On 23:48, Thu 11 Jan 07, humphrey nyapokoto wrote:
i am having problems when i try to make asterisk run
automatically on boot up. I am getting the mesg
asterisk died with code 127
Does asterisk start when the system is booted ?
If so, you have the order of your rc scripts wrong.
Most common
Hi again,
the problem is not about agents or queues, is locate in the fucked
Cisco7912. The issue is the # character, seems like it don't send to
asterisk correctly, because I tried with the same config with a ekiga
softphone and there is no problem.
Any ideas to solve it?
thanks
Marc
Hi All,
I seem to be having a problem with all my VSPs. When I am registering
with them I don't seem to be passing my port number. This problem
causes other users the inability to call my VoIP number with the VSP.
My VSP showed me what they are seeing.
I have changed my useragent to
That would depend on many things. One of which is: How do you set the
DND? In Asterisk? In the Phone? Via a web interface?
Thanks Eric, I'm using the asterisk DND
Cheers,
Pierre
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Kevin,
Any chance you could give us a basic example of what you need in
sla.conf and extensions.conf to set up a Shared Line Appearance?
By Mapping actual trunk lines, does this mean you can essentially have
a button on phones that (for example) connect / maps you directly to
Zap/1 - i.e the
On 1/11/07, Mike D'Ambrogia [EMAIL PROTECTED] wrote:
Ralph
Kind of new to asterisk, and really new to AGI but it looks like you were
trying to have the AGI script tell asterisk to read and lay the results into
my_var and then regain control in the AGI script, is that correct?
If so I don't
On Thu, 11 Jan 2007, Ken Williams wrote:
I've never understood why people would think it's a PSTN issue. I'm
sure 99.99% of Asterisk users are using Asterisk on lines they've
had regular phone lines hooked up to before moving to Asterisk. In my
case I've also tried having the server
On 1/11/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Steve Edwards wrote:
On Thu, 11 Jan 2007, Yuan LIU wrote:
AGI doesn't see the name var; all it sees is an array @ARGV (or
whatever in the respective language). As the documentation says,
values are passed like command line
On 1/11/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
What version of Asterisk? Perhaps it changed since the last time I had
to deal with the issue. Perhaps it was fixed for 1.0, or maybe it was
specific to asterisk-perl.
Mike D'Ambrogia wrote:
Not true for the php version, it will
Bruce Reeves wrote:
I am perplexed by this so I how someone can help me out.
On one of my servers the users began complaining that if they picked
up a parked call they could not use the # key to transfer the call.
This is a particualarly annoying issue since everyone has been taught
to use
(applogies for the layout, having to use outlook)
If I could crack the Echo problem then the VOIP PBX would pass the rigorous
wife test (WT). I've used the TDM400P card and reduced the echo on one of
my 2 BT lines (the other one seems much better, but not perfect) to a point
where I can live
On 1/11/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
I understand that I have to compile zaptel but what about asterisk? Is it
enough to extract it? Well, how I will run asterisk (without compilation and
installation)?
Any advise?
Regards
Bilal
Bilal,
which distro you use?
Using
Hi,
Ron McCarthy wrote:
Hi List,
Has anyone got the record button to work on the Snom's? Im looking to
have it send a email with a attachemnt of what the user records I
hope. It looks like you just point the button to [EMAIL PROTECTED] and just
have that extension record it. Any clue on how
Thanks, I'll give it a try.
Allan.
- Original Message
From: M.Hockings [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, January 11, 2007 3:45:44 PM
Subject: [asterisk-users] Re: calls to SPA942 disconnect after 15 seconds
(chan_sip.c set_destination: can't find
Hello,
Thank you Leo for your answer,
I manage to do what I want perfectly when both the caller and the callee are
set in SIP with canreinvite=no using SIP INFO method for DTMF.
Now, I can't figure out why this can't work when I set canreinvite = yes
with the same DTMF method. Running
Hi to nobody :)
Finally I solve my stupid problem, I must changed my original sip.conf
for cisco phones to
dtmfmode=auto
Now the pound key # works well, and I can use queues with dynamic agents.
That's all
Marc Patino Gómez wrote:
Hi again,
the problem is not about agents or queues, is
All the variables here was my_var, it worked for GET VARIABLE but didn't for
SAYDIGITS and odbc connection. How can I SAYDIGITS of my_var or insert
my_var value into a db?
- What I need more to use WAIT FOR DIGIT? Because it didn't stop to wait for
digits.
- STDIN shoudn't get the result of READ
I cannot seem to find any reference to labels in realtime extensions -
using 1.4.
I've googled until my eyes have bled, and also scoured voip-info.org.
Is there anything that helps me here ?
Many thanks.
Julian
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bilal ghayyad wrote:
Hi List;
To create the symbolic link, I read in the documenation that I have to type
this command:
# ln -s /usr/src/'uname -r' /usr/src/linux-2.4
1) What it means by 'uname -r'?
2) Why I have to create such symbolic link to do pointing for the kernel? For
what exctly
Hi. I've been researching very deep into SLA in Asterisk 1.4, and am unclear
as to if the feature exist. I know the commands and configs are available,
but there is no documentation, and I've heard that it will not be supported
until 1.4.1. Does anyone have a definitive answer on this?
If it is
From: http://www.oldskoolphreak.com/tfiles/voip/tts-imap.agi
#!/usr/bin/perl
#
# AGI Script that reads back e-mail from an IMAP account.
#
# Requires the Asterisk::AGI, Net::IMAP::Simple, and Email::Simple modules.
#
# Written by: Black Rathchet ([EMAIL PROTECTED])
#
#
Ken Williams wrote:
I've never understood why people would think it's a PSTN issue.
Echo *IS* a PSTN issue. Telcos have been dealing with this for years.
Telcos do have high latency paths (granted less so these days).
Cell companies use hardware echo cancelers to remove the echo. Cell
Hello,
Before throwing in the towel with my Sipura 3000 has anyone had much
success with that adapter connected to a door phone?
In our setup a doorphone is connected to the SPA's fxs port. When a
visitor rings, asterisk calls a group of Polycoms and the person who
answers has to enter *1 to
Marc Archer wrote:
Any chance you could give us a basic example of what you need in
sla.conf and extensions.conf to set up a Shared Line Appearance?
I won't waste your time, because the current SLA implementation is
broken. We expect to have replaced it when Asterisk 1.4.1 is released,
and
Hey Guys,
I apologize for my ignorance on this one.
I've got several 7960s running on Asterisk1.4 with 15 or separate queues and am
trying to figure out a way to identify to the 7960s, what queue the incoming
call is on? Is this possible at all?
Thanks!
--
Robert Norton
SophMedia LLC
Is there a way to detect which end of a call hung up? If so can I log
this to the CDR records? Any pointers or can anyone point me to where I
can get this info?
_
Kevin Savoy
Business Unit Telecom Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
Fax:
Savoy, Kevin - Williston, ND wrote:
Is there a way to detect which end of a call hung up? If so can I log
this to the CDR records? Any pointers or can anyone point me to where I
can get this info?
[northpark-trunks]
;
; 9-1-nxx-nxx-
;
exten = _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1},,g)
On 1/11/07, Steve Langstaff [EMAIL PROTECTED] wrote:
So, to be clear, I have to do this in the application using the
management interface (which I don't happen to control) rather than in
the Asterisk dialplan (which I do)?
Yes, because the channel you originate does not exists in the PBX yet,
Hi,
This does nothing for me at all! :( I dont have to map the record key to
dial *1 or anything? Also, have you figured a way just to make the record
button work when not on phone, like a memo button prehaps? Maybe set one
speed key as a memo type button to send them a voicemail, and use record
I think what you want is called a directory, no? I'm not positive because
the English language isn't my main expertise, I know more about Linux and
stuff like that. Maybe you can find a newsgroup about English and get an
answer to that -- or better yet tell them to write all your other mailing
Yes, search.
www.voip-info.org
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Perhaps you need to re-think the problem I highly doubt the problem is
multiple VSP faliure, so your question should read something like
I seem to be having a problem with my asterisk configuration
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The problem was actually introduced in issue 8406 in mantis, and in the SVN
release 48154. Kind of makes me feel better knowing it is not my dial plan.
I agree with the fix, but I need a way around it, since it does not apply to
my setups.
On 1/12/07, Doug Lytle [EMAIL PROTECTED] wrote:
Bruce
I assume there is one NAT router for the LAN and nothing fancy, so setup the
Asterisk machine on the router/firewall (or make it such) and have it listen
on both LAN and WAN interface.
Now use a hostname for the SIP server, and run a DHCP/DNS server that will
resolve that hostname to the LAN IP
How about:
Action: originate
Channel: Local/[EMAIL PROTECTED]
etc
Then in extensions.conf
[indirect]
exten = _X.,1,NoOp(Click to Call)
exten = _X.,n,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten = _X.,n,Goto(from-internal,${EXTEN},1)
Get the idea? Does that help?
Cheers,
Steve
On 1/12/07,
The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling.
Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be
the line1 tab on spa3000. This applies to the fxo (pstn) also if you are
using it for such things as ivr's.
Doug
On Fri, 12 Jan 2007, Louis-David
I think that the support for the SNOM button (which uses a SIP message
to request the recording) is part of the bristuff patch, otherwise
only *1 will work.
http://www.junghanns.net/downloads/
Cheers,
Steve
On 1/12/07, Ron McCarthy [EMAIL PROTECTED] wrote:
Hi,
This does nothing for me at
I think you can encrypt it by the macaddress, check out http://spc.pifiu.com
On 1/9/07, Benko [EMAIL PROTECTED] wrote:
On Mon, 8 Jan 2007 20:03:50 -0500
Andrew Joakimsen [EMAIL PROTECTED] wrote:
Good luck dealing with Linksys on that
On Jan 11, 2007, at 8:53 PM, Ken Williams wrote:
I tried to be thorough, but of course left fxotune out. I did try
fxotune, it resulted in something like 9,0,0,0,0,0,0,0 for each fxo
(I'm not at work now, I can post the results if it'll help).
Make sure you run fxotune from 1.4 (or
Curious if anyone has heard of, or implemented a solution for external
ringers with Cisco phones being used in a noisy / industrial environment
where the standard ringer is not loud enough to be heard...
Cory Andrews
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Please any one knows a script for IPphone auto configuration file from
asterisk(Ip phone Provisioning),
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without
Dears
Do any one have an idea to make a redundant plan for asterisk ,if a call
established between two points and the server interface became down ,do we
you have an idea how to let the call established till the collie or the
caller hang-up.Or a hardware redundancy
Regards
Hi,
I am having a weird problem with one of my incoming lines. After a
reboot everything is fine if I disconnect the line from the wall and
reconnect it. After an hour or so the lies goes busy but no indication
of this shows up on the Flash Operator panel. I also do not see anything
in the
Hi List,
I recently signed up with Voxbone to get some International DIDs. I
was just about to purchase a DID this morning... but when I went to
get it voxbone wanted the end user's address information. So I
started to put it in... unfortunately... the end-user is in the
U.Sbut the
Doug Crompton wrote:
The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling.
Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be
the line1 tab on spa3000. This applies to the fxo (pstn) also if you are
using it for such things as ivr's.
This will only work
anyone care to share their config for realtime queues - I'm having a
problem getting it to work ;)
Julian
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That's better - the originating phone now auto-answers. Thanks.
Unfortunately the terminating phone also auto-answers, so I guess I've
got to find out how to not inherit the ALERT_INFO variable across the
channels (I've tried ALERT_INFO without the '_' prefix, but then the
originating phone does
Andrew,
Thanks, for the response. That is a very clean solution and much less
work/complication, however, I am not sure that the security guy for this
network will allow me to put up the asterisk box dual homed to the public IP
and the LAN. If there is not another feasible way then I may end
On Fri, Jan 12, 2007 at 10:58:16AM -0500, Doug Crompton wrote:
The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling.
Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be
the line1 tab on spa3000. This applies to the fxo (pstn) also if you are
using it for
[Replying to my own post, but it's good news!]
Yipee - sorted.
I needed 2 contexts:
[click-to-call-originate-custom]
exten = _X.,1,NoOp(Click to Call Originator)
exten = _X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten = _X.,3,Goto(from-internal,${EXTEN},1)
[click-to-call-target-custom]
Hi,
I am having an odd problem with one of our Zyxel P-2000W v2 wireless SIP
phones. The phone can no longer receive calls. DND is not turned on and
the phone has the exact same configuration as the other 2 phones (they
each have unique extensions and such but all other settings are the
same.)
Hi,
I am having a weird problem with one of my incoming lines. After a
reboot everything is fine if I disconnect the line from the wall and
reconnect it. After an hour or so the lies goes busy but no indication
of this shows up on the Flash Operator panel. I also do not see anything
in the the
For some reason Dovecot also doesn't like the // in the string either.
After removing the imapflags option in voicemail.conf, It tries
{localhost:143/imap//user=username}INBOX
I also tried it with the novalidate-cert option and dovecot chokes on
it as well...
It's almost like if there are no
Intresting, there is no way to have *1 not place the tone while on the phone
is it?
On 1/12/07, Steve Davies [EMAIL PROTECTED] wrote:
I think that the support for the SNOM button (which uses a SIP message
to request the recording) is part of the bristuff patch, otherwise
only *1 will work.
We put analog ones in those areas and then Dialed both the SIP phone and the
analog device. seems to work rather well.
On 1/12/07, Cory Andrews [EMAIL PROTECTED] wrote:
Curious if anyone has heard of, or implemented a solution for external
ringers with Cisco phones being used in a noisy /
In article [EMAIL PROTECTED],
Matt [EMAIL PROTECTED] wrote:
Hi List,
I recently signed up with Voxbone to get some International DIDs. I
was just about to purchase a DID this morning... but when I went to
get it voxbone wanted the end user's address information. So I
started to put it
In article [EMAIL PROTECTED],
Matt [EMAIL PROTECTED] wrote:
Hi List,
I recently signed up with Voxbone to get some International DIDs. I
was just about to purchase a DID this morning... but when I went to
get it voxbone wanted the end user's address information. So I
started to put it
Eric ManxPower Wieling wrote:
Doug Crompton wrote:
The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling.
Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be
the line1 tab on spa3000. This applies to the fxo (pstn) also if you are
using it for such things
Ralph
Morning - no, what I meant was that *I* was kind of new to * and I'm
learning AGI while working thru my first AGI script too
Let me see if I can explain my point better this time, it was a pretty
weak attempt below.
The key point is that the $argv[] array variables are only available
Julian Lyndon-Smith wrote:
I cannot seem to find any reference to labels in realtime extensions -
using 1.4.
I've googled until my eyes have bled, and also scoured voip-info.org.
Is there anything that helps me here ?
You have to have numbered priorities with realtime.
This is because (as
Matt wrote:
Hi List,
I recently signed up with Voxbone to get some International DIDs. I
was just about to purchase a DID this morning... but when I went to
get it voxbone wanted the end user's address information. So I
started to put it in... unfortunately... the end-user is in the
Chuck Bunn wrote:
Hi,
I am having an odd problem with one of our Zyxel P-2000W v2 wireless SIP
phones. The phone can no longer receive calls. DND is not turned on and
the phone has the exact same configuration as the other 2 phones (they
each have unique extensions and such but all other
Hi list,
I have a strange problem.
Sometimes when a user is making a call asterisk is using 2 lines instead of
one.
The server is running Asterisk 1.2.12 with 2 tdm400.
Do you have any idea?
Thanks!
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Hi list,
I have this problem:
when someone is making a call, with asterisk and a TDM2400P connected to 8
fxo lines, the sound is good, but if three, for people are calling at the
same time the sound got worse and worse.
Using other voip cards the sound is much better even with all
Dear Sirs,
I'm looking for a tool which can do the following:
1) higher level of administration, only one person, it can create domains
and per-domain administration accounts
2) lower level of administration, many persons, each can add new extensions
and change passwords with their domains.
I am not sure that the security guy for this network will allow me to put
up the asterisk box dual homed to the public IP and the LAN.
Your security guy needs to go back to school. If eth0 is on the LAN and eth1
is on the WAN, and the WAN connection is properly secured with only the
ports you
just as a followup and potential solution to the passing of values, I've
been setting variables at the top of the dialplan using :
exten = 100,n,Set(my_var=SomeValue)
Now I can modify my_var in the dialplan, and I can get/set my_var in the
AGI script via GET VARIABLE and SET VARIABLE
You'll
Anyone have a suggestion on where I can get a decent new MB with 5v
capable PCI slots. It seems like every decent server MB on the market
has 3.3V slots only.
Is diving into the junkbin my only choice if I can't afford to replace
the 5v quad-T1 wildcard?
Thanks
Mark Farver
Peter Bowyer schrieb:
On 11/01/07, Markus Amann [EMAIL PROTECTED] wrote:
Hi
i have a trunk up and running with Asterisk and Sipgate.de and i can
make call out but no call in but the Enddevice Status on the Sipgate
Webpage says offline.
Maybe somebody had the same problem in the past and can
Well here's what I did to finally get some sort of ground. I was going
to move the card to a different slot, when I opened up the box I saw
there was a power split between the connector the card to feed a fan
in the system. I removed this split so it was straight in from the
power supply. I
Peter Bowyer schrieb:
On 11/01/07, Markus Amann [EMAIL PROTECTED] wrote:
Hi
i have a trunk up and running with Asterisk and Sipgate.de and i can
make call out but no call in but the Enddevice Status on the Sipgate
Webpage says offline.
Maybe somebody had the same problem in the past and can
From: Ralph Liebessohn [EMAIL PROTECTED]
On 1/11/07, Mike D'Ambrogia [EMAIL PROTECTED] wrote:
Ralph
Kind of new to asterisk, and really new to AGI but it looks like you were
trying to have the AGI script tell asterisk to read and lay the results
into
my_var and then regain control in the
I am using spa3000 hardware - 2.0.1(5673) firmware - 3.1.3(GWa)
I have used newer firmwares but find that 3.1.3 had less echo problems.
Connect a real analog phone to spa3000 fxs. Call it from another source,
when connected send DTMF tones from that source. You should hear at least
100ms or more
From: Mike D'Ambrogia [EMAIL PROTECTED]
You'll need to modify your current overall programming strategy in order
to implement this. But do the READ that you are attempting from within
the dial plan and not the AGI script, and either update my_var prior to
calling the AGI, or pass my_var in to
Markus Amann wrote:
Peter Bowyer schrieb:
On 11/01/07, Markus Amann [EMAIL PROTECTED] wrote:
Hi
i have a trunk up and running with Asterisk and Sipgate.de and i can
make call out but no call in but the Enddevice Status on the Sipgate
Webpage says offline.
Maybe somebody had the same problem
Kate Kretz wrote:
Dear Sirs,
I'm looking for a tool which can do the following:
1) higher level of administration, only one person, it can create
domains and per-domain administration accounts
2) lower level of administration, many persons, each can add new
extensions and change passwords
Colin Anderson wrote:
I am not sure that the security guy for this network will allow me to put
up the asterisk box dual homed to the public IP and the LAN.
Your security guy needs to go back to school. If eth0 is on the LAN and eth1
is on the WAN, and the WAN connection is properly
Hi,
Sorry I forgot to mention that the phone is showing registered and 'sip
show peers' shows that it is registered. Also the user can make outgoing
calls without a problem.
thanks
Eric ManxPower Wieling wrote:
Chuck Bunn wrote:
Hi,
I am having an odd problem with one of our Zyxel
In the current setup, asterisk is behind a different nat/firewall than
the LAN phones. The phones are using sccp. If the asterisk box is
compromised, it is not on the local LAN. This is what I think he
doesn't want to give up.
Andy
-Original Message-
From: [EMAIL PROTECTED]
On Friday 12 January 2007 1:42 pm, Ken Williams wrote:
Thanks for the tips, any ideas why either the power split or the card
slot would keep add echo, or at least make it so my echo settings
weren't taking? As soon as I modified the hardware the echo was vastly
improved, then the minor
Hello,
I have a small problem with musiconhod and dial. When i use the following
dialplan :
exten = s,1,Dial(exten||m)
I have an music during 5 seconds after the call is stoped if the destination
don't response.
With this dialplan, the music play allaws but the destination don't ring.
exten =
Hello all, iam setting up an asterisk box behind NAT to get SIP calls from
outside or internet.
In that eschema i can setup SIP calls but, while from the outside nat people can
hear me, Im unable
to listen anything behind NAT. Out of firewalls settings( I checked this to port
fowarding) what can
i am having a problem where the phones are registered and can make outgoing
calls but all incoming calls go directly to voicemail and do not ring any of
the phones
any ideas?
--
kevin
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Funny thing is now that I've been involved with VoIP for a while..
I hear echo on the regular PSTN :P There are a few numbers I call
regularly that are out in podunk, PA.. and I can hear the echo on the
line.. hehe.
On 1/12/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Friday 12
Chuck Bunn wrote:
Hi,
Sorry I forgot to mention that the phone is showing registered and 'sip
show peers' shows that it is registered. Also the user can make outgoing
calls without a problem.
A phone does NOT have to be registered in order to make outgoing calls.
Registration is only
actually, I was looking for Web thing. I'd like to delegate my customers (
i.e. companies) to manage their extensions via Web.
what are those *.ael files ?
let me desribe the task more precisely. we run telecom, and we sell phone
numbers to companies. what do we want to do ... I'd like to bound
I would really really appreciate it if someone would forward me
Release_GXP2000-BT200_1.1.0.16.zip for the Grandstream GXP-2000. I need
it in order to upgrade the phones to the latest stable.
Thanks,
Alex
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In the current setup, asterisk is behind a different nat/firewall than
the LAN phones. The phones are using sccp. If the asterisk box is
compromised, it is not on the local LAN. This is what I think he
doesn't want to give up.
Oho, now I see. Well, there's the philisophical endless debate
Is there a local dialplan on the phone?
Maybe these phones were recently upgraded or reset to factory and lost the 4XXX
dialplan.
That is where I would start.
--
--
Steven
http://www.glimasoutheast.org
Marco Mouta [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hi all,
I'm using trixbox and the asterisk agi. I downloaded a cepstral voice and
worked with it until I got the code to do what I wanted. I then registered
the voice today to get rid of the 'this voice is not yet registered, stuff
yet it still does that.
Any ideas on how to fix this? It told me my
On Jan 12, 2007, at 10:07 AM, Robert Norton - SophMedia LLC wrote:
Hey Guys,
I apologize for my ignorance on this one.
I've got several 7960s running on Asterisk1.4 with 15 or separate
queues and am trying to figure out a way to identify to the 7960s,
what queue the incoming call is on?
On 1/12/07, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
I am having a weird problem with one of my incoming lines. After a
reboot everything is fine if I disconnect the line from the wall and
reconnect it. After an hour or so the lies goes busy but no indication
of this shows up on the Flash
session border controller can be a solution.
Thanks
Khaled [EMAIL PROTECTED] wrote:
Dears
Do any one have an idea to make a redundant plan for asterisk ,if a call
established between two points and the server interface became down ,do we
you have an idea how to let the
On Jan 12, 2007, at 12:42 PM, Ken Williams wrote:
Well here's what I did to finally get some sort of ground. I was going
to move the card to a different slot, when I opened up the box I saw
there was a power split between the connector the card to feed a fan
in the system. I removed this
blackwater dev wrote:
I'm using trixbox and the asterisk agi. I downloaded a cepstral voice
and worked with it until I got the code to do what I wanted. I then
registered the voice today to get rid of the 'this voice is not yet
registered, stuff yet it still does that.
Any ideas on how to
Hi,
I am using 2 TDM400P in a Centos 4.3 box. When we call from a cell phone
to the line we get a busy signal...
Thanks
Lacy Moore - Aspendora wrote:
On 1/12/07, *Chuck Bunn* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
I am having a weird problem with one of my incoming
Hi,
Here the CLI output. SIP 498 is the calling phone and 411 is the phone
that cannot receive a call:
login as: root
[EMAIL PROTECTED]'s password:
Last login: Fri Jan 12 12:55:55 2007 from 10.0.0.72
[EMAIL PROTECTED] ~]# asterisk -r
Asterisk 1.2.7.1, Copyright (C) 1999 - 2006 Digium, Inc.
Am Freitag, den 12.01.2007, 11:31 -0500 schrieb Matt:
Hi List,
I recently signed up with Voxbone to get some International DIDs. I
was just about to purchase a DID this morning... but when I went to
get it voxbone wanted the end user's address information. So I
started to put it in...
Solved :)
Added at sip.conf :
silencesuppression=yes
Regards,
On Wed, 2006-12-13 at 21:51 -0300, Josué Conti wrote:
Kevin, contributes with the list, somebody can have this problem and
you it can help. The list is here for helping, but also we must
contribute with it. :)
Best Regards
What about Huawei to Asterisk ?
Is it the same problem with that ?
I get a weird error, call comes in, i answer and it disconnects.
Rehan
Subject:Re: [asterisk-users] Asterisk to a Huawei softX3000
problem has
already been solved ï¼
From: Guillermo
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