Also you should never call an application directly with an call file as
the CDR info won't get updated correctly Link your app to an extension
then call it like that.
bkw
On Mon, 19 Jan 2004, Marcin Kuzmicki wrote:
> Cytowanie Charles Hatchette <[EMAIL PROTECTED]>:
>
> > I'm trying to devise
echo cancellation is activated in /etc/asterisk/zapata.conf
However, how to confirm it?
Does "zap show channel 1" confirm the existence of echo cancellation?
--
David Kwok
Iaxtel/FWD # 17001813482
smime.p7s
Description: S/MIME Cryptographic Signature
Use account codes. That works ALOT better. If you require passwords then
look at app_authenticate.
bkw
On Tue, 20 Jan 2004, Ing Isianto Istiadi wrote:
>
> Dear all,
> I have a questions:
> 1. I have 3 FXO connected to 3 analog phones. But I have 5 users using those
> phone. I want to be able t
Cytowanie Charles Hatchette <[EMAIL PROTECTED]>:
> I'm trying to devise a way to playback more than one file per call when I
> copy my file 'Test.call' into .. var/spool/asterisk/outgoing
>
>Channel: Zap/1/put_your_phone_number_here
>Application: Playback
>Data: demo-thanks + a-second
Can you clarify this? Does it or doesn't it work?
bkw
On Mon, 19 Jan 2004, Asterisk User Group wrote:
> I had been running an older patched CVS to get VOIP working with NAT and
> everything had been running fine. I just built * on a new box with
> CVS-01/18/04-12:19:25. And now I can get remo
Hi
In ChangeLog the following is written down:
Asterisk 0.7.1
-- Fixed timed include context's and GotoIfTime
-- Fixed chan_h323 it now gets remote ip properly instead of 127.0.0.1
But all the same where that the bells and vanish as of it gets rid...
== New H.323 Connection created.
[EMAIL PROTECTED] <> wrote:
> I am running a few asterisk servers with 512M RAM memory, and
> as I have
> mentioned in previous notes, I have experienced frequent
> crashes when faced
> with more than 15-20 simultaneous calls. I have tried to find
> out if it
> could be due to (a) Xeon chip runni
Walter Doerr wrote:
On Mon, Jan 19, 2004 at 12:33:41PM +0100, Philipp von Klitzing wrote:
Ok,
here comes part two of the log quiz, this time SIP not MGCP:
WARNING[8201]: chan_sip.c:4821 handle_response: Got 200 OK on REGISTER
that isn't a register
This is most probably cause by registration
Hello Matt,
Is that the Wildcard TE410P you are using. Digium said that it had some
problems with Redhat 9.0 is that correct?
- Digium quad T1 card
- 3 T1's (2 x B8ZS ESF Long Distance and 1 x robbed-bit SF local)
- Redhat 9.0
Aram Ter-Martirosyan
Senior Account M
Hi all,
I just now receive the FXO X101P Card but can't at any way make then call out.
I can hear the signal, even call but always receive from my local operator error that or the number don't exist or need more numbers.
I play alot with txgain and rxgain, but none help me out.
Being honest i t
Dear all,
I have had an experience which I would run by all of you to see if this is
normal.
I am running a few asterisk servers with 512M RAM memory, and as I have
mentioned in previous notes, I have experienced frequent crashes when faced
with more than 15-20 simultaneous calls. I have tried t
Steve.
You are saying this from your view of 2004.
But at the time R2 was developed there were no microcontrollers and tones
were decoded with LC filters.
R2 provides interactive capabilities base on a simple tones protocol to
retrieve ANI, dialed numbers,
signalling status etc. It's compelled stru
Seems like you have experienced a few problems implementing this type of
configuration. My first hand experiences have been more positive then
yours. Could you share more with the list what went wrong? Also IP
phones connected to ports configured with Cisco portfast have never
caused me any pr
On Tue, 20 Jan 2004 01:37:07 +0100, Philipp von Klitzing
<[EMAIL PROTECTED]> wrote:
>Hi!
>
>> I've applied Steve Sokol's patch the the IAXClient source, and the IAX2 noanswer
>> bug is solved in iaxComm, as well.
>>
>> Win32 and Linux binaries are available at
>> http://iaxclient.sourceforge.net/
Patent licensing is usually, like this:
a) direct licensing if you rolled your own implementation of the specs
b) indirect - if you bought an implementation from another vendor, case
in point Digium licenses the G.729a binary from Voiceage.
Cheers
Eric Wieling wrote:
On Mon, 2004-01-19 at 14:01
I had been running an older patched CVS to get VOIP working with NAT and
everything had been running fine. I just built * on a new box with
CVS-01/18/04-12:19:25. And now I can get remote SIP users to register.
Has anything major changed...
[general]
port = 5060 ; Port to bin
the announcement/bugfix.
Earlier on that page is the announced version:
http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-20040119.zip
If you use the 20040114 version, make sure you are calling it with IAX1
protocol.
If you are using the 20040119 version, make sure you are calling it with IAX2.
> Although the OS may cache that information, the userland process
> can take quite some time to process a very full directory. I've had
> this happen quite a few times with Linux ext2 filesystems, where the
> fileglob * exceeded bash's limit of 32,768 characters. /bin/ls on
> those directories
Hello,
Our max for a single machine is 40 concurrent SIP -> Zap conversations for
about a 12 hour period and over 5000 total phone calls per day. We didn't
see crashes going over that, but we wanted to be safe and now have 2
identical machines handling upto about 30 concurrent SIP -> Zap calls(300
Dear all,
I have a questions:
1. I have 3 FXO connected to 3 analog phones. But I have 5 users using those
phone. I want to be able to log who is using the phones and where to. I'd
like to use password for each user so that I can keep track who is the
caller and for how long.
I read the authentica
So am I to assume this is not possible?
Can someone let me know one way or another, or just at least flame me
for asking?
Hello List
I have searched the lists, the wiki and the handbook and see how to use
distinctive ring inside however I can't find incoming.
I have 1 x100p and 2 phone numbers,
Aastra will have a production PT480i SIP phone in March for ~US180-$200.
Same phone as ADSI model just SIP, but has 4 extra buttons for virtual
lines. Got a beta SIP model under test. Designed for SIP v1 & v2. * is
one of PBX used for testing by development, so should be * friendly when
release
Andrew Thompson wrote:
- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, January 19, 2004 1:49 PM
Subject: Re: [Asterisk-Users] SS7 over Asterisk ?
The funny thing I see is
http://www.tdmoip.com/Home/0,,5,00.html
Compairs voip to TDMoIP... v
Hello
I've been running Asterisk on a 1GHz Dell: 512 Meg RAM, T100P with a
PRI, for 9 months, NO problems.
Ken
On Jan 19, 2004, at 4:50 PM, PBXtech wrote:
Are DELL PC's worthwhile to run asterisk on?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Olle E. Johansson wrote:
LQ (Asterisk) wrote:
Hi guys,
I was reading that Steve Underwood is working on Asterisk R2 signalling
support, and has the 95% of the work done.
What is R2? I'm curious.
Half of R2D2, of course.
Its also a stupid clunky multi-tone based telephone signaling system
wid
It's an old trunking signaling protocol,
It's inband, and the worst thing about it, apart from
how slow it is, because it's base on pulses, is that every
country has it's own implementation of it.
Luciano
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Olle
> ...or http://search.voip-forum.com
> Indexes our lists, the Wiki, asterisk.org and some related sites.
>
> Still experimental. Any feedback to me, please.
> /O :-)
Thanks! I'll use that myself. In the past I've been heavily involved in search
engine development that is particularly suited to se
Quoting [EMAIL PROTECTED]:
> > Why wouldn't you just use your existing Ethernet
> > infrastructure putting
> > the IP phones inline between the wall jack and the PC? There are a
> > number of IP phones that have builtin switch/hub that allows
> > the PC to
> > daisy chain off the IP phone.
>
Hi!
> I've applied Steve Sokol's patch the the IAXClient source, and the IAX2 noanswer
> bug is solved in iaxComm, as well.
>
> Win32 and Linux binaries are available at
> http://iaxclient.sourceforge.net/iaxcomm/index.html
Ok, solved it: First installed the IAX1 version, then installed the IAX2
One other option, although I hate to suggest it but will admit to
having done it a few times...
Ethernet uses only two twisted pair. Cat-5 wire has four pair
inside. You can crimp two RJ45 terminals on one lenght of CAT5
wire. It works and remains CAT5 if you keep the pairs twisted
all the way
Has anyone setup the Asterisk platform to run on a 64 bit Solaris box with
multiple processors to get more calls out of Asterisk??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jesse Peterson
Sent: Monday, January 19, 2004 12:23 PM
To: [EMAIL PROTECTED]
Hi!
> I've applied Steve Sokol's patch the the IAXClient source, and the IAX2 noanswer
> bug is solved in iaxComm, as well.
Very good.
> Win32 and Linux binaries are available at
> http://iaxclient.sourceforge.net/iaxcomm/index.html
Problem: Crash on Win 98 SE if I try to access the menu "Prefe
Many thanks to Mike, Steve and all the IaxComm and Diax users who
contributed to tracking down and solving this problem.
AJ
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To UNSUBSCRIBE or updat
Hmmm. I can't get it to accept any calls at all now (Windows 2000,
binary size
-rw-r--r--1 root root 546816 Jan 14 22:16 iaxcomm.exe
:-( But maybe
I'm doing something wrong. The client says that it registers. And the
client itself can call my other phones, but I can't call the clie
On Mon, 2004-01-19 at 14:01, Bob Knight wrote:
> Scott Stingel wrote:
>
> >>What *I* want to know is why someone has not made a CHEAP PCI card
> with
> >>4, 8, or 16 of these DSPs on it. This kind of card would provide
> >>
> >>
> >>
> >
> >Expanding a bit on Nicolas' message, DSP software is
How does Grandstream become patent indemnified for their hardware? I
would assume they did not pay for a license for G723,1 and G729 directly
to the patent holding company. Maybe they did. I always assumed the
indemnification came with a DSP that implemented the codec.
--Eric
On Mon, 2004-01-1
- Original Message -
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, January 19, 2004 2:50 AM
Subject: [Asterisk-Users] IAX2 bug in DIAX solved - Great Thanks to Steven!
> Hi all,
>
> Thanks to Steven Sokol great work, the IAX2 bug in DIAX is now solved.
> For th
On Monday 19 January 2004 13:37, Andrew Kohlsmith wrote:
> > This is actually a bad idea. While many filesystems today have
> > binary tree directory structures, some still do not. Allowing
> > too many miscellaneous sounds in a single directory is not only
> > difficult to browse, it may also co
Matthew Branton wrote:
Hi Everyone,
So I have been further exploring the integration of our asterisk server
and our lucent definity g3si system. I took the suggestion of setting up
an isdn-pri line added the two way tie trunk and the signalling group,
but can't seem to get the PRI signalling w
Hmmm. The one on voip forum searches more than the asterisk mailing list.
The one mail-archive can't do date or sender restricted searches.
I take your point that there are searching facilities, but sometimes it would
be nice to focus more, otherwise I'd simply use google. Which I do a lot.
Howeve
On Mon, 2004-01-19 at 18:38, Olle E. Johansson wrote:
> LQ (Asterisk) wrote:
>
> > Hi guys,
> >
> > I was reading that Steve Underwood is working on Asterisk R2 signalling
> > support, and has the 95% of the work done.
>
> What is R2? I'm curious.
A type of signaling for E1 lines.
--
Nicolas
Is the SIP bin same for IAX as well?
Kannaiyan
- Original Message -
From: "Christian Stredicke" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, January 19, 2004 7:08 PM
Subject: RE: [Asterisk-Users] SNOM IAX image
For those who are using snom 200 phones, I think we have a pro
Andrew Kohlsmith <[EMAIL PROTECTED]> writes:
>> Why wouldn't you just use your existing Ethernet infrastructure putting
>> the IP phones inline between the wall jack and the PC? There are a
>> number of IP phones that have builtin switch/hub that allows the PC to
>> daisy chain off the IP phone.
I'm trying to devise a way to playback more than one file per call when I
copy my file 'Test.call' into .. var/spool/asterisk/outgoing
Channel: Zap/1/put_your_phone_number_here
Application: Playback
Data: demo-thanks + a-second-file + a-third-file
Is there some way to do this?
Charlie H
Thanks all who replied, I think you've gotten me on my way.
Over the next few days, while I fiddle with the system I'm testing at
home, I'll try to churn out some documentation regarding my setup &
configuration that may be helpful to someone. I'll submit my notes to
the wiki when I'm ready.
Any
Looks like the list server is really lagging tonight. I found out some
more info so will just post it in a new email with the same subject.
I added: "search => freenum.org" to enum.conf and got a match (SIP
system) when doing the lookup Maybe I overlooked that in the
original instruction
I received a good response that recommends Kernel 2.4.0, but is that very
old version? Anyone out there who has had experience with running 0.7.1 with
a particular version of Kernel and Redhat?
THanks
Tommy
-Original Message-
From: T. Chan [mailto:[EMAIL PROTECTED]
Sent: Monday, January
Thanks, Matt !
So, am I correct in assuming that there are quite a few (or alot) of us who
have had not so good experiences with Asterisk? That Asterisk would crash
after it hit a certain number of calls or after a certain period of time
with 15-20 calls? I understand that there were others who we
Andrew wrote:
> > Why wouldn't you just use your existing Ethernet infrastructure putting
> > the IP phones inline between the wall jack and the PC? There are a
> > number of IP phones that have builtin switch/hub that allows the PC to
> > daisy chain off the IP phone.
>
> To quote myself:
>
> >>
Are DELL PC's worthwhile to run asterisk on?
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A better option and one Asterisk desperately needs is some kind of
--lint option,
Which would check the config for errors and useless misspelled options.
I personal find one or more typos or misspelling a month, On my PBXs.
Eric Wieling wrote:
Maybe someone will write a patch to print an error
Steven Critchfield wrote:
On Mon, 2004-01-19 at 05:19, Kim Hendrikse wrote:
Is there a search engine for this list?
Google
Use site:lists.digium.com to limit the search to just the list server.
...or http://search.voip-forum.com
Indexes our lists, the Wiki, asterisk.org and some related sites.
LQ (Asterisk) wrote:
Hi guys,
I was reading that Steve Underwood is working on Asterisk R2 signalling
support, and has the 95% of the work done.
What is R2? I'm curious.
/O
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[EMAIL PROTECTED]
http://lists.digium.com/mail
I've been pretty satisfied with the Aastra PT480.
There are some other people that say they don't like them, but I think
the $110-$120 ea. Works great for our office and the people I install
for.
Take it for what you paid for it.
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, January 19, 2004 1:49 PM
Subject: Re: [Asterisk-Users] SS7 over Asterisk ?
> The funny thing I see is
>
> http://www.tdmoip.com/Home/0,,5,00.html
>
> Compairs voip to TDMoIP... very funny. I
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Dustin Goodwin
> Sent: Monday, January 19, 2004 11:18 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] ADSI phone vs. IP phone
>
>
> Why wouldn't you just use your existing Ethernet
> in
Hi,
I've been having a hard time getting the absolutetimeout feature to work.
I've search all the messages in the news letters and tried what was
suggested and still have not gotten it to work. Below is a portion of my
extensions.conf and sip.conf. I've also been running these test on ver 0.5.0
e
I agree that it should be able to do more than 15 to 20 calls when NOT
transcoding, however, I WAS doing pass-through without any transcoding and
it was crashing after around 15 to 20 calls, that was the problem, while I
was expecting at least hundreds of simultaneous calls ( not channels ) doing
p
I appreciate all your feedbacks, but they seems to have diverted from my
original question which was
I have been using Asterisk "10 days ago" version loaded onto my Redhat 7.3
with kernel 2.4.18-3 running Jeremy's h323 driver. It has been running okay
with a bit of problems, like system crashing a
I'd like to know how everyone else is going about getting correct CDR
information for calls. Right now I notice that if a call come in and gets
parked the CDR info doesn't how the correct info on who picked that call up,
also when someone transfer a call there isn't a new record being made so the
d
It will suite you well to fumble around with asterisk for several days and to keep
reading all the documentation tidbits you can find. That will really help get you
aquainted with asterisk and the support/documentation that is available. Most of the
good info I've found has come from the wiki an
Scott Stingel wrote:
What *I* want to know is why someone has not made a CHEAP PCI card with
4, 8, or 16 of these DSPs on it. This kind of card would provide
Expanding a bit on Nicolas' message, DSP software is complex, and there is
not a huge number of people who do it well. So along with
I've applied Steve Sokol's patch the the IAXClient source, and the IAX2 noanswer
bug is solved in iaxComm, as well.
Win32 and Linux binaries are available at
http://iaxclient.sourceforge.net/iaxcomm/index.html
Any feedback is appreciated.
___
Asterisk-U
> This is actually a bad idea. While many filesystems today have binary
> tree directory structures, some still do not. Allowing too many
> miscellaneous sounds in a single directory is not only difficult to
> browse, it may also consume inordinate amounts of CPU, memory, and
> user time attempti
Top posting(sorry) then imbedding the answers to your questions. Otherwise
doesn't make sense.
Thanks for your reply. Sorry it took a while to get the answers. I'm in
Germany and your email came last night just as I was headed to the rack.
Robert
John Todd said:
>>
>>>
>>my sip.conf contains:
>>
Hi!
> Ok, sure. That's I guess somewhat like I've been doing now. The reason that
> I ask, is that I can provide one.
Not really needed, but thanks for the offer. Look here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ
Cheers, Philipp
__
I have not looked at products from every company but I do know a few
offer 100mbps FastEthernet connections to the switch and the PC.
- Dustin -
Andrew Thompson wrote:
- Original Message -
From: "Dustin Goodwin" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, January 19, 2004 11
For those who are using snom 200 phones, I think we have a promising image
now ready at http://snom.com/download/share. Its version number is 2.03m.
Please check this image; it should fix the known issues. The release notes
can be found at http://www.snom.com/snom200_release_notes_de.php. If
everyt
> It was my impression that these phones had 10MB ehternet connections and not
> 100MB. Not that most of us would notice the difference in browsing the net,
> it does defeat the purpose of having 100MB switches. (I believe I also saw
> people on this list talking about strange things happening whe
Hi,
I am trying to use the RoutCall application.
Do you guys have any more info on RouteCall info.
In particular what all those fields in the database should be used for?
Ta
SJ
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> It was my impression that these phones had 10MB ehternet connections and not
> 100MB. Not that most of us would notice the difference in browsing the net,
> it does defeat the purpose of having 100MB switches. (I believe I also saw
> people on this list talking about strange things happening when
> Why wouldn't you just use your existing Ethernet infrastructure putting
> the IP phones inline between the wall jack and the PC? There are a
> number of IP phones that have builtin switch/hub that allows the PC to
> daisy chain off the IP phone.
To quote myself:
>> True, but I don't have to re
The funny thing I see is
http://www.tdmoip.com/Home/0,,5,00.html
Compairs voip to TDMoIP... very funny. I use voip and I don't loose any
PBX features(and I sure as hell don't burn 11kbps) very comical.
bkw
On Mon, 19 Jan 2004, CW_ASN - Gus wrote:
> See http://www.rad.com/ , TDM-over-IP so
Title: Lucent and ISDN-PRI
Hi Everyone,
So I have been further exploring the integration of our asterisk server and our lucent definity g3si system. I took the suggestion of setting up an isdn-pri line added the two way tie trunk and the signalling group, but can't seem to get the PRI signal
On Mon, 2004-01-19 at 10:52, Jeremy Jones wrote:
> Hi folks,
>
> The obligatory newbie disclaimer:
>
> "Hi, I'm new to Asterisk and I have a couple questions..."
>
> OK, now that that's over with:
>
> I've just started working for a small CLEC, and I'm trying to sell * to
> my boss as a replace
Hi!
> Maybe , I never tried TDMoE ...
> Where can I found a documentation or at least a sample for doing that ?
http://www.asteriskdocs.org/current/docs-pdf/hgta.pdf
page 29
Note that this book is still in pre-alpha state...
Philipp
___
Asterisk-User
To be clear I meant using Chan)_h323 with Call Manager where CM is
configured
with * as a H.323 gateway, not client.
CM supports H.323 to direct calls through gateways, and in fact until
recently
that is all they supported. They now also have MGCP, but only to their
IOS
platforms, and SIP is comi
Hi!
> > I've just noticed that since swapping from the direct mysql
> > cdr driver to cdr_odbc, the call duration (and anything else
> > that's an integer) isn't logged - anyone else seen this and
> > know the reason. The cdr_odbc driver gives no error messages
> > and records any string base
On Mon, Jan 19, 2004 at 08:30:14AM -0800, Kostur, Andre said:
> OK, I'm having some trouble finding which equipment I need
>
> What I'd like to do is take about a dozen incoming analog lines and bring
> them into an * server. Of course one is going to have a hard time fitting a
> dozen X100P
On Mon, 2004-01-19 at 10:41, Andrew Kohlsmith wrote:
> > I have 3 Digium X100P cards, and I'm sure there must be some way of
> > configuring zapata.conf to allow each line to identify itself with a
> > different Caller ID string.
>
> You cannot set outgoing caller ID on PSTN lines. PRI only. For
I've tried to use that script, but the phones seem to ignore it. I am in
the process of upgrading to 6.1 on the phones, maybe they will behave like
they're supposed to.
B. J.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Friday, Ja
Title: Channel Banks
Well, you have several options. A T100P and a
device such as a Adtran Altlas or simpler Channel bank. But since at this time
as you point out Digium only has 1 FXO port per PCI slot(FYI I hear they
are working on a 4 port per PCI slot). The other options are MediaTrix,
V
We were seeing hanging symptoms when the dns entries in resolv.conf were
not reachable. Don't know if this applies to you.
Tan
telappliant.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: 19 January 2004 17:21
To: [EMAIL PROT
>
> What *I* want to know is why someone has not made a CHEAP PCI card with
> 4, 8, or 16 of these DSPs on it. This kind of card would provide
>
Expanding a bit on Nicolas' message, DSP software is complex, and there is
not a huge number of people who do it well. So along with the board layou
Title: Channel Banks
We use
an Adtran Atlas 500 for this job (not for * but for our Mitel ICP 3300) you can
aggregate FXO to T1 / PRI or any which way you want. It's a killer box and very
easy to work with. Adtran support is, in a word, phenomenal. Very pricey,
but ebay has some 800 models:
On Monday 19 January 2004 08:34, Eric Wieling wrote:
> On Sun, 2004-01-18 at 22:22, [EMAIL PROTECTED] wrote:
> > It will probably be impossible to divide audio clips into
> > different directories without duplication of clips or massive
> > headaches determining direcories. My suggested method of h
Hi!
> Has anyone experienced * hang/exit when issuing -
> asterisk -r -x reload
Yes, see also here and add your comments if applicable:
http://bugs.digium.com/bug_view_page.php?bug_id=725
Philipp
P.S.: Next time please open a new top posting when you create a new topic
instead of replying
- Original Message -
From: "Dustin Goodwin" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, January 19, 2004 11:17 AM
Subject: Re: [Asterisk-Users] ADSI phone vs. IP phone
> >
> > I'm looking at ADSI phones simply because I don't have to re-tool my
entire
> > building; I can use
Ok, sure. That's I guess somewhat like I've been doing now. The reason that
I ask, is that I can provide one. I write search engine software and would
be happy to set one up, but I can't host it. Google is good as a general
purpose search engine it's a fact, but with the software in the context
of
Hi guys,
I was reading that Steve Underwood is working on Asterisk R2 signalling
support, and has the 95% of the work done.
I was trying to contact him, on-list and off-list, and didn't receive any
answer.
Does anybody know something about his project or know a release date?
Thanks in advance,
Pa
Hi folks,
The obligatory newbie disclaimer:
"Hi, I'm new to Asterisk and I have a couple questions..."
OK, now that that's over with:
I've just started working for a small CLEC, and I'm trying to sell * to
my boss as a replacement for some expensive/inflexible/closed-source
software he's been u
Apologies, I've got it to work.
I didn't realise by just specifying the channels individually and resetting
the Caller ID before each channel would work.
Regards,
Steve
On Mon, Jan 19, 2004 at 03:59:54PM +, Steve Foy wrote:
> Hi there,
>
> I'm wondering if there is a way to assign a differe
Look for the recent 'capacity testing' thread here. We've had some
discussions on it, but so far the bottom line sounds like you won't
be able to run more than 20 - 25 decent quality calls before
asterisk dies.
jesse
[snip]
Your statement relies completely on assumptions which may be
incorrec
> I have 3 Digium X100P cards, and I'm sure there must be some way of
> configuring zapata.conf to allow each line to identify itself with a
> different Caller ID string.
You cannot set outgoing caller ID on PSTN lines. PRI only. For INCOMING
caller ID (i.e. prefixing the received number) yes y
On Mon, 2004-01-19 at 05:19, Kim Hendrikse wrote:
> Is there a search engine for this list?
Google
Use site:lists.digium.com to limit the search to just the list server.
--
Steven Critchfield <[EMAIL PROTECTED]>
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Title: Channel Banks
OK, I'm having some trouble finding which equipment I need
What I'd like to do is take about a dozen incoming analog lines and bring them into an * server. Of course one is going to have a hard time fitting a dozen X100P cards in a case, so an alternative would be
On Mon, Jan 19, 2004 at 08:44:36AM -0600, Eric Wieling wrote:
> On Mon, 2004-01-19 at 02:34, Nicolas Bougues wrote:
> > These are quite cheap components (the most expensive part is the $6
> > DSP).
>
> What *I* want to know is why someone has not made a CHEAP PCI card with
> 4, 8, or 16 of these D
Why wouldn't you just use your existing Ethernet infrastructure putting
the IP phones inline between the wall jack and the PC? There are a
number of IP phones that have builtin switch/hub that allows the PC to
daisy chain off the IP phone.
- Dustin -
I'm looking at ADSI phones simply because
Eric Wieling ([EMAIL PROTECTED]) wrote:
> CallManager uses Cisco's own SCCP aka Skinny Protocol, not H323.
> Asterisk has two SCCP channel drivers available. One is included with
> Asterisk, one is available for download from somewhere (check the
> mailing list archives). I don't know if they wo
> Is there a search engine for this list?
www.google.com, search for what you want and say "site:lists.digium.com" at
the end of your search terms.
Regards,
Andrew
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