On Sun, Jun 03, 2007 at 11:29:57PM -0500, Anton Krall wrote:
> True, maybe I didnt make myself clear on that point, what i meant was, Im
> not looking for an app that would let people click a sip: URL type to
> make a call using their already installed softphone but rather allow any
> user that
One comment if I may:
http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos does need some
cleanup. Let alone merging of the two separate procedures and an update.
Also:
On Sun, Jun 03, 2007 at 06:22:57PM -0400, Lee Jenkins wrote:
> Khaled Chehab wrote:
> >I suffered a lot from installing zap
Hi All,
I need nvlinedetect and installation guide for
my asterisk server version 1.4, I appreciate if some one
can help me on this.
I have spent so much time to search google for the answer but
no luck. The only solution that I found is nvlinedetect (0.9) for
asterisk 1.2. I encountered compil
On 3 Jun 2007, at 03:37, Steve Totaro wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Saturday, June 02, 2007 10:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users]
On 4 Jun 2007, at 08:14, Tzafrir Cohen wrote:
On Sun, Jun 03, 2007 at 11:29:57PM -0500, Anton Krall wrote:
True, maybe I didnt make myself clear on that point, what i meant
was, Im
not looking for an app that would let people click a sip: URL
type to
make a call using their already inst
Has anyone out there tried to mix the envelope metadata for
voicemails into the audio payload that's stored by Asterisk? I would
like to have the CID and Timestamp baked into the beginning of the
WAV file, not just as text in the email itself.
Thanks!
-Rob.
___
On 4 Jun 2007, at 01:00, Andrew Kohlsmith wrote:
On Sunday 03 June 2007 4:30 pm, Alex Crow wrote:
No frills, specs look good, price seems excellent!
http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519
That's a terrible phone. I've tried them. the screen is pretty
much usel
Hi all,
sorry if i have missed something, i was just curious what unicall
actually is, what the main features are and did a quick search on
voip-info.org and plain old google but didn't find some source of
information that simply says : Unicall is this and does that ...
Can anyone point me to som
HI
I'm looking for a card that support both PRI and TDM. Please suggest me ?
thanks
arun
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailm
Hi everybody,
I have bought one 496 to use as ATA for two analog extensions in my office.
I'm experiencing strange behaviuors. The ATA blocks itself and needs to be
rebooted. Sometimes it hungs the lines(ISDN bristuffed HFC single isdn
line). It was update to last available firmware.
Has anybody th
Hi,
we are searching for wireless IP Phones (DECT preferred) with have an
solution for an external telephone book. We don't want to enter all of
our numbers into every telephone, but have one location for all the
numbers and every phone looks them up there, e.g. an ldap server.
We have tried Kirk
On Mon, Jun 04, 2007 at 01:08:15PM +0400, Arun Kumar wrote:
> HI
>
> I'm looking for a card that support both PRI and TDM. Please suggest me ?
Non such card. You need two cards.
Why exactly do you need a single card?
--
Tzafrir Cohen
icq#16849755jabbe
We're running 1.4.0 of asterisk
1.4.2.1 of zaptel
And kernel 2.6.20-1.2316.fc5smp
The information contained in this email is intended for the personal and
confidential use
of the addressee only. It may also be privileged information. If you are not
the intended
recipient then you are hereby no
HI Tobias,
look in www.soft-switch.org/unicall/unicall/index.html ;p
--
Humberto Figuera - Using Linux 2.6.20
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603
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I have 2 servers, one connected to internet and the other is on a private
lan have no access to internet.
On the first server I update the kernel by yum update
And installed asterisk prerequisite module
yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf \ libtool
make automake au
On Sun, 3 Jun 2007, Anton Krall wrote:
Hi Gordon
So, mexuar solution was that java softphone that you talked about?
Yes.
Any other small softphone type solution around, something on the same lines
of what you described, something that the user could download but could be
preconfigured or pa
On Sun, 3 Jun 2007, Andrew Kohlsmith wrote:
On Sunday 03 June 2007 4:30 pm, Alex Crow wrote:
No frills, specs look good, price seems excellent!
http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519
That's a terrible phone. I've tried them. the screen is pretty much useless,
the
independently install each rpm via rpm command :-/
On 04/06/07, Khaled Chehab <[EMAIL PROTECTED]> wrote:
I have 2 servers, one connected to internet and the other is on a private
lan have no access to internet.
On the first server I update the kernel by yum update
And installed asterisk pr
Tzafir
Most small/medium companies have a T1 for all their phone needs.
Internally there is a need for some analog lines.
* Fax Machine - FXS
* Security System (most ask/demand two lines) FXS
* Paging - FXO
* Dialup systems
Andrew
On 6/4/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
On Mon, J
I am using the free
http://www.babarnazmi.citril.com/forum/viewtopic.php?t=7&sid=fd8047cffb13074969d3418064f4eb31
It is working as you described.
It appears to be working well.
--
--
Steven
http://www.glimasoutheast.org
"Anton Krall" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECT
On Monday 04 June 2007 8:24 am, Bryan Laird wrote:
> - Physically the phone feels very light and cheap, that if you were
> to drop it that it might not survive very many of them. The buttons
> feel more
> like a toy than anything else but once you get beyond that it works.
How are they for big
Not perfect solution but aastra sets support xml. Which then can do server
based directory. If you can wait 2 weeks or so, I should be able to tell you
how welworks as its my next project.
D
Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]
For those who just want it to work...
I use CentOS4.4. successfully. I do something which is very odd for a Linux
admin. I do a "install everything". There is/was a reason for this.
I was in a hurry to get a system online and didn't have time for a research
project. I wrote a simple shell script to compile the apps (zaptel, libpri,
ast
Right now I can only speak to the WIP300 but
I've been evaluating it for about a week now and really I have to say
I'm fairly pleased. It works it works //well// but that's not to say
it's perfect.
- Physically the phone feels very light and cheap, that if you were
to drop it that it mi
Hi
I've two boxes connected over IAX2 trunk but suddenly my cli is getting
flood with these messages:
iax2_trunk_queue: Maximum data space exceeded
and b'coz of that my agents are not able to hear any thing.
when this happened that time there were 9 calls.
my * version is 1.2.18 and 1.2.14
t
HI
I bought 20 license from Digium and install in my server and b'coz of some
problem I've to change my server is it possible that I can use those lice
and register again in my new server ?
Is it possible that I'll be able to use those lice in my old box also ?
thanks
arun
On Jun 4, 2007, at 8:31 AM, Andrew Kohlsmith wrote:
On Monday 04 June 2007 8:24 am, Bryan Laird wrote:
- Physically the phone feels very light and cheap, that if you were
to drop it that it might not survive very many of them. The buttons
feel more
like a toy than anything else but once you
Hi
iam keep getting this log in my asterisk log
is this harm anything, and how can stop this, any suggestions
Jun 4 18:21:47 DEBUG[2093] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match Found
Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Com
you can register twice
after that you'll have to explain the reasons of changes to Digium
Olivier
Arun Kumar a écrit :
HI
I bought 20 license from Digium and install in my server and b'coz of
some
problem I've to change my server is it possible that I can use those
lice
and regist
Jim Suber wrote:
I use CentOS4.4. successfully. I do something which is very odd for a Linux
admin. I do a "install everything". There is/was a reason for this.
I was in a hurry to get a system online and didn't have time for a research
project. I wrote a simple shell script to compile the apps
On 6/4/07, Arun Kumar <[EMAIL PROTECTED]> wrote:
HI
I bought 20 license from Digium and install in my server and b'coz of some
problem I've to change my server is it possible that I can use those lice
and register again in my new server ?
Is it possible that I'll be able to use those lice in
On Thu, 2007-05-31 at 15:44 -0500, Rob Schall wrote:
> A simple question but one I can't seem to find easily...
>
> I have 90 or so DIDs. For all outbound calls, I edit the callerid so
> that it will always read out main line's number. This poses a problem
> though, because the CDR detail isn't wr
Tzafrir Cohen wrote:
One comment if I may:
http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos does need some
cleanup. Let alone merging of the two separate procedures and an update.
Also:
On Sun, Jun 03, 2007 at 06:22:57PM -0400, Lee Jenkins wrote:
Khaled Chehab wrote:
I suffered a lot
Gang Chen wrote:
> - Original Message - From: "Mauro Zanin" <[EMAIL PROTECTED]>
> To:
> Sent: Sunday, June 03, 2007 12:32 PM
> Subject: [asterisk-users] Loud noise instead of MOH
>
>
>> Hi Everybody,
>> I'm experiencing this kind of issue.
>> One ASTERISK 1.2.17 is connected to a Bristuf
Gordon Henderson wrote:
On Sun, 3 Jun 2007, Andrew Kohlsmith wrote:
On Sunday 03 June 2007 4:30 pm, Alex Crow wrote:
No frills, specs look good, price seems excellent!
http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519
That's a terrible phone. I've tried them. the screen is
On Mon, 4 Jun 2007, Tobias Wolf wrote:
Hi,
we are searching for wireless IP Phones (DECT preferred) with have an
solution for an external telephone book. We don't want to enter all of
our numbers into every telephone, but have one location for all the
numbers and every phone looks them up there
Our call detail is located in 2 places. The master.csv file and in a
mysql database. All outbound calls have:
exten => _91NXXNXX,1,Set(CALLERID(all)=000-000-)
(the zeros being our number)
Right after that, it moves to the Dial command. The problem we are
seeing, is in both the csv fil
Hi,
I've tested several wifi phone (UtStarcom, Hitachi 3000 and 5000, and one
Siemens). The Siemens is the best one, for a really cheaper price than
hitachi. And was the only one which roams well between AP (same SSID, same
channel) with WPA. Battery is still a problem, especially if the covertu
I am having a difficult time with the transition from agentcallback
login...
Here are a few of the isssues, I am logging in using chan_ local
ie:local/8000 as the extension
Call Detail records no longer show agent/ as the dstchannel
show agents no longer shows the channels state
show queues d
Steven
Have you been able to "custommized" the interface for babar's iax solution?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Sent: Lunes, 04 de Junio de 2007 07:17 a.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: clic
A lot of dependencies required for each module, I don't know the sequence of
the rpms.Any way to know that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Monday, June 04, 2007 4:51 AM
To: Asterisk Users Mailing List - Non-Commercia
Zoa wrote:
Gordon Henderson wrote:
On Sun, 3 Jun 2007, Andrew Kohlsmith wrote:
On Sunday 03 June 2007 4:30 pm, Alex Crow wrote:
No frills, specs look good, price seems excellent!
http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519
That's a terrible phone. I've tried them. t
Here's a possible bug, or more likely, I'm just missing something.
We have a pots card in one of our asterisk boxes. Its a simple asterisk
setup with one FXO/FXS card and basic static extensions file, etc. When
we dial out over the pots line, 4 out of 5 times, it will work. However,
every 4 or 5 t
Hello.
Is it possible to dial in background 2 or more different numbers while the
same uninterrupted soundfile is playing? Something like this:
exten => Answer
exten => Playback (hello-bla-bla-we are trying to connect you-play-music)
exten => Dial (SIP110/15 and after 15s DIAL SIP111 without int
Asterisk by default uses the codec preferred by other device/client .
Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough
to check if it can avoid transcoding by forcing same codec on other
side of conversation . If both sides prefer g729 then asterisk does
not do transcoding bu
Rob Schall wrote:
Here's a possible bug, or more likely, I'm just missing something.
We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. H
On Monday 04 June 2007 10:28 am, Paul Hayes wrote:
> Looking at the OP's requirements list in the first post, there is
> nothing currently on the market which will cover anything like all those
> features (and do it well!).
I've got the WIP300 and 330 on my list, with the latter being the more lik
It just might be that your carrier is not sending ring . You can use
'r' in asterisk dial command in extensions.conf to generate ring from
asterisk .
On 31/05/07, dima <[EMAIL PROTECTED]> wrote:
Hello, everyone.
Could anyone explain me how does ringback detection works in asterisk.
Sometimes, wh
That just seems really, REALLY dumb for a program of this magnitude.
I know this has been patched here and there by one person or another,
but does anyone know if any of these patches to make CODEC negotiation
actually, you know, negotiate a CODEC will ever make it into the core src?
Jaswind
Alban,
Thanks! Where on earth did you source this? I can't seen to find hide
nor hair of it here in the UK :(
Alex
On Mon, 2007-06-04 at 16:01 +0200, Alban wrote:
> Hi,
> I've tested several wifi phone (UtStarcom, Hitachi 3000 and 5000, and one
> Siemens). The Siemens is the best one, for a rea
On Mon, Jun 04, 2007 at 06:34:37PM +0530, ram wrote:
> Hi
>
> iam keep getting this log in my asterisk log
>
> is this harm anything, and how can stop this, any suggestions
>
>
>
> Jun 4 18:21:47 DEBUG[2093] chan_sip.c: Stopping retransmission on
> '[EMAIL PROTECTED]' of Request 102: Match Fo
But if this was the case, then why would the message playback (from the
provider) read back the digits from the start. I mean, I dialed
630-XXX-XXX and it reads back 312 (my local area code)-630-XXX-X
I would think if it wasn't waiting, then it would do like a 312-30X-XX-XX
Rob
John Novack wrot
PROBLEM...NO DTMF ON OUTBOUND CALLS
1
ASTERISK FORWARDS THE DIGITS
Got rfc2833 RTP packet from 66.108.217.191:2256 (type 101, seq 279, ts
-1975142833, len 4, mark 0, event 0009, end 1, duration 1600)
Sent RTP packet to 209.212.88.161:32872 (type 101, seq 63407, ts 60776, len
4)
Sent R
some work has been done here:
http://bugs.digium.com/view.php?id=4825
but seems to be quite death and probably not directly applicable to
current asterisk src :'(
SIP wrote:
That just seems really, REALLY dumb for a program of this magnitude.
I know this has been patched here and there by on
Don't overthink this
Error messages from providers are frequently misleading and inaccurate
Some only use one recording for anything they think they can't process.
Add at least one "w" to the dial string and see if all the misdials
don't go away.
Also check the list archives for MANY such compla
has anybody made a patch for asterisk 1.4*?
On 6/4/07, Jaswinder Singh <[EMAIL PROTECTED]> wrote:
Asterisk by default uses the codec preferred by other device/client .
Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough
to check if it can avoid transcoding by forcing same cod
Also, in the dial command the "w" says its for the *1 recording. Not
waiting. Is the documentation wrong? What is the correct way to wait in
the dial command?
Rob
John Novack wrote:
>
>
> Rob Schall wrote:
>> Here's a possible bug, or more likely, I'm just missing something.
>>
>> We have a
We have it (in belgium)
http://www.voipsolutions.be/phones/dect-sip-phones/siemens-gigaset-sl75-wlan-voip-phone.html
I still think DECT is better though :)
Zoa
Alex Crow wrote:
Alban,
Thanks! Where on earth did you source this? I can't seen to find hide
nor hair of it here in the UK :(
Alex
Alex,
I bought them at Beronet (shop.beronet.com), a german company.
I'm located in France, no problem for them to send them here...
Alban
Le Lundi 4 Juin 2007 17:29, Alex Crow a écrit :
> Alban,
>
> Thanks! Where on earth did you source this? I can't seen to find hide
> nor hair of it here in the
Not option w
add a "w" in the DIAL STRING
Guess you didn't search the archives?
Sometime in 1.2 this feature was fixed to work with pulse dial as well.
example:
exten => s,1,Dial(ZAP/g4/w(${ARG1:3:4}),360,Tt)
John Novack
Rob Schall wrote:
Also, in the dial command the "w" says its fo
Asterisk does NOT wait for dialtone when going offhook on an FXO port.
Asterisk is sending digits to the telco too quickly.
Add a "w" (.5 second wait) before the extension to be dialed. Example:
exten => _91NXXNXX,n,Dial(${TRUNK}/w${EXTEN:${TRUNKMSD}},,wW)
Rob Schall wrote:
Here's a poss
well,
the best I got is the tc300/arcor/twintel a gsm/wifi from pirelli -
http://www.pirellibroadband.com/en_IT/browser/attachments/pdf/DPL10.pdf
tried many wifi phones, that's the best we got.
Long lifetime for the battery, good reception, roaming between Access
points with the same networ
Where's Steve Murphy when you need him? :-)
This doesn't seem to work in AEL2...
Macro foo(arg1) {
.
Includes {
Hangup;
}
}
The error is: File: /etc/asterisk/extensions.ael, Line 59, Cols: 5-12:
Error: syntax error, unexpected KW_INCLUDES, expecting '
What happens if you connect the fxo to the fxs and try several attempts
at completing a call? This should at least tell you if the issue is
outdialed digit issues or telco receipt issues.
Dave
On Mon, 2007-06-04 at 10:30 -0500, Rob Schall wrote:
> But if this was the case, then why would the mess
the 'make' command would typically recompile and re-link only the files
that have changed. Not sure how well this works with asterisk, but I
think that's the idea.
Mojo
Arpit Mehta wrote:
hi,
This might be the most obvious thing to you. I need to change some parts
of the source code of Ast
You want Asterisk to dial 1-630-XXX-XXX, but the telco only got
630-XXX-, which the provider will see "630-" and assume your
current area code of 312.
Rob Schall wrote:
But if this was the case, then why would the message playback (from the
provider) read back the digits from the sta
Humberto Figuera schrieb:
> HI Tobias,
>
> look in www.soft-switch.org/unicall/unicall/index.html ;p
>
Thank you. Not very complete but it has given me an idea what to think
of unicall.
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aster
Hi,
I'm having complaints from some users about calls into dynamic meetme
sessions failing. I'm guessing that they are dialling the wrong DTMF
keys, OR that DTMF is hearing the digits entered wrong (or not hearing
some).
I've put debug => debug into logging.conf, and searched through the
file,
Hi
I have configured attended transfer in features.conf like this
[general]
parkext => 70 ; What ext. to dial to park
parkpos => 00-99; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
p
Can anyone enlighten me as to why it takes 40 minutes or more for a
posting to the list to appear. This seems excessive, as other forums do
not take this long.
Dave
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asterisk-users mailing l
OPTIONS to the Dial line (at the end) are different the special digits
in the number string.
Rob Schall wrote:
Also, in the dial command the "w" says its for the *1 recording. Not
waiting. Is the documentation wrong? What is the correct way to wait in
the dial command?
Rob
John Novack w
John Novack wrote:
Not option w
add a "w" in the DIAL STRING
Guess you didn't search the archives?
Sometime in 1.2 this feature was fixed to work with pulse dial as well.
If he had searched the archives, he never would have posted the message
in the first place.
_
I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2
... error appears >
>From where I can get the missing rpms .or kernel source
>From where I can get the centos 4.4 server kernel source.
Regards
*
No employee or a
We have the latest version of asterisk, on a xeon dell server (2gb ram),
with 6 snom320's(latest firmware) and 3 grandstream gxp 2000's (latest
stable firmware) and are having a few problems. We have a basic menu that
transfers calls to different extensions. The problems can be found on all
exte
Having scoured the web, I still am no better off ..
I have 2 Aastra480i's , and 120+ cisco 7940's :) . I am trying to decide
which model to use going forward when we purchase more kit. They both
seem much on a par regarding features.
Q1: Is there anyway of making the cisco auto-answer _withou
Hi everyone,
in ldap realtime sip peers i need "fullcontact" set to
"sip:[EMAIL PROTECTED]" for asterisk to correctly match the peers (at least
for the natted peers to reach them)...
anyway, how do I populate fullcontact "on the fly" with information from
exten and userip?
of course, i could
the config file is basically the sample file
gdb of the core files show the below.
Loaded symbols for /usr/lib/asterisk/modules/chan_ooh323.so
#0 0xb7d203e7 in strcasecmp () from /lib/libc.so.6
Any suggestions?
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Well, the claim of 1 members MIGHT have something to do with it!
E-mail delivery is notoriously erratic as well.
If you think this is slow, try some on Yahoogroups!
John Novack
David Boyd wrote:
Can anyone enlighten me as to why it takes 40 minutes or more for a
posting to the list to ap
On 6/4/07, David Boyd <[EMAIL PROTECTED]> wrote:
Can anyone enlighten me as to why it takes 40 minutes or more for a
posting to the list to appear. This seems excessive, as other forums do
not take this long.
Dave - my postings consistently show up nearly immediately. Perhaps
your SMTP server(
Hi all -
Not really sure where to post this question as I am just starting to
research this issue.
We want to allow users to dial into our did voip number. Our service will:
1. get their phone number via caller ID. look up data with the caller id.
2. generate a wave file based on the data
On 6/4/07, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
Q2: Is there any real difference between the 480i and 9112 / 9113 phones
apart from number of lines and display size ?
I have no experience with the Cisco's, so I can't answer those
questions. However, I have deployed quite a few Aastras
I have some Analog card on a PCI slot of a remote computer, Is their a way I
can figure out remotely the name of the card.
I have FC6 installed on the machine.
Regards,
Sanjay Rajdev
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asterisk-
I finally got a chance to investigate this further. The fundamental
problem seemed to be that I was using a context name of "iax-trunk".
When I changed this to "intrunk", it worked. What are the rules for
context names? Was it the length or the special character that caused me
problems?
Thanks
Hi,
Is it possible to get the remote channelname that will be bridged when
the call is answered, only having the channel that is in the Ring(ing)
state? As far as I can see no variable seems to fit when doing the show
channel command.
I want to be able to redirect/manipulate an incoming call
modprobe the default analog drivers. then dmesg.
-bk
- Original Message -
From: "Sanjay Rajdev" <[EMAIL PROTECTED]>
To: "asterisk-users"
Sent: Monday, June 4, 2007 12:29:37 PM (GMT-0800) America/Tijuana
Subject: [asterisk-users] Detecting card on the PCI Slot
I have some Analog card on a
On Tue, 5 Jun 2007, Sanjay Rajdev wrote:
I have some Analog card on a PCI slot of a remote computer, Is their a way I
can figure out remotely the name of the card.
I have FC6 installed on the machine.
Try the 'lspci' command.
Eg:
:00:14.0 Communication controller: Tiger Jet Network Inc.
On Tue, Jun 05, 2007 at 12:59:37AM +0530, Sanjay Rajdev wrote:
> I have some Analog card on a PCI slot of a remote computer, Is their a way I
> can figure out remotely the name of the card.
> I have FC6 installed on the machine.
lspci
What driver handles it?
--
Tzafrir Cohen
Marcus Carlson wrote:
Is it possible to get the remote channelname that will be bridged when
the call is answered, only having the channel that is in the Ring(ing)
state? As far as I can see no variable seems to fit when doing the show
channel command.
I want to be able to redirect/manipulat
The 9112 also doesn't have a ethernet bridge in it. The cost of adding
a switch to a local office puts this unit nearly priced as a 9133 which
I deploy mostly.
As for the 480...I had that..upgraded to the 57i (CT now - the wireless
handset supporting one). Love it. Button feel a little different
There is a yum command to install from cache and not look online. Details
are in "man yum".Never tried what you are doing, so i can't say for sure
it would work. But you probably have to pass yum the makecache command on
the first machine and install from cache on the second. But this is j
Thanks Eric.
That is exactly what the problem was. I actually added a D(1) instead of
the w, but either would work I'm sure. Its odd that you couldn't get a
better error than that. I mean, asterisk should be able to tell what was
actually sent and received by the other end. I shouldn't say it
"sho
I have installed FC6 on it, want to configure it with Asterisk. It had some
driver earlier but the machine has been formatted yesterday, so no idea.
Also I am new to Linux.
Regards,
Sanjay Rajdev
Phone : +1 (877) 342 2329 x 1702
Fax : +1 (815) 261 5907
http://www.featherstoneinformatics.com
Co
Thanks for the suggestion, I figured out the cards.
I have 2 Digium TDM400P card and a Sangoma A101 single port card on the machine.
Any suggestion on installing them.
Regards,
Sanjay Rajdev
- Original Message -
From: "Gordon Henderson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing Li
This notifies you that it has been used (IIRC).
Hi
what does that mean , it has been IIRC ?
ram
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On 04/06/07, Caio Zanolla <[EMAIL PROTECTED]> wrote:
Hi everyone,
in ldap realtime sip peers i need "fullcontact" set to
"sip:[EMAIL PROTECTED]" for asterisk to correctly match the peers (at least
for the natted peers to reach them)...
anyway, how do I populate fullcontact "on the fly" with inf
means (I)f (I) (R)emember (C)orrectly
On 6/4/07, ram <[EMAIL PROTECTED]> wrote:
> This notifies you that it has been used (IIRC).
Hi
what does that mean , it has been IIRC ?
ram
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asteri
Dear Sir,
Thank you very much
ASLAY
- Original Message -
From: "Nasir Iqbal" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, June 03, 2007 5:43 PM
Subject: Re: [asterisk-users] Auto Dial Problem
> Hi,
>
> > I setup auto dial on my ast
that becasue the reinvite is using a private ip probably..
sip debug
pastebin the results..
look in the re-invite part..
On 6/4/07, Compnet Bobby <[EMAIL PROTECTED]> wrote:
We have the latest version of asterisk, on a xeon dell server (2gb ram),
with 6 snom320's(latest firmware) and 3 gr
I have two Asterisk servers. One is my primary server that I link to all of
my providers and the other is at an office building with multiple tenants.
If I tell Asterisk to dial an entry in the iax.conf that is for one customer
off that second box, why does it use a different account for a dif
On 6/4/07, Jordan Novak <[EMAIL PROTECTED]> wrote:
I am having a difficult time with the transition from agentcallback login...
Here are a few of the isssues, I am logging in using chan_ local
ie:local/8000 as the extension
I'm not sure if this will solve any of your problems or not, but I've
f
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