Connectivity&L=DS1_E1_Twisted_Pair_Products
http://www.pcmag.com/encyclopedia_term/0,2542,t=DSUCSU&i=42059,00.asp
any help?
> Maybe I need to be a little more specific.
>
> I know what signal attenuation is. What I don't
On 10/9/05, Michael Stearne <[EMAIL PROTECTED]> wrote:
Does Asterisk work with Avaya? If so, is there any documentation on it?Thanks,Michael
It does:
http://www.voip-info.org/tiki-index.php?page=Avaya+4602+configuration
Unfortunately, the SIP firmware only exists for the entry-level 4602 model,
The VoIP Connection is growing and we have opportunities for talented
individuals with Asterisk and general VoIP experience.
We have a need for assistance with the following activities:
1) Level one and two customer support
2) Asterisk custom application development
3) Asterisk product developme
try this link http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain
Shayne
On 10/10/05, asterisk <[EMAIL PROTECTED]> wrote:
Hi all,
I was wondering if anyone knows of a Asterisk Voice
Mail USER MANUAL.
With a tree breakdown of the options?
With description
Yeah... sorta.
So the CSU settings may be used when the E1 is pulled down to my premises, and I
have a short cable connecting directly to the CSU device. I don't know why I'd
need to change the LBO settings in that case, but I guess that doesn't really
matter to me at the moment.
In my case,
On 10/9/05, Andy Vega <[EMAIL PROTECTED]> wrote:
> On 10/9/05, Michael Stearne <[EMAIL PROTECTED]> wrote:
> > Does Asterisk work with Avaya? If so, is there any documentation on it?
> >
> > Thanks,
> > Michael
>
>
> It does:
> http://www.voip-info.org/tiki-index.php?page=Avaya+4602+configuration
Hi all,
I've got a TE410p connected via T1 to four channel banks (2 FXO and 2 FXS), no
PRIs.
Some users are complaining that they hear clicks and pops on the FXS lines,
generally when they pick up the phone it's noisy. This happens only after a
while, e.g. after a fresh restart of everything,
On Sun, Oct 09, 2005 at 01:51:41PM -0400, Paul wrote:
> Mike M wrote:
> >
> Mike, the context was regarding security by obscurity. It has nothing to
> do with stealing a product to sell to others. The only reverse
> engineering I ever did had nothing at all to do with bootlegging or
> counterfei
Danny wrote:
> http://www.stellenboersen.de/stellenboersen/
Can we get this person removed from the list. This is pure spam.
--
Cheers,
Matt Riddell
___
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http://www.sineapps.com/rssfeed.ph
On Sun, 9 Oct 2005 14:28:14 -0600, you wrote:
The initial release of Avaya's SIP firmware for the 4620 phone was
released on August 17, 2005. It is available from support.avaya.com.
That said, I have had mixed results with this phone. I'm sure it
works with Avaya gear, but it's glitchy with *.
Hi Obelix,
You can check that network is working fine. The ipaddresses in
sip.conf are correct.
/Gurmi
On 10/9/05, Obelix <[EMAIL PROTECTED]> wrote:
>
>
> I have been trying to connect via sip and things don't seem to work. What do
> messages like this mean?
>
> Oct 9 00:33:57 WARNING[2284
On Sun, Oct 09, 2005 at 07:37:41PM -0400, Steve Gladden wrote:
> The debian package installs something else called "mpg321" and creates
> an alias or symlink called mpg123 to mpg321.
Get the package mpg123 from non-free
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il
Mike M wrote:
On Sun, Oct 09, 2005 at 01:51:41PM -0400, Paul wrote:
Mike M wrote:
Mike, the context was regarding security by obscurity. It has nothing to
do with stealing a product to sell to others. The only reverse
engineering I ever did had nothing at all to do with bootlegging or
- Original Message -
From: "asterisk" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, October 11, 2005 12:15 AM
Subject: Re: [Asterisk-Users] Re: www.openpbx.org
The other thing that I think many are missing is the recent deal with
In
Hi
I am a newbie to * and I am having a problem which appears strange as I did not find any mention of it anywhere in my search.
Simply speaking, I have an external SIP proxy server which I am trying
to configure for incoming and outgoing calls from my asterisk
installation. So here is my con
On Mon, 2005-10-10 at 14:16 +0800, Craig Guy wrote:
> The practical part of the exam showed a distinct USA bias - It was in terms
> of T1's and analog zap extensions. I am from Australia, and the exam was in
That is ok, most of this list seems to be the same way regarding the
US/North American
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