Is there a way to config a sip user so that he appears to be connecting from a
different IP address?
I want to use different IP addresses to authenticate different accounts with
service providers rather than the username/password combo.
Are there SIP settings to allow that?
/Obelix
Hi,
We have done it a little different...that said though, we have had
periods of weird things happening, like digits dropping, (we blame the
Nortel though!, so not the gospel ; ) :
zaptata.conf
[channels]
context=incoming
switchtype=euroisdn
pridialplan=local
signalling=pri_net
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, October 17, 2005 7:18 PM
Subject: Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk
On Mon, Oct 17, 2005 at 06:56:39PM +0100, Steve Daniels wrote:
Try a a good old
Here (Croatia) is also possible to get partial E1 (PRI/PRA)... from one of
our telcos (DT T-com) we can get PRA in 10 increments:
10B,
20B and
30B
We have a partial T1 (5B + D, iirc) from Allstream - there may be a
provider in your area that does something similar.
Regards,
--
Anthony Rodgers
I think I ran in to this problem a while back as well. I'm also running
a CVS version of Asterisk. I talked to David and he switched me to SIP
from their gateway to their Asterisk proxy which solved the issue.
On Mon, 2005-10-17 at 22:05 -0500, Rob Fugina wrote:
On 10/17/05, Rich Adamson [EMAIL
We recently migrated a couple of PRIs to Asterisk 1.0.9.2 and a TE406P
and are getting reports of talkoff (spurious/random DTMF tones heard by
people on SIP equipment connected to the Asterisk server. We previously
were using 1.0.3 with a T100P without any talkoff.
(a) We have not set
GoIAX, the Asterisk community's free IAX provider, is offering free US
dids now. I loaded about 175 dids in and put up a very beta sign in page.
Unfortunately I had to restrict the free us/canada outbound calling back
down to toll-free only. There was a lot of war dialing and prank
calling
On Mon, Oct 17, 2005 at 07:01:17PM -0400, Walt Reed wrote:
On Sun, Oct 16, 2005 at 09:21:09PM +0200, [Ludwig IT-Services - GMAIL ] -
Michael Ludwig said:
I'm very new to this list and to asterisk and stuff at all.
To build my asterisk server I installed a new machine running the new
SUSE
Hi,
I'm currently using chan_capi-cm-0.6, with the following capi.conf:
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de
[ISDN1]
msn=8304490
incomingmsn=8304490
isdnmode=msn
group=1
controller=1
softdtmf=1
context=demo
echosquelch=1
echocancel=yes
echotail=64
That really is a shame, goiax.com has been the best free termination service
I have seen. The call quality was excellent, better then some paid services
I have used.
One idea, I'm not sure if you already did it, only allow one concurrent call
per account?
And now DIDs, thanks from all of us for
Hi, When i place a call on hold, and
then return to it, the caller then hears my voice in a delay usually equal
to the amount of time i put them on hold. I have the problem only with G729
codec and with my voip provider (i live in france
and i use wengo)
My
configuration : - Pentium III
On Tue, 2005-10-18 at 02:36 -0500, Kevin Scott wrote:
That really is a shame, goiax.com has been the best free termination service
I have seen. The call quality was excellent, better then some paid services
I have used.
One idea, I'm not sure if you already did it, only allow one concurrent
On 18 Oct 2005, at 08:05, Matthew Simpson wrote:
GoIAX, the Asterisk community's free IAX provider, is offering free
US dids now. I loaded about 175 dids in and put up a very beta
sign in page.
Fantastic, got one, thanks.
Unfortunately I had to restrict the free us/canada outbound
Hi :)
I have a little 'slow dialling' problem. When I dial, e.g.
200# for the Asterisk 'echo test' demo application from my PBX extension
1010, I see this in the console the instant I press the # key:
-- Starting simple switch on 'Zap/65-1'
-- Accepting overlap call from '1010' to '200' on
Hello,
iam trying to connect an analogue Fax (as opposed to a ISDN Fax device)
behind a TDM400P. However, when i connect the Fax to the Card, asterisk
shows it as always being offhook. Iam currently out of ideas what might
be wrong. The Fax device is connected using a 1:1 four-wire RJ cable.
Dear Asterisk developers,
I run the same asterisk version on the home machine and on the work. On
the home machine I have Slackware 10.0 (kernel 2.4.24) while on the work
machine I have Mandrake 10.1 (kernel 2.6.8.1).
When I run asterisk on the work machine, these warnings and error appear
Stefan Günther schrieb:
With the above configuration the display always shows 8304490.
I've tried to change the number in the dialplan, but this doesn't change
anything:
exten = _90[23456789].,1,SetCIDNum(83044912)
Try to use SetCallerID instead of SetCIDNum and see if it helps.
exten =
Hi!
msn=8304490
incomingmsn=8304490
Each user has a different numer, e.g. 83044910, 83044911, 83044912 and
so on.
This number should appear on the display of the called party, but how do
I configure this?
With the above configuration the display always shows 8304490.
I've tried to
On Tue, 18 Oct 2005, Stefan Günther wrote:
..
Each user has a different numer, e.g. 83044910, 83044911, 83044912 and
so on.
This number should appear on the display of the called party, but how do
I configure this?
With the above configuration the display always shows 8304490.
I've tried to
Hello All,
I am trying to create a custom callback on asterisk.
[custom-callback]
exten = s,1,Wait(2)
exten = s,2,Hangup
exten = s,3,Dial(Zap/g0/92962676)
exten = s,4,DigitTimeout(5)
exten = s,5,ResponseTimeout(10)
exten = s,6,Authenticate(1234)
exten = s,7,DISA(no-password|from-internal)
Corrado wrote:
Dear Asterisk developers,
I run the same asterisk version on the home machine and on the work. On
the home machine I have Slackware 10.0 (kernel 2.4.24) while on the work
machine I have Mandrake 10.1 (kernel 2.6.8.1 http://2.6.8.1).
When I run asterisk on the work machine, these
I need some help generating configuration files for Asterisk. Since I'm
running under Solaris I'm having trouble with some of the utilities that
are more linux-centric.
Can anyone recommend a free/low-cost package to generate conf files that
is not linux-dependent and will handle a IAX2 and
Hi,
I'm running 1.2 beta1 in a mini call center.
I have 3 queues with 10 operators, and I'm running into some trouble because
when all the operators are busy answering call asterisk still sends them more,
resulting in a beep beep (call waiting) over and over again in Xlite audio.
An easy
I was wanting to use the new MuxMon application to record calls - it
seems to be a nicer way of recording than the Monitor application.
However, there is a slight issue with agents - we use the recordcalls
option in agents.conf to record inbound agent calls - and I believe from
looking at the
carl is the link.
http://www.tmcnet.com/usubmit/2005/Aug/1170660.htm___
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi,
This issue has been discussed probably a million times on every asterisk forum
in the world and I have the same problem too. Another problem you would have
with the agents is that when they make an outgoing call they are not regarded
as busy by asterisk and it sends more calls to the
If you're using AgentCallBackLogin it should be fairly easy to to do what you're looking for in step 'b'. On 10/18/05, Julian Lyndon-Smith
[EMAIL PROTECTED] wrote:I was wanting to use the new MuxMon application to record calls - it
seems to be a nicer way of recording than the Monitor
Hello,
iam trying to connect an analogue Fax (as opposed to a ISDN Fax device)
behind a TDM400P. However, when i connect the Fax to the Card, asterisk
shows it as always being offhook. Iam currently out of ideas what might
be wrong. The Fax device is connected using a 1:1 four-wire RJ
This behaviour is totally senseless since the whole purouse of queues is to
_queue_ the callers until the agent is available. available usually means
not on the phone -- whether or not it's an incoming or outgoing call.
Agree!
I solved this problem by using single-line clients and
My suggestion would be the one-line eyeBeam phone under development. Check out
support.xten.com.
//Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]
Skickat: den 18 oktober 2005 14:48
Till: Asterisk Users Mailing List -
I need some help generating configuration files for Asterisk. Since I'm
running under Solaris I'm having trouble with some of the utilities that
are more linux-centric.
Can anyone recommend a free/low-cost package to generate conf files that
is not linux-dependent and will handle a IAX2
On Wed, Oct 19, 2005 at 09:05:45AM -0400, asterisk wrote:
I need some help generating configuration files for Asterisk. Since I'm
running under Solaris I'm having trouble with some of the utilities that
are more linux-centric.
Can anyone recommend a free/low-cost package to generate
Hello,
I have to deploy an Asterisk PBX with this requirements:
- 1 or 2 ISDN lines in input/output
- 14 internal analog phones (yes, I know, analog ones... ;( )
- Billing interface for the operator (for usage of analog phones)
For the external interface I'm thinking about Beronet Quad Span
Hi,
I looked at the docs and probably missed it: is there a way to set a
dialplan on the GXP-2000? (to avoid having to press Send)
Thanks,
--
Computers are useless. They can only give answers. - Pablo Picasso
___
--Bandwidth and Colocation sponsored
On Wed, Oct 19, 2005 at 09:05:45AM -0400, asterisk wrote:
I need some help generating configuration files for Asterisk. Since
I'm
running under Solaris I'm having trouble with some of the utilities
that
are more linux-centric.
Can anyone recommend a free/low-cost package to
Hello,
I have to deploy an Asterisk PBX with this requirements:
- 1 or 2 ISDN lines in input/output
- 14 internal analog phones (yes, I know, analog ones... ;( )
- Billing interface for the operator (for usage of analog phones)
For the external interface I'm thinking about Beronet Quad
Hi
Is it possible with
Asterisk to tellthe called party which number was dialled by the
caller? Or in place of the number dialled have a description such as
'Sales' or 'Accounts'? Ideally, I would like to show a description
corresponding to the number dialled followed by CIDName.
How might
Jerry, here are the relevant parts of my 7206 config. Some things have been
changed to protect the innocent. ;)
dspint DSPfarm1/0
codec med
!
isdn switch-type primary-ni
!
!
voice call send-alert
!
voice service pots
fax protocol pass-through g711ulaw
!
voice service voip
fax protocol
Hi all,
I have a colleague who is very stuck on dialogic boards. I now the
asterisk web site says it supports some dialogic boards but has anyone
actually
installed one and gotten it to work. I tried once to install Dialogic SR
5.1.1 with a D/41JCT-LS but gave up and ended up reformatting and
Hi
I have a asterisk working in Costa Rica and everything is working well
except when an incoming call from the PSTN hangs up, asterisk wont hang up.
The port is busy
I probe the brazil configuration, but not work.
Any ideas?
,
Olger Merlos V.
I use a partial T1 as well (12B + 1D). Most CLECs offer them.
--
Tom
On 10/18/05, Goran Skular [EMAIL PROTECTED] wrote:
Here (Croatia) is also possible to get partial E1 (PRI/PRA)... from one of
our telcos (DT T-com) we can get PRA in 10 increments:
10B,
20B and
30B
We have a partial T1
Hello,
you should use asterisk agents and you'll see that the problem will go
away.
Bye
l.
On Tue, 18 Oct 2005 14:13:32 +0200, [EMAIL PROTECTED] wrote:
Hi,
I'm running 1.2 beta1 in a mini call center.
I have 3 queues with 10 operators, and I'm running into some trouble
because when
Not that I've seen.. about all you can do is adjust the inter digit
timeout..
Louis-David Mitterrand wrote:
Hi,
I looked at the docs and probably missed it: is there a way to set a
dialplan on the GXP-2000? (to avoid having to press Send)
Thanks,
begin:vcard
fn:Matt Hess
n:Hess;Matt
Hello,
Is there or can anyone provide a comprehensive guide (designed for Linux/Asterisk novices) to installing/setting upAsterisk in order to support Fax2Mail service?
In my case, I would like Asterisk to receive fax calls to predefined numbers (ranges) and to associate each of these numbers to
Is it possible to route a call from an Asterisk box through the
Internet to a IAX device (in this case Digium IAXy) without
using an IAX service like IAXTel? I have it working on my
local Ethernet LAN so it should be possible to use VPN to
cross the internet. Anyone using VPN or other method to
On Oct 19, 2005, at 9:26 AM, asterisk wrote:AMP's dialplan and setup is quite complex. Requires, e.g, a number ofAGIs.This is normally not the type of thing you'd like to hand-edit laterafter the initial adaptation to the target system.Who said anything about hand editing?That is why you would
Hi,
I have installed AAH beta 4 and I am getting this error. I have installed it
from aahbeta.tar.gz so I can make the server dual boot.
this is what I am getting in error, any clue how I can fix this?
Thanks
Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission
Chadwick E. Labno wrote:
Is it possible to route a call from an Asterisk box through the
Internet to a IAX device (in this case Digium IAXy) without
using an IAX service like IAXTel? I have it working on my
local Ethernet LAN so it should be possible to use VPN to
cross the internet. Anyone
I am attempting to unify how numbers come to me from a
specific T1, this T1 acts as an ingress for about 4000 DIDS. About 98% of
those DIDS come in as a 10-digit DNIS, what I would like to do is have asterisk
log when a number comes in 7 or 11 digit so I can contact my upstream provider
AMP's dialplan and setup is quite complex.
Requires, e.g, a number of
AGIs.
This is normally not the type of thing you'd like
to hand-edit later
after the initial adaptation to the target
system.
Who said anything
Hi,
I have installed AAH beta 4 and I am getting this error. I have installed it
from aahbeta.tar.gz so I can make the server dual boot.
this is what I am getting in error, any clue how I can fix this?
Thanks
Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream:
Permission
I've the following installation :
|asterisk client| --- |asterisk server| --- |other asterisk server|
all the connections are made in IAX, the client and first server allows
711 and 729
the other server only allows 729 since it has low bandwidth at disposal
all the numbers but a few are
Jeff Herring wrote:
Phone won't register on LAN port registers but doesn't work on PC port.
SIP to SIP works.
Anyone have a Configuration that works out there?
Phone has 4.63 Firmware
Make sure you have nat=never (or nat=route).
Regards,
--
Jason Becker
Director CEO
Coalescent Systems
Is it possible to route a call from an Asterisk box through the
Internet to a IAX device (in this case Digium IAXy) without
using an IAX service like IAXTel? I have it working on my
local Ethernet LAN so it should be possible to use VPN to
cross the internet. Anyone using VPN or other
Title: Newbie Question: Help with incoming dial plan
Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning.
Since all inbound calls come through my T1, I would like to setup a dial plan that
Title: Newbie Question: Help with incoming dial plan
This is how I do it.
[default-incoming]exten =
2691,1,Goto(extensions,3212,1)exten =
2692,1,Goto(extensions,3204,1)exten =
2693,1,Goto(extensions,3207,1)exten =
2694,1,Goto(extensions,3212,1)exten =
2695,1,Goto(extensions,3205,1)exten =
Title: Newbie Question: Help with incoming dial plan
I do not use any DID, all calls come in on the same number
111222 so what I would like to do is simply prompt the caller to enter the
extension they wish to reach, then redirect to that extension in the [default]
context.
David A.
Title: Newbie Question: Help with incoming dial plan
exten =
s,1,Answerexten = s,2,Wait,2exten =
s,3,Background(enter-ext-of-person)exten = s,4,DigitTimeout,5exten
= s,5,ResponseTimeout,10
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
MorrowSent: 18
Title: Newbie Question: Help with incoming dial plan
add this context
[default-incoming]exten =
111222,1,Goto(default-incoming,s,1)
exten = s,1,Answerexten =
s,2,DigitTimeout(10)exten = s,3,ResponseTimeout(20)exten =
s,4,Background(swelcome)exten = t,1,Hangupinclude =
extensions
add
On FC4 is better to use pwlib 1.9.1 and openh323 1.17.2.
I think, that OPENH3232DIR= is wrong. Better is OPENH323DIR= :-).
If You use standard prefix for instalation o packages there is a better way
instad copy library edit /etc/ld.co.conf and use /usr/local/lib/ as next
source of shared
I am doing some experimenting with Asterisk 1.0.9 and Polycom IP501's. I have
the extensions setup, and everything is working well up to this point. Now, I
want to setup my system so that a user at an extension can start a recording on
demand. I have tried various Google searches, but am
They are chown to asterisk:asterisk and chmod 777 . But I am still getting
those error.
Any other suggestion?
Thanks
Quoting asterisk [EMAIL PROTECTED]:
Hi,
I have installed AAH beta 4 and I am getting this error. I have installed it
from aahbeta.tar.gz so I can make the server dual
On Wednesday 19 October 2005 15:34, asterisk wrote:
Hello,
I have to deploy an Asterisk PBX with this requirements:
- 1 or 2 ISDN lines in input/output
- 14 internal analog phones (yes, I know, analog ones... ;( )
- Billing interface for the operator (for usage of analog phones)
For
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
I've never seen that, it's always when we call out. Certain numbers will
always trigger it. 888-737-4787 (IPC Resistors, it dumps into an IVR so it's
safe to call) is one such number, but we have local numbers that hit other
I just tried this
Hi,
I wish to use Asterisk as a SIP server.
How do I use Asterisk in a redundent network?
So, if one Asterisk server fails, how does failover work?
James
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Has anyone experienced problems with Vontage and Asterisk. I'm using
Asterisk (Current Stable) and Sipura-841 phones.While talking on an
outbound call the transmission seems to fade out until the other person can't
hear me but I can hear them.
I've updated the firmware on the 841
Note: forwarded message attached.---BeginMessage---
Someone will be in contact with you within the next couple of hours to discuss
your account.
Regards,
I am Jerry Richmond CEO of ByVolution LLC. We have purched some did's from
you that we use to test with, weare going to order our first
I will be on my cell 919 606 7685.
We need help bad.Jerry Richmond [EMAIL PROTECTED] wrote:
Note: forwarded message attached.Date: Tue, 18 Oct 2005 11:04:09 GMTTo: [EMAIL PROTECTED]From: "Sales Support" [EMAIL PROTECTED]Subject: {100-1287} RE: DID"sSomeone will be in contact with you within the
I have this in sip show history for a particular channel marked as dead
(should be removed) in sip show channels:
1. TxReqRelINVITE / 102 INVITE
2. Rx SIP/2.0 / 102 INVITE
3. CancelDestroy
4. Rx SIP/2.0 / 102 INVITE
5. CancelDestroy
6. Unhold SIP/2.0
Jerry if you have something to ask or say about your vendor on this list
do so.
But please stop dumping a copy here of all communications with them.
Jerry Richmond wrote:
Note: forwarded message attached.
big snip
___
--Bandwidth and
Title: Newbie Question: Help with incoming dial plan
Thanks Steve, this works like a charm!
Might I ask how I setup that Directory?
David A. Morrow
Technical Systems
Lead Autodata
Solutions Company [EMAIL PROTECTED] http://www.autodata.net
Tel: (519) 951-6079
Fax: (519) 451-6615
Poor
I am a newbie and want to step up to VoIP and switch from analog
connetion to my Astrisk/Lineox box.
Any suggestions on configuring Vontage and what to get/ask
when signing up?
Has anyone experienced problems with Vontage and Asterisk. I'm using
Asterisk (Current Stable) and Sipura-841
Hallo all,
i have a problem on loading chan_misdn. The misdn is running and all
cards (TDM40B+AVMFritz) is initialized. When im going to start asterisk
with the chan_misdn.so module i get the following error in the log (on
console) and asterisk ist hanging.
i use the current CVS-HEAD of
Hi James,
[EMAIL PROTECTED] wrote:
I am doing some experimenting with Asterisk 1.0.9 and Polycom IP501's. I have
the extensions setup, and everything is working well up to this point. Now, I
want to setup my system so that a user at an extension can start a recording on
demand. I have
Subject: RE: [Asterisk-Users] PRI echo issues: solvable?
Kris Boutilier [EMAIL PROTECTED] wrote:
On Tuesday 11 October 2005 11:49, alan wrote:
After solving the other low hanging fruit audio issues in our Asterisk
PBX, we are left with occasional cases of severe echo which we have not
Hi,
I wish to use Asterisk as a SIP server.
How do I use Asterisk in a redundent network?
So, if one Asterisk server fails, how does failover work?
James
James, I've been working on the same thing. I think it's pretty
important too because phone providers shoot for five-nine
Kevin Bockman wrote:
Patrick wrote:
is there anyway to make the dial command return and execute
the next line in the dial plan after the channel hangs up?
Try with h (for hangup):
exten = 1234,1,Dial...
exten = 1234,h,...
He actually meant the 'h' exten and not priority:
exten =
Hi James -
I am doing some experimenting with Asterisk 1.0.9 and Polycom
IP501's. I have the extensions setup, and everything is working
well up to this point. Now, I want to setup my system so that a
user at an extension can start a recording on demand. I have tried
various Google
This is with asterisk 1.20beta1:
I was experiencing moments of sporadic silence, so I thought to turn on
the jitter buffer in iax.conf.
I started with the following settings, which are basically ripped from
the sample config:
jitterbuffer=yes
forcejitterbuffer=no
maxjitterbuffer=1000
To avoid any confusion, you may note that the Dial Application does not
time out in this log excerpt as I described. That's because I hung up
the cell phone instead of waiting for the timeout.
And before anyone asks, setting jitterbuffer=off made the problem go away.
On Tuesday 18 October 2005 12:18, Doug Meredith wrote:
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
I've never seen that, it's always when we call out. Certain numbers will
always trigger it. 888-737-4787 (IPC Resistors, it dumps into an IVR so
it's safe to call) is one such number, but we
On Oct 18, 2005, at 2:03 AM, George Pajari wrote:
We recently migrated a couple of PRIs to Asterisk 1.0.9.2 and a TE406P
and are getting reports of talkoff (spurious/random DTMF tones heard
by people on SIP equipment connected to the Asterisk server. We
previously were using 1.0.3 with a
Title: Forwarding Extensions using dialplan
Hi all. So far this list is proving it's worth, even on my first day using it!
I hope that someone might know an easy solution to this one.
I would like to create a dial plan which will allow me to have all extensions 6XXX cause a dial-out of
Hello people,
i have a question concerning a quad-pri card (tor2 is the module for
this card)
i want the span to be completely shut down when alarms occur on it; i
want the span to be shut down immediately to avoid compromising the
whole box if one E1 line goes crazy and to be activated only by
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of alan
Sent: Tuesday, October 18, 2005 10:34 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: PRI echo issues: solvable?
Subject: RE: [Asterisk-Users] PRI echo issues: solvable?
I have been using goiax for my outgoing and has been excellent in quality
ofvoice and also the service.To put the abusers out, I propose that
each account holder shouldpre-register the phone numbers they typically
call. For a legitimate user,pre-registration is certainly acceptable
Hi, I´ve just installed an Asterisk Server on a Fedora Core 4, and made it
work between diferrent extensions in the office and now I need to make it
work on calling outside the office and I think I need a Dial Plan, can
somebody help me a little with this?
Thanks a lot
Hi,
well, some clients have strange ideas and wishes (at least to my mind).
Yesterday I gave a presentation about asterisk to a CEO.
At the end he asked me whether asterisk is able to do the following:
When a call for the CEO comes in, the calling number should be shown on the
display of his
On Tue, 2005-10-18 at 14:50 -0400, Dave Morrow wrote:
Hi all. So far this list is proving it's worth, even on my first day
using it!
I hope that someone might know an easy solution to this one.
I would like to create a dial plan which will allow me to have all
extensions 6XXX cause a
On Tue, Oct 18, 2005 at 09:10:38AM +0200, Tzafrir Cohen said:
On Mon, Oct 17, 2005 at 07:01:17PM -0400, Walt Reed wrote:
I was unable to get a clean compile of the kernel or * with gcc 4.
You can ask about this in Debian lists. I don't have unstable so I can't
test for myself, but the
I don't know of a comprehensive guide, but you can set it up using
NVFaxDetect, NVFaxEmail, and SpanDSP or Hylafax. NVFaxEmail can pull the
e-mail addresses from it's own config, voicemail.conf, a database, or thru
realtime.
Simple extensions.conf:
[incoming-dids]
exten =
Why don't you just try calling sellvoip directly? They are very responsive via
phone and email normally...
Their numbers are right on the website.
John
-Ursprüngliche Nachricht-
Von: Paul [mailto:[EMAIL PROTECTED]
Gesendet: Tuesday, October 18, 2005 9:55 AM
An: Asterisk Users Mailing
Message: 18Date: Tue, 18 Oct 2005 21:02:28 +0200From: Stefan-Michael. Guenther (in-put GbR)
[EMAIL PROTECTED]Subject: [Asterisk-Users] One phone ringing, one phone flashing ?To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]Content-Type: text/plain;charset=utf-8Hi,well, some
Hello All -
I've got an asterisksetup using 3 broadvoice lines and 5 Polycom IP300 phones. We have 1.5Mbit up and down via cable. 40ms (ave) pings to the broadvoice proxy and no packetloss.
The phones sound like cell phones. The person on the other end complains about it cutting in and out. On
Hello
I have two Asterisk's connected via IAX, they are sitting on the same
network, via a VPN, so there should be no problems with firewalls.
My problem is that when a person calls from A to B, A will not hear B
speak. B hears A fine.
I doesn't matter who initiates the call.
One of the
Felix Amaral wrote:
Hi, I´ve just installed an Asterisk Server on a Fedora Core 4, and made it
work between diferrent extensions in the office and now I need to make it
work on calling outside the office and I think I need a Dial Plan, can
somebody help me a little with this?
I have the
Why not just ask for a small one time payment $1 or something from a
credit card, or paypal, or something along those lines so you would have
someway to trace back to an abuser.
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Simpson
Sent:
The Asterisk I biult only does outbound calls, and it do them by LAN, I
don´t have any special hardware. Please help with the Dial Plan.
Thanks a lot
Felix Amaral
I.T. - Information Technology
Grupo PyD S.A.
Reconquista 1011 4º (C1003ABU)
Cap. Fed.- Argentina
TeL: +54-11--4800 Ext. 555
Has anyone got a
monit test written for IAX2?
I've tried:
check host blah with
address blah
if failed port 4569 use type udp then alert
But it seems to pass
even when I choose a fake port that I know is not open, like
4500
I'm wondering if
someone has used send|expect to do a basic
On Tue, 2005-10-18 at 13:52 -0500, Rajesh kumar wrote:
To put the abusers out, I propose that each account holder should
pre-register the phone numbers they typically call. For a legitimate
user,
pre-registration is certainly acceptable (especially for a free
service),
and if we give option
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