Ive read on the wiki that Xten and sometimes Firely tend to choke a bit when
in use, depending on the computer and such, is this true?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Tim Connolly
|Sent: Martes, 16 de Agosto de 2005 12:30 a.m.
|To:
That should be controllable by a weight, for example 2 peers:
A -- G729, G711
B -- G711, G729
What's currently happening is that * starts transcoding between the two
(g729 for A and G711 for B), what i would like is to apply a weight to
peer A so that the codec of choise at both sides becomes
Anton Krall wrote:
Ive read on the wiki that Xten and sometimes Firely tend to choke a bit when
in use, depending on the computer and such, is this true?
Correct.
I've had them on some computers where there has been an apparent memory leak.
However, EyeBeam (the new one from Xten) seems to
On Monday 15 August 2005 21:08, Innocent Evil wrote:
I am getting this whenever I start asterisk.
Aug 15 15:03:58 WARNING[11754] chan_oss.c: Read error on sound device:
Resource temporarily unavailable
sounds like your soundbard is blocked by another program. Sometimes
applications like KDE
Hi!
I tried install-misdn.tgz from http://www.beronet.com/download/ , some
minutes ago. Also I switched to an older kernel (2.6.8), but I get the
same error.
I think that I made the correct changes in the Makefiles, but I will
attach them to this e-mail, maybe you see something wrong.
Hello this is a newbie question,
is there any compatability issues with the HP Proliant ML370G3 Server if i
installed on it fedora core3 and asterisk 1.09 with E1 digium board and leatest
zaptel and libpri??
please note that the HW specs. is
Processor:Intel Xeon Processor 2.8 GHz/400 MHz -
Hi there:
I have installed the astcc successfully in my asterisk box.
When I dial the astcc calling card system number and access into the
system, but the system just returns a sound saying 6 and then hang
me up after around 30 seconds whatever I press on the keypad on sip
phone.
The number 6
Hello this is a newbie question,
is there any compatability issues with the HP Proliant ML370G3 Server if i
installed on it fedora core3 and asterisk 1.09 with E1 digium board and leatest
zaptel and libpri??
please note that the HW specs. is
Processor:Intel Xeon Processor 2.8 GHz/400 MHz -
Vahan Yerkanian said:
Try reinstalling sox - it is responsible for mixing the caller and
callee channels.
Nope it is not a sox issue. I listened to the ..in.wav and
..out.wav before they were soxed and the ..in.wav files are
distorted and running at a slower speed.
Anyone have an idea why
The way I said is the "gospel" of how it happens. /bOn Aug 16, 2005, at 1:42 AM, Erik Versaevel wrote:That should be controllable by a weight, for example 2 peers: ___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
hello
but this conf was working for me when i installed
asterisk last time. and UA was successfully reg and
working.
i think port forwarding is not the solution. because
it was working without port forwarding in my last
installation.
it is a simple case UA behind NAT and Asterisk is on
public
Joseph wrote:
I'll second that.
Hylafax has can handle the job. If you put asterisk in between you are
looking for problems.
I've the following setup working with asterisk NVBackgroundDetect
implemented.
PSTN -- asterisk -- hylafax
It woks, I would say 90% of the time. There seems to
You were right and I was wrong.
New sound card fixed all problems. Still can not beleive that problem was
caused by audio hardware, but there we are.
Thanks to all who replied.
Rudolf
- Original Message -
From: Rob Lith [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Hi,
I had configured Asterisk with the following
1). X100P - Card
2). Two -Greadstream100 SIP Phones.
I am able to make calls from SIP Ext to SIP Ext and PSTN calls from outside
to SIP Extn.
But I am not able to make calls from SIP Extn to PSTN out going calls-it
gives BT error message- The
Hi,
I had configured Asterisk with the following
1). X100P - Card
2). Two -Greadstream100 SIP Phones.
I am able to make calls from SIP Ext to SIP Ext and PSTN calls from outside
to SIP Extn.
But I am not able to make calls from SIP Extn to PSTN out going calls-it
gives BT error message- The
Dear Asterisk community,
sorry if I'm so stupid, but I couldn't register myself with Asterisk.
I created the [sip-incoming] context in the sip.conf:
[sip-incoming]
type = peer
username = elzhov
port = 5062 ; my kphone listens port 5062
host = 127.0.0.1
Then run Asterisk, and
hello,
You're not stupid but you have to create an account in
sip.conf for registration on ser
Look at sip.conf
Harry
--- Timur V. Elzhov [EMAIL PROTECTED] a écrit :
Dear Asterisk community,
sorry if I'm so stupid, but I couldn't register
myself with Asterisk.
I created the
Timur V. Elzhov wrote:
So I definitely misunderstand something in Asterisk SIP channel
engine :-/ Where I'm wrong?
You are wrong in not reading the available sample configurations and
configuration files. Read the sip.conf that is installed when you
install with make samples and check
post your dialplan, it's pretty safe to say that's where the problem is.
without it, there's no way to help you.
-yair
On 8/16/05, Appan KH [EMAIL PROTECTED] wrote:
Hi,
I had configured Asterisk with the following
1). X100P - Card
2). Two -Greadstream100 SIP Phones.
I am able to make calls
Does anyone know if the te110p would have any problems running on one
of these chipsets?
Need new server quickly and the acer altos g310 boxes look relatively
good...
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
The Dial plan is given below
[incoming]
exten = 197,1,Dial(SIP/197,20,tr)
exten = 197,2,Hangup
exten = 198,1,Dial(SIP/198,20,tr)
exten = 198,2,Hangup
exten=_0.,1,Dial(Zap/1/SIP/197,20,tT)
exten=_0.,1,Dial(Zap/1/SIP/198,20,tT)
appan kh
- Original Message -
From: Yair Hakak [EMAIL
The Dial plan is given below
[incoming]
exten = 197,1,Dial(SIP/197,20,tr)
exten = 197,2,Hangup
exten = 198,1,Dial(SIP/198,20,tr)
exten = 198,2,Hangup
exten=_0.,1,Dial(Zap/1/SIP/197,20,tT)
exten=_0.,1,Dial(Zap/1/SIP/198,20,tT)
Those last two statements are incorrect. You want
I am confused. what do you expect to happen when you call the PSTN?
let's say you call 023459823 (assuming you are in a country where
dialing codes begin with 0)
first of all, why do you have 2 lines that match the same extension
and tell asterisk to do different things? I am referring to these
This rocks! Use xten or diax.
Isamar
On Tue, 16 Aug 2005, Anton Krall wrote:
Anybody using Plantronics USB headsets? What softphone are you using and
whats your overall experience? Any comments/suggestions?
___
Asterisk-Users mailing list
Why don't you try Web Meet Me from the same author:
http://areski.net/Web-MeetMe/about.php
It's so much easyer to install.
regards
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Araba, Michael
Sent: lunedì 15 agosto 2005 7.35
To:
I use a DSP 500 and I love it. Great sound, good price.
IaxComm is hands down the best softphone I have found.
As you can guess it is for IAX though...
Cheers,
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Monday, August 15,
I think the easiest way to tell if you don't get an answer is to see if
it uses IRQ sharing and if it allows you to assign IRQs individually.
A check of the BIOS instructions for that Mobo should be available at
the manufacturer.
W
-Original Message-
From: [EMAIL PROTECTED]
Hello there,
Does anybody has already get asterisk to work with R2 E1s ? If so, what
version combination have you used, between asterisk, libmfcr2 and
unicall ?
I've already compiled asterisk 1.0.9 patched against some unicall
versions (0.0.3pre4 and 0.0.2[a,b,c]), after
It is an HP all in One t45.
It is plugged into a dedicated port on a sipura 2002.
The person is able to send faxes fine.. but when trying to receive it
just never gets anywhere.
We tried turning off the ECM on the fax machine.
There did not seem to be a place to lower the modem speed. It is
Topology:
PSTN-T1 PRI-NEAX2400-T1 PRI-Cisco 3825-Ethernet- Asterisk VoIP server
When I make a call to a VoIP user from the PSTN, the call gets routed
through the PBX, and Cisco. Because of that the DTMF tones are passed
inband, which I can hear on the VoIP end of the call. However, I have
one
Just make sure that your E1 card has the latest Digium firmware. Older
cards are know not to work in newer HP machines.
Mark
bodra wrote:
Hello this is a newbie question,
is there any compatability issues with the HP Proliant ML370G3 Server if i
installed on it fedora core3 and asterisk
Peter Svensson wrote:
On Fri, 12 Aug 2005, Bruce Ferrell wrote:
Hardware, possible. Unlikely to be cabling. It's usually a timing setting.
The blue alarm is really a very specific alarm condition normally. It
cannot quite see how it can be generated accidentally. Something along the
just a suggestion, but why don't you try using RFC2833 dtmf relay
between the cisco and the asterisk box.
use dtmfmode=rfc2833 in sip.conf, and you can also set the dtmf mode
per peer in sip.conf
also, if you use inband dtmf, this would only work with u-law and
a-law, and not g729.
on the cisco,
This is not an answer but rather an addition to the question. We're using a
large scale VOIP only asterisk system that has PAP2 enduser units using
inband as their DTMF mode. sip.conf is set for using inband as well, and we
pass PSTN calls through a provider.
Here's the problem, when our users
I run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF.
works very well and have never had a problem with it.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From:
I'll pass that on to my lead engineer, he was under the assumption that
rfc2833 was too unreliable. I personally don't know, but will look further
into the matter.
Thanks for the help
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Brian C. Fertig
Does HP Proliant ML370G4 Server is ok ?
--- Mark Phillips [EMAIL PROTECTED] a écrit :
Just make sure that your E1 card has the latest
Digium firmware. Older
cards are know not to work in newer HP machines.
Why new firmware make E card work with HP machines ?
Harry
Mark
bodra wrote:
Debug below
[voipbuster]
type=peer
host=iax.voipbuster.com
;host=213.61.187.150
secret=x
notransfer=yes
context=default
qualify=yes
disallow=all
allow=ulaw
allow=alaw
---
subspace*CLI iax2 debug
IAX2 Debugging Enabled
-- Executing SetCallerID(SIP/100-b0b3, xx) in new stack
--
Sherwood McGowan wrote:
This is not an answer but rather an addition to the question. We're using a
large scale VOIP only asterisk system that has PAP2 enduser units using
inband as their DTMF mode. sip.conf is set for using inband as well, and we
pass PSTN calls through a provider.
Here's
Sherwood McGowan wrote:
I'll pass that on to my lead engineer, he was under the assumption that
rfc2833 was too unreliable. I personally don't know, but will look further
into the matter.
You need a new engineer. OOB DTMF like RFC2833 is more reliable than
inband. With inband even a tiny
Yes Why new firmware make E1 cards work with HP
machines ?
http://h18004.www1.hp.com/products/servers/proliantml370/index.html
Harry
--- harry gaillac [EMAIL PROTECTED] a écrit :
Does HP Proliant ML370G4 Server is ok ?
--- Mark Phillips [EMAIL PROTECTED] a écrit :
Just make sure that your
Very interesting. I've already challenged him on it, since I found several
things online (including the Hitchikers Guide to Asterisk) that said
rfc2833 was better and that inband was generally not suggested.
After reading that, I would have gone with rfc2833...I'm not sure why he
didn't
Hi,
I saw your posting on Hipath and Asterisk.I have some doubts on the same.it would be really nice of you if you can help me out.My Doubt is as follows
Currently I am using Hipath HG1500 V3.0 with Opticlient4.0. But i am not satisfied with the performance of Opticlient. I wanted to use
Hi !
Did anyone had issues/managed to solve issues with DISA over Zap channels on
* 1.0.X (STABLE) ?
I have a situatuion where DTMFs that should be recognized in DISA work over
SIP channels and do not work over ZAP channels (Zap channels are on TE110P)
I have in default context:
exten=
Hi,
I search how to send a 12khz or 16khz to a payphone throught and FXS port.
It seem that asterisk sample is 8khz but in the documentation of si3215
(the slic of FXS module) samples rates is 16khz.
Anyone can help me ?
Thanks
--
GSM : 00212 60 54 65 68
WEB : http://www.jeremy-salmon.org
This is my features.conf
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
parkingtime = 45 ; Number of seconds a call can be parked for
;
sip show register will display the sip registrations the server has
performed to other peers, not other peers to it
this is also true for iax
not sure why you split the registrations into 2 instead of using
friend, friend works fine for me and I have not heard of any issues
of using it
This problem was solved by changing the preferred codec from
G729A to ulaw.
Eric Smith said:
We are using the following to record conversations.
exten = _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP})
exten = _1XXX.,2,Monitor(wav,${CALLFILENAME},m)
exten =
Jeremy Salmon wrote:
Hi,
I search how to send a 12khz or 16khz to a payphone throught and FXS port.
Not going to happen.
It seem that asterisk sample is 8khz but in the documentation of si3215
(the slic of FXS module) samples rates is 16khz.
Except that currently Asterisk deals with every
On Tuesday 16 August 2005 09:12, Jeremy Salmon wrote:
I search how to send a 12khz or 16khz to a payphone throught and FXS port.
It seem that asterisk sample is 8khz but in the documentation of si3215
(the slic of FXS module) samples rates is 16khz.
Anyone can help me ?
Basic sampling theory
Thanks for your help,
I have already seen this page but since the head version of ztmonitor is
able to show the real number value of the rx and tx (ztmonito -vv), I
was thinking that maybe someone could confirm to which value we want the
rx of ztmonitor when we try to calibrate the system with
Hi all!
I have 2 queues and 6 agents.
I don't like use the 6 agents in two queues at the same time.
I like use the following way:
The user select what queue s/he goes to participate.
Anybody can help me ?
Fernando Patzlaff
[EMAIL PROTECTED]
Hello All!
Just a notice that our PHP/Smarty-based GPL version of PhoneCALL version
2.6.1 has been released, and is the current stable release.
http://www.vecsector.com/phonecall
We're always looking for feedback/testers to help us enhance it and make
it even easier for everyone to use. The
Thanks I give give that a try. One follow up question. If the call
is coming in via the PSTN, and going through the NEAX (PBX) then to
the Cisco, can I control the way the PBX sends the DTMF, or is the
cisco some how able to split out the DTMF tones from everything else?
I was assuming that
Hi,
you must read
http://zarzamora.com.mx/asterisk/17
Regards
El mar, 16-08-2005 a las 09:09 -0300, Arnaldo M. Pereira escribió:
y get asterisk to work with R2 E1s ? If so, what
version combination have you used, between asterisk, libmfcr2 and
unicall ?
I have Asterisk working with one FXO card (clone x100P card PCI card).
I am trying to add 2 more cards so my question is - do I just
increase the channel count on the zaptel.conf and zapata.conf files?
[original]
/etc/zaptel.conf
fxsks=1
loadzone = us
defaultzone=us
/etc/asterisk/zapata.conf
The value of 14800 is correct.
I had issues with my TDM400p with 2x FXO's installed and using the Xlite
client. I could not get the echo stable at the initial call.
Changing to a hard phone made everything work correctly. I still had problems
with the off location I called, but mostly worked
Nenad Radosavljevic wrote:
Hi !
Did anyone had issues/managed to solve issues with DISA over Zap
channels on * 1.0.X (STABLE) ?
I have a situatuion where DTMFs that should be recognized in DISA work
over SIP channels and do not work over ZAP channels (Zap channels are
on TE110P)
I have already seen this page but since the head version of ztmonitor is
able to show the real number value of the rx and tx (ztmonito -vv), I
was thinking that maybe someone could confirm to which value we want the
rx of ztmonitor when we try to calibrate the system with a test line
from
Does anyone know of any USB ISDN adapters that work with Asterisk. My
gateway box is an old Compaq laptop (PIII 800, recently upgraded from a
Toshiba P120) and their are obviously no PCI slots. PCMCIA/Cardbus or
SIP gateway products are also an option.
Thanks,
Julien
signature.asc
Description:
On Tue, Aug 16, 2005 at 09:04:34AM -0500, Ric Moseley wrote:
I have Asterisk working with one FXO card (clone x100P card PCI card).
I am trying to add 2 more cards so my question is - do I just
increase the channel count on the zaptel.conf and zapata.conf files?
Basically, yes. See
Christian Wengel schrieb:
Hi!
I tried install-misdn.tgz from http://www.beronet.com/download/ , some
minutes ago. Also I switched to an older kernel (2.6.8), but I get the
same error.
I think that I made the correct changes in the Makefiles, but I will
attach them to this e-mail, maybe you
On Aug 15, 2005, at 11:28 AM, Bjørn Ove Kristiansen wrote:
Hello!
The issue is simply that I don't know which IP address the phone
tries to
connect to. I am not very familiar with dhcpd (never put it up by
hand), so
I'm not sure how the below would help me, but from what I can tell,
I
Fernando Patzlaff wrote:
Hi all!
I have 2 queues and 6 agents.
I don't like use the 6 agents in two queues at the same time.
I like use the following way:
The user select what queue s/he goes to participate.
Anybody can help me ?
Fernando Patzlaff
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Geia sou Irakli,
I would have to agree with Bryce that from the debug output the problem
seems to be with the dialed number.
Unkown Number Type Unkown Number plan point to that.
You should probably check out if you can start extensions with 3 ...
You can assume that Bearer capability means the
Not just HP machines. There's some issue with the chipset on the HP
machines (and some others using the same chipset).
If Zaptel/Asterisk cannot see your card then you'll have to send it back
to Digium for upgrade/replacement.
I do not know if this is the case for Sangoma or other makes of
This must be a question asked before but can't find it so here I go:
I have a Asterisk box connected, thou a x100p, to a PSTN PBX. When we
get a incomming call on that PBX the phones in the office wil ring and
there will also be a ring signal on the x100p. At my current
configuration the call
I like it!
Not quite as simple as AMP but it does seem to be more powerfull.
Keep up the good work and write a manual!
Mark
Dustin Wildes wrote:
Hello All!
Just a notice that our PHP/Smarty-based GPL version of PhoneCALL version
2.6.1 has been released, and is the current stable release.
Great, that worked. What about the /etc/asterisk/zapata.conf file?
Do I gust increase the channel count?
[channels]
language=en
;
; X100P #1,#2,#3 plugged into PSTN
;AMPLABEL:Channel %c - Button %n
context=from-pstn
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
You would put 30 in the includedseconds field and then 6 in the inc
increments field.
Darren Wiebe
[EMAIL PROTECTED]
maka wrote:
hi all,
Can I make the astcc script to rate the calls in a 30 seconds initial
period, and then in periods of 6 seconds? I noticed the inc field in
the brands
Check your extensions.conf on the context setted on zapata.conf
probably you have the command answer you should remove it.
On 8/16/05, Hubert Hoefsloot [EMAIL PROTECTED] wrote:
This must be a question asked before but can't find it so here I go:
I have a Asterisk box connected, thou a x100p,
Dustin,
It is pretty amazing, that you PhoneCALL has so many features
incorporated into a GUI of the tool, that needs little manual
modifications to the Asterisk config files.
I am sure that this will make all those closed source Commercial GUIs
redundant in near future.
Kudos and keep up the
If you are using a TDM card its also important to use the new tool fxotune.
This should help as it will match the fxo card to the line. Hybrid balance
will help echo as well and I assume fxotune is helping to balance the line.
With a matched hybrids on both ends of a 2 wire interface you will
Thanks Mark!
You're right - this version is intended for the 'advanced' admin, one
who is very knowledgable with Asterisk, but we are working on
simplifying the interface in the next revisions that will make
administration easier for most user types.
Basically - think of it like this:
The
Remarque : message transféré en pièce jointe.
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur
Also, what does the RED mean in this?
[EMAIL PROTECTED]:~]#more /proc/zaptel/*
::
/proc/zaptel/1
::
Span 1: WCFXO/0 Generic Clone Board 1
1 WCFXO/0/0 FXSKS (In use)
::
/proc/zaptel/2
::
Span 2: WCFXO/1 Generic Clone Board 2 RED
I am trying to connect two Xten eyeBeam Video Phone
No problems in voice connecting.
I tryed to modify my sip.conf
[general]
language=it
videosupport=yes
; enable Asterisk video support
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all
Great job! PhoneCall is very similar to the interface that I'm writing now,
with many of the same features! I love it!
Sherwood McGowan
--Original Message-
-Dustin Wildes wrote:
- Hello All!
-
- Just a notice that our PHP/Smarty-based GPL version of PhoneCALL
- version
- 2.6.1 has
Hi,
I installed this program but I am not able to configure, it does not
want to work.
Someone can help me?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
Hello,
How do you guys implement LCR in Asterisk?
Thanks,___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
On Fri, Aug 12, 2005 at 09:03:42PM -0500, Tim Connolly wrote:
You might start by running /usr/src/zaptel/zttest. See if you stay at 100%.
That's going to be the first thing digium checks. You might also run the
autosupport script and take a look at it for anything obvious.
I'm having lots of
Many of your questions have most likely already been answered either
on this list or on the wiki http://www.voip-info.org. Might want to
check there if you're just looking for a basic overview of how things
work and the various config files.
On 8/16/05, Il Neofita [EMAIL PROTECTED] wrote:
Hi,
Also check out this getting started page
http://www.oneunified.net/support/asterisk/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris
Edwards
Sent: Tuesday, August 16, 2005 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello,
How do you guys implement LCR in Asterisk?
I have experimented with 2 ways, both seem to have issues and further
testing is taking place now.
Method1, use realtime for extensions and load your routing tables in an
outbound context. Our requirements are LCR for the ~150,000 USA
Hi,
I checked but I did not find any info regarding the config files. I
tough that I configured everything in the right way but I am not able
to see anything on the web page.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
It is amazing to me at this point that there is not an official
Digium list of supported servers (including 1u models!). Clearly the number 1
issue with the Digium PRI cards is the server that they are used in.
The new cards even go as far as listing server that DO NOT
work on the
Is there a method in SIP to set the CALLING number type to
national and the calling number plan to isdn? I am dealing with an issue where
a media gateway is not sending the correct values and would like to know if SIP
has an equivalent parameter that can be set and mapped in the media
Damon Estep wrote:
What 1u server combos work with the new quad pri cards UNDER LOAD (more
than 75% channel use). Every user that buys a Digium PRI card should not
have to play hit or miss with 2 or 3 servers that cost more than the
card to get it to work.
We use a Sangoma 4 port T1 card in
I have some IP 500s that I bought used, but the connectors are different
than the new ones. There is a Modem/Power RJ11, a Line RJ11, and Handset
and headset connector. Does anyone know how they work?
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax:
My company is thinking about putting most of the logic for determining
which route is the cheapest in an AGI script. Some dialplan logic
wrapped around it to attempt to dial the routes in the order returned.
It hasn't been fully implemented yet.
The idea is to move the routing decision
If you are interested in helping test a solution like this, please let
me know. I'm currently working on implementing something along these lins.
Darren Wiebe
[EMAIL PROTECTED]
Johann wrote:
My company is thinking about putting most of the logic for determining
which route is the cheapest
It sounds like you are missing the sound files. There are a bunch of
.gsm files in the astcc/sounds source directory. They should be copied
to /var/lib/asterisk/sounds directory.
Darren Wiebe
[EMAIL PROTECTED]
wei li wrote:
Hi there:
I have installed the astcc successfully in my asterisk
Don Fanning wrote:
CALLED NUMBER : 1516308
Is that a valid number? AFAIK all voipbuster numbers have to start with
0 as there's no local dialing.
Tony
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
It's working for me, this is my config:
iax.conf
[from-pstn]
username=my-username
type=peer
secret=My-Password
qualify=yes
notransfer=yes
host=iax.voipbuster.com
disallow=all
allow=gsm
[voipbuster]
username=my-username
type=peer
secret=my-Password
notransfer=yes
insecure=very
I use this box with no problems at all.
http://www.tyan.com/products/html/gx28b2881.html
-bill
On 16-Aug-05, at 12:32 PM, Damon Estep wrote:
It is amazing to me at this point that there is not an official
Digium list of supported servers (including 1u models!). Clearly
the number 1 issue
On Tue, Aug 16, 2005 at 10:22:01AM -0600, Damon Estep wrote:
Any input from others that have already done what I am doing would be
helpful, what works best?
For 100k routes+, you will have trouble holding them in a SQL database,
particularly if your route selection query is complex. With a
And no RJ45 connectors? Doesn't sound like an IP phone at all.
Sure you did not get a phone for a Polycom PBX solution of some sort?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent: Tuesday, August 16, 2005 9:38 AM
To:
On 22:31, Mon 15 Aug 05, Chris Coulthurst wrote:
I set up a context to allow me to call in to my * server (via Teliax in this
case using IAX2) from my cellphone, and let me do a number of things,
including dial other extensions, AND dial outbound again so callers could see
my proper work
How does one block registration of sip phones using asterisk if that
sip phone is on a subnet other than the one allowed?
Thanks,
Brent
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
My personal fav.
http://www.provantage.com/buy-7ASUN00P-asus-networking-1u-p4-bbns-e7210-52x-cd-2x64-2sata-h-2geth-300w-ap140r-e1-aa2-shopping.htm
On 8/16/05, William Lloyd [EMAIL PROTECTED] wrote:
I use this box with no problems at all.
http://www.tyan.com/products/html/gx28b2881.html
1 - 100 of 197 matches
Mail list logo
