Any why would that make it work with cvs-head but not cvs-stable?
By the way, I no_load the module so I can load it manually later and see the
console output. Either way, it still kicks out the error and crashes, or
just kicks out the error if I no_load it first...
-Original Message-
Hi,
I have just bought another TDM400P card from Digium directly, purchased
last Thursday, received it today:
Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1
1 WCTDM/0/0 FXOKS (In use)
2 WCTDM/0/1 FXOKS (In use)
3 WCTDM/0/2 FXSKS (In use)
4 WCTDM/0/3
I was wondering if anyone has managed to get a working solution with
asterisk behind a NAT connecting to external sip users behind another NAT.
I have been using iax for the asterisk box without any issues including
internal and external connections as well as connecting multiple asterisk
boxes
Hi,
I make a call to my mobile, now I would like to transfer the call to another
extension from my mobile, I try with #1 (which is configured in
features.conf as unattended transfer), and pbxtransfer is played back to me,
but when I try to enter an extension I just get an error.
What am I doing
Tony Mountifield wrote:
Yesterday I wrote:
I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on
Fedora Core 3.
[... snip ...]
Well I gave up with chan_h323, which is a pity, because it should be the
solution that is better integrated with Asterisk. I would still like to
hear from
Hi all,
I have a VoIP PBX box with asterisk and one x100p card. I setup some sip users in sip.conf.
The asterisk will quit aperiodically, sometime it will work for several days before quit, but I find its quit time is almost in 18:00 to 19:00.
I can not find any clue from log file. The
Hi all,
I built a VoIP PBX box with asterisk and one x100p card. Every thing is ok except there is a short audio latency from PSTN to SIP and no delay in the reverse direction.
At the beginning of a call, the latency is not very long, but it becomes more and more obvious along with time. If
Quintum are good
Selon Chad Brown [EMAIL PROTECTED]:
There are many analogue gateways to choose from:
http://www.voip-info.org/wiki-VoIP+Gateways
Does anyone have experience with several that could point me in the
right direction? I need 5-8 ports. At some point I see us going digital
but
In article [EMAIL PROTECTED],
Michael Manousos [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
When testing the ability of dual 3GHz Xeons to handle many simultaneous
OH323 calls (G.711 so no heavy transcoding), I discovered that chan_oh323
is EXTREMELY profligate with file descriptors!
Hi there!
I compiled asterisk on a HP DL380 G4 with Suse Linux Enterprise Server 9
(gcc 3.3.3). It compiled without any errors.
I also had no problems with installing my digium hardware (WC TE110P).
But when I try to start asterisk, I get the following error messages:
The error messages
Apr 12
Hello,
Is there a way to put a voicemail quota to a SIP user? I mean a quota on the
user's mailbox instead
of a particular message of the user like the 'maxmessage' does currently.
Quata can be total file size of message or
total minutes of messages of a mailbox.
Thanks
Foong
I'm aware of the legal issues surrounding my request, but any help
technically would be greatly apreciated
On site we have a fully staffed hospital and fire service (its a
temporary event for a childrens charity) and an onsite 911 number.
If a user dials the number, they goto the emergency crew,
David John Walsh wrote:
I'm aware of the legal issues surrounding my request, but any help
technically would be greatly apreciated
On site we have a fully staffed hospital and fire service (its a
temporary event for a childrens charity) and an onsite 911 number.
If a user dials the number, they
http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone
Sorry for the 170 or so who have already signed. This list supposedly
has 10,000 or more subscribers. 170 isn't very impressive. Please
sign!
___
Asterisk-Users mailing list
Qiao Yuansong wrote:
Hi all,
I have a VoIP PBX box with asterisk and one x100p card. I setup some
sip users in sip.conf.
The asterisk will quit aperiodically, sometime it will work for
several days before quit, but I find its quit time is almost in 18:00
to 19:00.
That is the time
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Michael Manousos [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
When testing the ability of dual 3GHz Xeons to handle many simultaneous
OH323 calls (G.711 so no heavy transcoding), I discovered that chan_oh323
is EXTREMELY profligate with
However, and I know this is a running issues, I cannot get external sip
users behind a NAT to be able to successfully connect to asterisk when it's
behind a NAT as well.
I have done port forwarding at both ends dealing with the usual ports of
5060, 4569 and 5036 as well as opening up the rtp
HI:
I have compiled and installed Asterisk 1.0.7 without
any problems.I have also installed mysql and
DBD::mysql successfuly / When I tried to make
asterisk-addons, it
showed me the problem like these:
[EMAIL PROTECTED] asterisk-addons]# make install
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE
Hello list,
I have a question about Asterisk and H323. Wich H323
channel driver is the best for Asterisk? Asterisk-oh323 or OH323. Im
asking this question because I have big problem running my asterisk with
asterisk-oh323. all is well installed but when there are some calls, my
asterisk
Hello,
I still try to connect a TE110P card to a TMS2 card in a Siemens HiPath
3750.
The TMS2 card can be used to connect to an NT (Amtsanschluss)
or to connect to another S2M-Line (PRI). When connecting to another PRI,
I can select between CorNet (proprietary), ECMA-QSIG and ISO-QSIG.
It seems
No offense taken. In fact, it sounds like you have 'spotted' an error or
potential error in the way I have configured this. I would appreciate any
and all comments/suggestions you may have on how I could configure asterisk
to change dtmfmode depending on the codec being used.
Thanks,
-
Hi Rich,
Thanks for writing back to me. Yep, just like you, I too am looking for a
lower bandwidth codec for my outbound. And, yes, broadvoice only officially
supports G.711. That being said, is there even a way to do this scenario in
asterisk?
Thanks,
- Andre
-Original
rusty*CLI show version
Asterisk CVS-HEAD-03/26/05-17:05:44 built by [EMAIL PROTECTED] on a
i686 running Linux
rusty*CLI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Monday, April 11, 2005 6:47 PM
To: Asterisk Users Mailing List -
Hi,
i am trying to use my fxo card for analog calls ..
fxo card seems to be ok, working properly but when trying to call
outside ( from a sip phone ot pstn ) i get the following error on
asterisk .
Apr 12 11:59:24 DEBUG[4231]: chan_sip.c:4633 build_route: build_route:
Contact hop: Drugo
The only firmware upgrade procedure is for you to call digium support.
Hi,
I have just bought another TDM400P card from Digium directly, purchased
last Thursday, received it today:
Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1
1 WCTDM/0/0 FXOKS (In
Version 0.80 - 12. April 2005.
* Swedish language added - thanks Daniel Nylander
* Bug fixes
Download for FREE: http://ipswitchboard.thorben.dk
Would you like to help translate IPS into your language? Please click the
link below for details. I will add your language as soon as I receive it.
Now it would be even more interesting to see if Cisco or maybe
Siemens/Polycom would bring out a firmware for IAX, now that would be a
revolution.. :)
On Tue, 12 Apr 2005, Wilson Pickett wrote:
http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone
Sorry for the 170 or so who have
Funny, they sell these old cards.. it seems like they are selling refurbs
as new.. ... anyways RMA is on its way, would be nice if they would send
one as a replacement first, so that we could continue our work and don't
have to delay it.
On Tue, 12 Apr 2005, Rich Adamson wrote:
The only
Rich Adamson wrote:
Hi, I want to download the CVS HEAD version. Any one can show how to get
this version ?
Is the version from: http://www.asterisk.org/index.php?menu=download the
CVS Head version?
How I can check if my version is CVS HEAD or not?
phoenix*CLI show version
Asterisk
Hi all
Inewly added a X101P into my asterisk that already have a TE410P running 2 E1s namely span1 and span2
I am unable to get * to recognized the new X101P after i did modprbe wct4xxp and then modprobe wcfxo. ztcfg -vv reported all 63 channels are configured but zttool tells me that span 1,2,3
Bruno Hertz wrote:
Joe S [EMAIL PROTECTED] writes:
Hi,
I am new with asterisk. I was wondering if there is a way to call a
OH323 user or SIP user using Netmeeting/SJPhone with H323 as the
default protocol without having a gatekeeper.
I can make a call from SIP to OH323 by specifying it in the
Wilson Pickett wrote:
http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone
I´ve signed before (in 90th posicion).
Sorry for the 170 or so who have already signed. This list supposedly
has 10,000 or more subscribers. 170 isn't very impressive. Please
sign!
Thank you for the reply.
exten = 1,1,SetVar(CALLFILENAME=${CALLERIDNUM})
exten = 1,2,SetVar(CALLTIME=${DATETIME})
exten = 1,3,SetVar(CALLPATH=/var/calls)
exten = 1,4,Monitor(wav,${CALLPATH}/${CALLTIME}-${CALLFILENAME},m)
? exten = 1,5,DIAL(SIP/something,15,t) - do I need to change SIP/something
In article [EMAIL PROTECTED],
Henry Jensen [EMAIL PROTECTED] wrote:
Is there any way I can switch the TE110P card to NT-Mode ?
In /etc/asterisk/zapata.conf, change signalling=pri_cpe
to signalling=pri_net
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
On Tue, Apr 12, 2005 at 12:23:52PM +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Henry Jensen [EMAIL PROTECTED] wrote:
Is there any way I can switch the TE110P card to NT-Mode ?
In /etc/asterisk/zapata.conf, change signalling=pri_cpe
to signalling=pri_net
Wait a minute,
If there are multiple asterisk boxes in use is there a
way to "link" them
together so when the manager api command "show channels" is executed
ALL boxes are reported?
Certainly I can connect to each box and execute the command show
channels
but was just wondering if there was something
Guillermo Salas M. [EMAIL PROTECTED] writes:
Bruno Hertz wrote:
Joe S [EMAIL PROTECTED] writes:
Hi,
I am new with asterisk. I was wondering if there is a way to call a
OH323 user or SIP user using Netmeeting/SJPhone with H323 as the
default protocol without having a gatekeeper.
I can make a
In article [EMAIL PROTECTED],
Henry Jensen [EMAIL PROTECTED] wrote:
On Tue, Apr 12, 2005 at 12:23:52PM +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Henry Jensen [EMAIL PROTECTED] wrote:
Is there any way I can switch the TE110P card to NT-Mode ?
In
Just set qualify=yes in sip.conf
On Apr 12, 2005 3:41 AM, Ronald Wiplinger [EMAIL PROTECTED] wrote:
Is there a possible settings for a remote SIP phone, so that a router
will not close the connection due to long time inactivity?
___
Asterisk-Users
Thanks for writing back to me. Yep, just like you, I too am looking for a
lower bandwidth codec for my outbound. And, yes, broadvoice only officially
supports G.711. That being said, is there even a way to do this scenario in
asterisk?
Yes, there are frequently multiple ways to do things
Funny, they sell these old cards.. it seems like they are selling refurbs
as new.. ... anyways RMA is on its way, would be nice if they would send
one as a replacement first, so that we could continue our work and don't
have to delay it.
They can, its called cross-shipment, but they need a
On Monday 11 April 2005 15:15, Jesus Mogollon wrote:
Hi all
I installed asterisk on a dual PIII 700 with two NICs. I then proceeded to
configure both NICs with bonding enable (bonding miimon=100 mode=1). I know
certain features (like load balancing) under a bonded configuration is not
Hi Guys,
Is it possible to get the UAC supported codec list when making
a call. I want to assign to variable1 and variable2 the first 2
supported codecs using AGI script e.g.
$variable1=g723
$variable2=g729
Somebody can help me ? Any help is appreciated.
Thanks,
Pavel Siderov
Hi,
Actually I guess what I am looking for is semi-sealed box that I can add
1 or 2 PCI cards too. A regular PC work work in most cases since I do
not want a keyboard or mouse attached to it. I do not want users
screwing with the system. If it is sealed with no monitor/keyboard/mouse
then they
Try re-downloading Asterisk-Addons. It sounds like you have the version
that is meant for CVS HEAD and not the stable 1.0 series.
-mishehu
lie ka wrote:
HI:
I have compiled and installed Asterisk 1.0.7 without
any problems.I have also installed mysql and
DBD::mysql successfuly / When I tried
But voicemailboxes have to exists on all asterisk servers
right?
Also, what happens if for example, the user is accessing
his VMB on server 1 and changes his password, then travel to where server 2 is
and tries to access his VMB? the config on server2 would still have the old
one so
Version 0.80 - 12. April 2005.
You spit out the versions faster than I can reinstall them :)
Did you by any chance had the time to take a look a transfer problem
when there are two active calls on a monitored extension ?
Ivan
___
Asterisk-Users
Depending on how many users you want to support and price, there are
lots of options.
Smallest form factor will be SOC (System on Chip)
These are little more costly and not going to carry a huge load.
Next would be Mini-ITX
A bit bigger and will carry more load.
VIA is the king in this arena
Has anyone ever explored what would be required to enable * to produce
NENA standard CAMA signaling for interconnection with conventional e911
services?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Monday 11 April 2005 22:36, Tim Connolly wrote:
I'm assuming I would see an error if this was bad:
ldd /usr/lib/asterisk/modules/chan_zap.so
linux-gate.so.1 = (0xe000)
libpri.so.1 = /usr/lib/libpri.so.1 (0xb7f89000)
libtonezone.so.1.0 =
Hi there
I am using Meetme and am connecting with clients that use
VAD. The clients have been built with RTC Client API. What Meetme seems to do
is cut users off from the conference if it does not receive any audio packets
from the user for 1 minute 45 seconds. The solution I have found
Hello all,
Sometimes my ADSL internet conection, gets down and I cannot access to internet.
When this happens, my asterisk gets crazy and it cannot route my calls.
Actualy I have an scape secuence (111) followed with the PSTN number,
and the call is routed trought my ISDN lines.
When my ADSL gets
Eric Wieling wrote:
[EMAIL PROTECTED] wrote:
Hi,
How can i implement VAD/DTX using zaptel with asterisk towards PSTN.
TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not
even a valid idea.
Doing VAD on audio coming _from_ the TDM world certainly is something
you might want to
Have you actually tried that cvs-up script? Not knowing how to check my
version and using that cvs-up thing caused me a lot of grief. I thought
I was on CVS-HEAD when I was on um.. something else that didn't really
have a version in show version.
Andy Hamilton wrote:
The fastest way to obtain
can anyone help ??
trying to get Polycom IP300 to utilize both lines,
would like calls to rollto open linewhen incoming call arrives while
user is on line 1. Looked everywhere and tried many things with no
luck.
___
Asterisk-Users mailing list
LeeLee-
Try configuring all 4 spans first and then the single channel (125)
above that - works for me.
Modprobe in the same order, then ztcfg.
Regards
Scott Stingel
www.evtmedia.com
Lee Lee wrote:
Hi all
I newly added a X101P into my asterisk that already have a TE410P
running 2 E1s namely
Check out call parking. It's basically the same thing.
-mishehu
Ben Ryan wrote:
I have a question probably for those in the know in business Asterisk
solutions. I have searched high and low but have not been able to get
any answers. I hope there is someone on the list that can answer my
question.
I have no Idea of the strange errors, but as far as i know, the proper
way of calling is:
Zap/g${group}/${phone_number}
where ${group} is a valid group inside zapata.conf, and
${phone_number} is the desired PSTN phone to call. In you email you
wrote the messages and i can see that you missed
On Tue, 2005-04-12 at 05:14 -0700, Lee Lee wrote:
Hi all
I newly added a X101P into my asterisk that already have a TE410P
running 2 E1s namely span1 and span2
I am unable to get * to recognized the new X101P after i did modprbe
wct4xxp and then modprobe wcfxo. ztcfg -vv reported all 63
On Tue, 2005-04-12 at 13:40 +0200, Thorben Jensen wrote:
Version 0.80 - 12. April 2005.
* Swedish language added - thanks Daniel Nylander
* Bug fixes
Any chance of integrating some sort of input text box, where you can
just type in the extension number and hit enter to transfer a call?
On Tuesday 12 April 2005 10:18 am, MobilPete wrote:
can anyone help ??
trying to get Polycom IP300 to utilize both lines, would like calls to roll
to open line when incoming call arrives while user is on line 1. Looked
everywhere and tried many things with no luck.
Do you have your lines
Hi *,
Does anyody know, what power consumption this card have? The technical
descripten is really quiet at this point ..
Tfh,
Oliver
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Steve Kann wrote:
Eric Wieling wrote:
[EMAIL PROTECTED] wrote:
Hi,
How can i implement VAD/DTX using zaptel with asterisk towards PSTN.
TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not
even a valid idea.
Doing VAD on audio coming _from_ the TDM world certainly is something
It is alright to sell hardware, and it is alright to sell labor when
dealing with open source software. But selling licensing on something
that does not exist (extension licensing???) is wrong. What if somebody
started charging extra licensing to use the include music tracks for
MOH? Also,
On Apr 11, 2005 8:11 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
nat=no
disallow=all
allow=g729
allow=g726
auth=plain
context=default
canreinvite=yes
username=USERNAME
secret=PASSWORD
dtmfmode=info
fromdomain=REALM
fromuser=USERNAME
qualify=1000
insecure=very
I am using
Andres wrote:
Can you confirm if there will be some sort of DSP daughther card add on
of some sort for the DS3000 so that we can run G729 transcoding? I
don't see how the DS3 interface would be usefull unless we could offload
transcoding stuff to onboard DSPs. Or is Digium only going to
Andrew Kohlsmith wrote:
secondary card for DSP functions is very inefficient of the PCI bus. I'd be
curious to know if the Digium cards can even do PCI-PCI DMA.
The Digium TDM cards can DMA into any RAM accessible over the PCI bus,
regardless of whether it is located on the motherboard or on a
What is considered an acceptable time delay between
two servers for a fair (not neccessarily great) voice
quality.
I use voipjet to connect my calls from iax2 to the
pstn. Although the sound quality is good, there is
considerable time delay, I wait seconds before the
other party hear what I say.
On 11:27, Tue 12 Apr 05, Wilson Pickett wrote:
However, and I know this is a running issues, I cannot get external sip
users behind a NAT to be able to successfully connect to asterisk when it's
behind a NAT as well.
I have done port forwarding at both ends dealing with the usual ports of
If you have two lines registered to one phone then you need to do the
following...
This assumes extensions 1001 and 1002 are your line appearances...
exten = 1001,1,Dial(1001,20,trf) ;we are dialing line 1
-- After 20 seconds it will timeout and go to the next line
exten =
chawki hammoud wrote:
What is considered an acceptable time delay between
two servers for a fair (not neccessarily great) voice
quality.
I can't really deal with anything over 150ms, although regular users
will tolerate ~200ms.
I use voipjet to connect my calls from iax2 to the
pstn.
Message: 14
Date: Mon, 11 Apr 2005 17:35:05 -0400
From: dean collins [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home
and AMPfor over 1000 dollars
To: [EMAIL PROTECTED],Asterisk Users Mailing List -
Non-Commercial Discussion
Thanks for that Rich. Etheral trace is going to be almost impossible
for various reasons, but will try the other two options.
Can't find much online re. debugging - any chance of killing the box by
turning this on?
SIP show channels and the various CAPI show commands do not show
Sign it: http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone
Now it would be even more interesting to see if Cisco or maybe
Siemens/Polycom would bring out a firmware for IAX, now that would be a
revolution.. :)
Cisco et al won't exactly be blown away by the not even 200 sigs :)
I
we tried both, setting it as same and also seperate. but niether worked.
- Original Message -
From: Josiah Bryan [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 12, 2005 9:41 AM
Subject: Re:
Around 250ms max. Over that and you will have the walkie-talkie effect
you are experiencing.
So with you 600ms delay you are way over the top.
There is also the delay on the call on the PSTN side you have to take
into account.
For example, I am in Europe and making a call to the UK via Voipjet is
On Tuesday 12 April 2005 10:18 am, MobilPete wrote:
can anyone help ??
trying to get Polycom IP300 to utilize both lines, would like calls
to roll
to open line when incoming call arrives while user is on line 1.
Looked
everywhere and tried many things with no luck.
Do you have your lines
Ronald Wiplinger wrote:
Adam Goryachev wrote:
On Tue, 2005-04-12 at 13:40 +0200, Thorben Jensen wrote:
Version 0.80 - 12. April 2005.
* Swedish language added - thanks Daniel Nylander
* Bug fixes
Any chance of integrating some sort of input text box, where you can
just type in the extension
Julian J. M. wrote:
Just set qualify=yes in sip.conf
This I have already, but does not help.
I believe it is the ADSL router at the remote end, which may disconnect
due to inactivity.
I think I can change the ttl parameter on the phone, but than I have to
go there. I was looking for something
[EMAIL PROTECTED] is believed to have said:
http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone
Sorry for the 170 or so who have already signed. This list supposedly
has 10,000 or more subscribers. 170 isn't very impressive. Please
sign!
Just signed; more hardware side support to the
mmm i think Agi by itself does not provide a way to do so. And the
codecs are negotiated depending upon the codec that both call sides
support. So, i belive that the only way is making your own
implementation of AGI in res_agi.c :)
Hopefully someone will come up with a better idea :-)
best
[EMAIL PROTECTED] wrote:
Andres wrote:
Can you confirm if there will be some sort of DSP daughther card add
on of some sort for the DS3000 so that we can run G729 transcoding?
I don't see how the DS3 interface would be usefull unless we could
offload transcoding stuff to onboard DSPs. Or
Polycom enables call waiting on each line button. If you wish the
second call to go directly to the second button you need o keep track
of this with group in * and control with your dial plan.
On Apr 12, 2005, at 9:41 AM, Josiah Bryan wrote:
On Tuesday 12 April 2005 10:18 am, MobilPete wrote:
This guy is not selling extension licensing, he is selling a pre-configured
system and charges extra to configure more extensions and says other people
charge extra licenses for extensions of which I can only find the big PBX
manufacturers that do that. Regardless, you sure can charge extension
Sorry about before, I sent the message from the wrong address and didn't
repaste the entire message when I sent it from the right address...
I am having a problem using my backup dialout termination from asterisk.
The server I am registered to for back up is running SER 0.90. If I dial
NUMBER1,
On Apr 12, 2005 9:38 AM, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
Actually I guess what I am looking for is semi-sealed box that I can add
1 or 2 PCI cards too. A regular PC work work in most cases since I do
not want a keyboard or mouse attached to it. I do not want users
screwing with the
Does anyone know how setting the TOS bits in iax.conf corresponds to
the Cisco TOS types?
For example, if I set:
tos=0x04
in iax.conf, and on the Cisco, I use:
access-list 110 permit ip any any tos 4
I can't get the Cisco to match any packets. I've tried various
combinations of numbers on both
Hi everyone,
I have the following annoying problem with my Digium TE410
Quad-Pri-Card: I sometimes hear strange noises on bridged calls from our
PBX to the PSTN (a colleage called it clipping?)
We have the following setup running:
PSTN - Asterisk - od PBX
(Trunk one to the PSTN, Trunk two to
I am a little confused as to the purpose of agents. My old phone
system required that a user/agent be logged into a phone in order to use
that phone, regardless if the agent was joining a Queue. It seems that
agents in the context of Asterisk are more for dealing with Queues. So
it seems
Also, what happens if for example, the user is accessing his VMB
on server 1 and changes his password, then travel to where server
2 is and tries to access his VMB? the config on server2 would
still have the old one so you need to sync voicemail.conf on
all servers too ...
If you use the
On Tuesday 12 April 2005 11:00 am, MobilPete wrote:
we tried both, setting it as same and also seperate. but niether worked.
I've never used the IP300, but I do have an IP500 on our network. It has 3
line buttons, each line can do 2 simultaneous calls. Each line button
registers as its own SIP
On Apr 11, 2005 11:49 PM, Greg Boehnlein [EMAIL PROTECTED] wrote:
On Mon, 11 Apr 2005, Noah Miller wrote:
This this may sound ridiculous, but we've had problems with this when
the
users did not plug the handset cord in completely. 8 out of our 12
employees
made the mistake, as
Hi All,
I'm trying to install 2 TDM400x cards on the same [EMAIL PROTECTED] box, and I've currentlyhaving
issues where one card is identified by ztfcg, and the other isn't at all. Any
idea what
i
may be doing wrong here? has anyone got an [EMAIL PROTECTED] working in such a
manner?
Nir
Hi,
I get an echo only from the caller end when I am making calls. I only get
it for some VOIP providers. I am using asterisk Asterisk
CVS-v1-0-03/26/05-16:54:47 and Grandstream HandyTone 486 and 488. My
default codec is ulaw. Is there any way I can reduce the echo without
comprising
What I use. At provantage.com
Part
Description Price
ANTG02V
Antec Mini-Tower with 8 Drive Bays - BLACK 45.93
ASUS1FQ
ASUS A7V400-MX Motherboard KM400A 400/333FSB VID LAN 3PCI 49.49
AAMD16U
AMD Sempron 2600+ Processor-In-A-Box 77.54
SEGE155
Seagate Barracuda 7200.7 40GB EIDE ATA-100 7200
Does anybody know how I could set a default provider for LCDial? Also, how
could I use it for national calls, dialling without international prefix?
TIA,
Alex
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Here it is
exten = s,1,answer
exten = s,2,SetCIDName('PMG')
In a lot of config files I see exten = s,snip ..
Is s just an extension or system variable for all extensions ? or
something else ?
Thanks
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Does anyone know how setting the TOS bits in iax.conf corresponds to
the Cisco TOS types?
For example, if I set:
tos=0x04
in iax.conf, and on the Cisco, I use:
access-list 110 permit ip any any tos 4
I can't get the Cisco to match any packets. I've tried various
combinations
On Tuesday 12 April 2005 11:49 am, Luki wrote:
Also, what happens if for example, the user is accessing his VMB
on server 1 and changes his password, then travel to where server
2 is and tries to access his VMB? the config on server2 would
still have the old one so you need to sync
Bicom Systems wrote:
What is target release date for DS3000P?
That has not been announced; sometime after today would be a safe
assumption :-)
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