Is probably not the best way to handle it, but you could store all sounds
in one directory and then create another directory that has subdirectories
like weather. The items that are most frequently used would then be
symlinked to the original sound directory. Just another way of organizing
it.
I have also been trying to research this exact same problem, but as yet
have not had much luck!
I'm hoping there is a solution, as at the moment, it makes transfering
calls to another PBX user impossible.
Let me know if you find a solution!
Cheers,
Steven
This is what DUSTIN WILDES at Sun,
On Mon, 19 Jan 2004 at 06:26, Franz Edler wrote:
From: Dustin Knuttgen on Sunday, January 18, 2004 11:47 PM
We tried to use SuSE initially and had no luck compiling zaptel on
either 8.2 or 9.0. We even had Digium take a look. After working
on it for days we finally switched to Red Hat 9.
Hi Iain,
-Original Message-
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
Hi Franz,
On Sun, 18 Jan 2004 at 21:47, Franz Edler wrote:
Now I learned, that I have to provide also the kernel-sources for
compiling zaptel. I have done that,
have you also copied the running kernel's configuration to the kernel
sources
zcat /proc/config.gz /usr/src/linux/.config
Hi,
-Original Message-
How do I pass the flash button to the PBX? It seems the
ATA-186 wants to control the flash by putting my first call
on hold and prompts me to dial another extension. DTMF is
fine, just can't use the native Flash functions of our PBX
with the ATA-186 and
Paul Mahler wrote:
I didn't think any of the manufacturers are shipping native serial ATA disks
yet. I think all the disks have hardware to convert from IDE to serial ATA,
thus there is no real advantage yet, just greater expense.
Paul,
I have just got hold of a couple of Seagate SATA disks
Hi all,
Thanks to Steven Sokol great work, the IAX2 bug in DIAX is now solved.
For the interested people, you can download the new DLL (just the IAX2
version) from the following location:
http://www.laser.com/dante/diax/wiax2.zip
Replace the wiax2.dll file in the app directory with the new
T. Chan wrote:
Dear All
Should one enable HT in the chip when running Asterisk or if we don't, would
that offer alot less processing power?
T
I have read before that HT did not help Asterisk so should be dissabled,
but as the chipsets and other hardware get better at using and
controlling HT
We can provide DID in all over southern California.
Thanks,
Aram Ter-Martirosyan
Senior Account Manager
Hi-Tech Gateway, Inc.
http://www.hi-teck.com
1225 Grand Central Ave.
Glendale, CA 91201
[EMAIL PROTECTED]
tel 818.546.4601
fax 818.546.4617
Turning Technology Into Business
On Sat, Jan 17, 2004 at 04:34:34PM +0200, Alexandru Coseru wrote:
All I'm trying right now is to get raw data from the E1 (from each
timeslot) , transmit it to another asterisk server and push it to the other
E1..
Doesn't TDMoE do that (provided that you're on the same subnet) ?
--
On Mon, Jan 19, 2004 at 01:28:02AM +, Robert Murray wrote:
Hi
Has anyone opened up a grandstream phone or handytone ATA to find out what is
inside? What is the CPU? How much RAM?
The HandyTone 286 features :
- 1 Mb Flash
- 256 Kb SRAM
- a TI TMS320VC5402 100 MHz DSP
- an RTL8019AS
Hi,
Anyone know how to set up tftp server for grandstream.
I gues it should be somethink like
tftpserver-dir
mac-address
firmware.bin
config.txt
Is this correct ?
And how should the config-file look like. ?
I had search sipphone.com but did'nt find anything.
/HHA
Hans,
Attached is the config file I send to my Grandstream.
Change IP address Phone ID to suite.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik
Andresen
Sent: 19 January 2004 08:43
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
I have asterisk running on SuSE 8.2. I'm using zaptel with no problems.
Steven.
On Sunday 18 January 2004 23:04, Reinhard Max wrote:
On Mon, 19 Jan 2004 at 06:26, Franz Edler wrote:
From: Dustin Knuttgen on Sunday, January 18, 2004 11:47 PM
We tried to use SuSE initially and had no
I'd recommend 2.4.0.
Steven.
On Sunday 18 January 2004 21:35, T. Chan wrote:
Dear All,
Based on your experience and knowledge, which Redhat (7.3, 8 or 9) and
which kernel is most stable and reliable running the 0.7.1 version of
Asterisk?
Thanks
Tom
---
Outgoing mail is certified
Hi,
Is it possible to connect a BRI isdn line to a E100P PRI card?
The location where I want to use it has a BRI line an will
switch to PRI in 6 month.
regards,
mark
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Olle E. Johansson wrote:
I have been compiling information on this configuration onto the Wiki:
http://voip-info.org/wiki-Asterisk+cisco+FXO
I can call out to the PSTN just fine, but inbound calls all appear in
my default [bogon-calls] context, not in [pstn-incoming]
As I understand it, the Cisco
On Mon, Jan 19, 2004 at 09:20:33AM -, David J Carter wrote:
Hans,
Attached is the config file I send to my Grandstream.
Change IP address Phone ID to suite.
That's great. Is it documented somewhere ?
And how do you manage tens or hundreds of phones ? Are they all in the
same
Thanks.
How is the directory structure ?
or do you add all you phone to the one file cfg.txt and have it in the root
of your tftp-dir ?
/HHA
Attached is the config file I send to my Grandstream.
Change IP address Phone ID to suite.
Maybe , I never tried TDMoE ...
Where can I found a documentation or at least a sample for doing that ?
Second , there is a small problem... Their are not on the same subnet, but
this can be fixed(i hope) using tunneling..
Regards
Alex
- Original Message -
From: Nicolas Bougues
On Mon, Jan 19, 2004 at 10:44:29AM +0100, [EMAIL PROTECTED] wrote:
Hi,
Is it possible to connect a BRI isdn line to a E100P PRI card?
The location where I want to use it has a BRI line an will
switch to PRI in 6 month.
No, BRI and PRI are different things. The easiest and cheapest way
This is the URL I got the config file from, http://www.plugndial.com/ it's
on a link from the SipPhone URL.
I just modified the text for my phone.
There is a bit more info on there, and there is a MAC address on the top
line of the file.
Still just playing with this myself so don't know all the
Hi there,
The asterisk website mentions support for some models of
Intel Dialogic cards. I looked up in the Asterisk handbook, which has a
footnote saying
Dialogic hardware is not supported by standard Asterisk
but is available as a pay for add on for customers with Dialogic hardware.
I
Thank your for the link - now I wil try it :)
/Hans-Henrik Andresen
This is the URL I got the config file from, http://www.plugndial.com/ it's
on a link from the SipPhone URL.
_
Learn how to choose, serve, and enjoy wine at Wine @
Is there a search engine for this list?
- Kim Hendrikse
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hi List,
Take a look at http://www.voip-info.org/wiki-Codecs.
Which is de best codec to use with asterisk.
Let's say that we have a asterisk that works it SIP, H323, i4l, capi, etc ..
Which codec should i use if i want to make call between SIP phones ? And with
H323 phones (with a gatekeeper)
Hi there,
from my logfile (with Swissvoice ip10 phones):
WARNING[23575]: chan_mgcp.c:847 mgcp_indicate: Don't know how to indicate
condition 14
Anyone has a clue as to what this means?
Cheers, Philipp
___
Asterisk-Users mailing list
[EMAIL
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 of * with FWD.
Cheers, Philipp
___
Hi,
We have a system that recorded voicemail for about an hour after the caller
hungup. I'm going to put a timeout on it but is there anything to look for
that can help prevent this? The system is running on a telenet line in
Belgium. The answer dialplan I used was:
[macro-stddial]
exten =
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 of * with FWD.
Hi,
Ive got two H.323 Client connected to Asterisk, when one of them
requests boeing connected to the other I use CALL application and both get in
touch trhough asterisk, but using Call Asterisk stays on the middle and the
sound quality gets poor. Is there any way to transfer the call so
We have a system that recorded voicemail for about an hour after the caller
hungup. I'm going to put a timeout on it but is there anything to look for
that can help prevent this? The system is running on a telenet line in
Belgium. The answer dialplan I used was:
[macro-stddial]
exten =
Use something like the following in voicemail.conf
; How many seconds of silence before we end the recording
maxsilence=10
; Silence threshold (what we consider silence, the lower, the more sensitive)
silencethreshold=128
Rich
Ah, great. Thanks! Do you know how to find out what the
On Mon, Jan 19, 2004 at 10:51:02AM +0100, Nicolas Bougues wrote:
Attached is the config file I send to my Grandstream.
Change IP address Phone ID to suite.
That's great. Is it documented somewhere ?
And how do you manage tens or hundreds of phones ? Are they all in the
same
Use something like the following in voicemail.conf
; How many seconds of silence before we end the recording
maxsilence=10
; Silence threshold (what we consider silence, the lower, the more sensitive)
silencethreshold=128
Rich
Ah, great. Thanks! Do you know how to find out what
Hi Terence,
Terence Parker wrote:
Hi there,
After a lot of valuable insights from the list, incoming and
outgoing calls finally work through OpenLine4! Thanks for all the
input!
Now I have 2 minor issues:
Sometimes Voicetronix dials too quickly before an actual dial tone
is
How about a hashed directory structure? Something like this would be
easily human and machine readable. This can also be an opportunity to
lay the groundwork for internationalization.
Numbers and digits would have their own directories, as would the demo
phrases, agent and voicemail sounds.
Alexandru,
I think the subject line has a tendency to confuse the issue we're discussing
here. At least remove SS7 from it and call it, maybe, TDMoIP, TDMoPW (it's
actually a pseudo wire you're looking for, i think). You want to transport E1
over an IP cloud, right? You don't want the IP cloud to
Hi all,
I am planning to use VoIP gateways to connect remote offices to Asterisk.
Not having much experience with these and Asterisk I would appreciate any
info/experience that you might share with me as to their interoperability
with Asterisk.
I am interested with in rather bigger gateways
Hello,
I've had Asterisk installed on HT capable machines in both HT mode(with SMP)
and non HT mode (with non-SMP) and did not notice any differences
functionally between them. The processor load was always less in HT SMP mode
than non HT and I have experienced Asterisk deadlocks in both modes so
See http://www.rad.com/ , TDM-over-IP solutions.
- Original Message -
From: Alexandru Coseru [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 19, 2004 6:56 AM
Subject: Re: [Asterisk-Users] SS7 over Asterisk ?
Maybe , I never tried TDMoE ...
Where can I found a
From: Dustin Knuttgen on Sunday, January 18, 2004 11:47 PM
We tried to use SuSE initially and had no luck compiling zaptel on
either 8.2 or 9.0. We even had Digium take a look. After working on it
for days we finally switched to Red Hat 9.
Is there anyone who succeeded in compiling
Hi all,
I`ve
installed succesfully asterisk wiht h323 protocol, I need kwon how many concurrenst
call support asterisk working with h323 clients.
My
other questions is: I have a sound file in g.723.1 format in sound directory,
my h323 clients have the g.723.1 codec but when I make a
Yes. Check the mailing list archives.
On Sun, 2004-01-18 at 19:28, Robert Murray wrote:
Hi
Has anyone opened up a grandstream phone or handytone ATA to find out what is
inside? What is the CPU? How much RAM?
Cheers
Rob
___
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 work with CallManager or
now, I *think* they were
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 handling this is to have all of the sounds
in one
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 DSPs on it. This kind of card would provide
hardware assisted DSP functions
Cool - thanks Florian. I'll give that a try.
I guess there isn't a away to just pass the native flash via SIP yet?
-Original Message-
From: Florian Overkamp [mailto:[EMAIL PROTECTED]
Sent: Monday, January 19, 2004 2:30 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] ATA-186
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
hardware assisted DSP functions as well as patent indemnification.
Would you even have to USE the DPSs in order to be patent indemnified?
Using the DSP
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
-Original Message-From: Cesar Rico
[mailto:[EMAIL
Pretty much no. The ADSI specification was crippled from the start to
specificly not compete with PBX offerings. It has one advantage of
(very limited) programmability, but a phone like the SNOM has an
open-source core. It also has the dubious value of being interchangeable
with a regular
Stay away from Auidocodes... No support
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dawid Mielnik
Sent: Monday, January 19, 2004 5:21 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] pri gateways and asterisk
Hi all,
I am planning to use VoIP
Hi there,
I'm wondering if there is a way to assign a different Caller ID to each Zap
interface.
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.
Many thanks,
Steve
--
Steve Foy
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
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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 work
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
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 DSPs on
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, 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]
___
Asterisk-Users mailing list
[EMAIL
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 you
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
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 different
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
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,
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
- 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 the
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
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
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:
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 layout
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
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 FXOport per PCI slot(FYI I hear they
are working on a 4 port per PCI slot). The other options are MediaTrix,
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,
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
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
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 based fields
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
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-Users
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
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 retool
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
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
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
quote who=Andrew Thompson
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
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
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 11:17
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
___
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:
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 attempting
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.
___
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
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
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
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
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
-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
- 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 use voip
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
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
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.
smile
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
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