On Thu, 9 Sep 2004, hank smith wrote:
> I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that
> what I put in the xml file?
Go read: http://www.colinux.org/wiki/index.php/coLinuxNetworking
Specifically:
"If in doubt, the name of the card can be found in colinux-daemon st
On Thu, 9 Sep 2004, Karl Brose wrote:
> In order to dial out to a sip provider, you need to configure that
> provider in your sip.conf file as a peer with your proper username
> and secret, etc.
Cool! Just found that in the handbook too a second or two ago :-)
Thanks for taking
[EMAIL PROTECTED] wrote:
> Problem was with asterisk.. Mark had made a change in chan_sip.c
> that affected noncodec capabilities, it's been fixed.
Do you have a bug number? Or something else to find it in the bug database?
--
Andreas SikkemaRits tele.com
Scheepmakersstraat 11
traceroute A -> B:
traceroute to 192.168.2.44 (192.168.2.44), 30 hops max, 38 byte packets
1 192.168.11.1 (192.168.11.1) 1.964 ms 1.181 ms 0.852 ms
2 10.138.3.2 (10.138.3.2) 43.428 ms 49.634 ms 47.601 ms
3 192.168.2.44 (192.168.2.44) 53.440 ms 49.320 ms 48.968 ms
traceroute B -> A:
trac
Greetings All,
I have a new post on the blog. It goes a little bit more in depth on
wcfxo.c and touches on zaptel.c. Two more screen shots. Loads of fun.
Take a look: http://zapteldoc.blogspot.com
Regards,
Victor
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All,
I am very new to pbx hardware and equipment and any help will be greatly
appreciated. I am now the proud owner of a TDM422p and Iaxy/S100I. The server is
running debian testing so I first installed the asterisk deb package. To get the
zap modules, I compiled zaptel-1.0-RC2. After some con
> Something the user list in
> Microsoft Messenger. I was thinking on some sort of web page that can
> check the registration of the sip clients on the asterisk but want to
> know if already exist to avoid to reinvent the wheel.
That is actually quite easy and there are some projects that achive t
is there going to be a gui for co linux and astwind?
I will have to see if either there is going to be a gui or if yasr a screen
reader for the blind will work with this thing.
thanks
hank
- Original Message -
From: "Greg Boehnlein" <[EMAIL PROTECTED]>
To: "arsal siddiqui" <[EMAIL PROTECT
I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that
what I put in the xml file?
- Original Message -
From: "Greg Boehnlein" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Thursday, September 09, 2004 9:
On Thu, 9 Sep 2004, Matt - Telcom Products wrote:
> Hello,
> Does anyone know how to conference a call on the SNOM 200 phone?
>Whenever I push the cnf/tran button it just hangs up on the active call.
>The manual says you have to push the cnf function key but it doesn't
>appear in the lcd o
On Thu, 9 Sep 2004, Martin Mielke wrote:
> Hi all,
>
> due to the rather big email traffic regarding this issue, I decided to
> publish the script so people can download it at their own risk... :-)
>
> Please, visit:
>
> http://www.leals.com/~mm/asterisk
>
> for further information.
>
>
On Thu, 9 Sep 2004, arsal siddiqui wrote:
> dear khurram,
>
> i need to know the price for x100p. i've emailed convergence.com.pk
> and never get a reply. If you could help me in this regards, i'll be
> greatful. I need to know the price.
>
> send me an email off the list. if you can help me in
On Thu, 9 Sep 2004, Chris HARIGA wrote:
[SNIP]
>
>
>
Yep.. Don't you hate the network driver names that these things pick?
Couldn't they just call it "Intel PRO 100" or something simple like "Pro
100"
CoLinux 0.6.2 will allow you to use the name that YOU assing to the
connection i
The Polycom VOIP phones(IP500 and IP600) work wonderfully in our conference
rooms we have an IP500 in one and an IP600 in another and they have great
audio quality. They also work great with Asterisk.
MATT---
-Original Message-
From: Joe Dennick [mailto:[EMAIL PROTECTED]
Sent: Thursday, S
On Wed, 8 Sep 2004, Chris HARIGA wrote:
> I make it work!!
>
> My Astwind is up and running!
> Now is 11:53 PM and I'm going to bed. Tomorrow morning I will post how I fix
> the Ethernet connection.
I bet you followed the following directions! ;)
From: http://www.colinux.org/wiki/index.php/
On 9 Sep 2004, khurram bhatti wrote:
> Well I wanted to test astwind and consulted * person
> he gave me this comment
> "lord help us all ... why would you want to simulate a linux system on
> top of a windows system in the first place?"
It's not a "simulated linux system". CoLinux is a kernel t
On Wed, 8 Sep 2004, hank smith wrote:
> hello I am fitteling with the astwind-installer-0.1.1.exe asterisk for
> windows and am having trouble getting the thing to connect to the meers
> to download the updates and stuff. I looked at the wiki and set up
> networking and stuff with no success,
when you get this up up can you give the phone number?
this sounds rather interesting, and fun!!!
- Original Message -
From: "bagattin jerome" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, September 09, 2004 7:54 PM
Subject: [Asterisk-Users] Store data from call to database
I have a Toshiba Strata with 200 extensions and I already have an
approved budget to move to *, but I'd really like to do this in
stages, and it would be nice to use the existing extensions and save
the $20k (or at least delay the $20k expense).
Can you explain more about how you're using the E&M
Sounds like it be best as a custom app or AGI depending how many calls
you will be taking and how bad the performance hit of using an AGI vs
Compiled app is for your needs
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Hi,
The song with "linux pakage in one week" I get it 5 weeks ago, 3 weeks ago
and this week :(
Best regards,
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josh Roberson
Sent: Thursday, September 09, 2004 7:48 PM
To: Asterisk Users Maili
On 10/09/2004 07:07 Leo Ann Boon said the following:
The 3502 does register with 2 accounts.
not the 3504A in Proxy Mode. i think i've nailed it down to the fact
that the 3504A (firmware 107a) uses the same SIP Call-ID but changes the
tag= parameter in the From header when it responds to the 407
Hi,
I use asterisk for a phone quiz game.
I need to store data in a database (MySql, postgres) :
telephone number, name (voice), ... and of course the
answers at the quetions.
What's the best way to store my data ?
- script with system() command ?
- AGI script
- CDR
- others ...
Thanks
Je
Okay. I read that it should be wcfxs in a list somewhere, makes sense
that it's the card not the module though. :)
If I load it under a 2.4 kernel the pci comes out as
-
Bus 0, device 8, function 0:
Network controller: Tiger Jet Network Inc. Intel 537 (rev 0).
Just to be clear, wcfxo is for the X10xP cards. And wcfxs is for the TDM
card irregardless of whether it has FXO or FXS modules.
However I still think you have a problem as you don't have any card
identified as communication controller: Tiger Jet networks in your
/dev/proc. You should have that
On 09/09/2004 21:16 Evert Meulie said the following:
But... when I boot it, I get:
Registered to '192.168.11.6', who sees us as 10.138.3.2:4569
Why doesn't server A see server B as 192.168.2.44??
there's most likely a box between A and B which is doing NAT, and it's
most likely 10.138.3.2. however
If it were me; I'd opt for one of the Polycom Conference phones (they
are just regular analog phones), and use an FXS card to connect it to
Asterisk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown
Sent: Thursday, September 09, 2004 4:13 PM
To:
Hi Lyle,
I don't have lspci on my system. It's a dump of what is in the
/proc/pci anyway.
Yep, zaptel loads fine. The X100P loads and is working well, with
Asterisk. It's actually in another machine, I moved this TDM400p to a
machine that I can play around on.
I haven't loaded wcfxo as that's
Mine was doing exactly the same thing until yesterday. I deleted my agi
directory and did a cvs update again. I then recompiled and reinstalled
asterisk.
Darren Wiebe
[EMAIL PROTECTED]
Doug Harris wrote:
Hi,
I posted to this list couple of days ago, that my astcc is not writing
the card bal
What does lspci -v show? I just looked at my /proc/dev and it shows two
Communication Controller: Tiger Jet Network Inc in there. I have a TDM22b
and a X100P on a 2.4.x kernel.
Did you modprobe zaptel first? Then wcfxs and then wcfxo?
Lyle
- Original Message -
From: "Colin Haxton" <[
> > Did I see something on here about using an AGI script to do reverse
> > lookups via anywho.com? I have a PRI that only gets caller-id number and
> > no Alpha.
[...]
I put a copy of it here...
http://www.voiping.com/calleridnamelookup.agi
It was written by James Golovich <[EMAIL PROTEC
Hi all,
I have just purchased the DevKit from Digium and received a X100P and a
TDM400P (it has one FXS module).
The problem is that I can't get the kernel module (wcfxs) to load and
run. I have searched the archives and can't find anything about this.
Do the messages below ring any bells wit
> Not necessarily so. Recently I discovered that Artisoft's Televantage
> Soft PBX can support Toshiba Strata CS digital phones (DKT 2000 and
> 3000) through a PCI 16-port digital station card (Toshiba part
> #CS-DKTU-TV). Apparently, the Strata CS is an OEM licensed version of
> Televantage. It w
Jonathan Moore wrote:
Unfortunately we missed on big problem with Call waiting in our testing. When
using asterisk 0.9.1 or rc2 the phone will reboot when you hit the flash button
to roll over to the second call.
We were experiencing random uip200 rebooting after upgrading the phone
firmware. It
On Fri, 2004-09-10 at 04:50, [EMAIL PROTECTED] wrote:
> QSIG passes callername and other variables by a mechanism that asterisk
> cannot interpret at the moment. It sends them either in a information
> element in the setup message for the call or in an additional facility
> message after the fact
dear khurram,
i need to know the price for x100p. i've emailed convergence.com.pk
and never get a reply. If you could help me in this regards, i'll be
greatful. I need to know the price.
send me an email off the list. if you can help me in getting * hardware.
Waiting for your reply
Arsal
On 0
I wrote cepstral regarding this at the beginning of the week, thought it
might be relevant to post the reply:
"Thanks for contacting us. Our Linux package is off the site right now
because we are releasing a new version, 3.02, next week. This is an
incremental release. The major update of this ver
> Another question, along the same kind of lines, has anyone figured out
> how to keep the SoundPoint IP 600 receiver on-hook? Mine keeps being
> pushed up by the little piece of plastic that is supposed to detect if
> it's on-hook. It looks like the handset already has the little hole
> for the
Danny Zak wrote:
Hello Ariel's,
i got this back from welltech
For the 38 unit
It couldn't support only one account for the registeration as so far...
The 3502 does register with 2 accounts.
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[EMAIL PROTECTED]
ht
Michael Little wrote:
As far as I know, you will not be able to keep the digital phones. They
are PBX specific. I am faced with the same situation. Unfortunately, I
cannot get the funds to replace all the phones, so I am stuck with using
the Strata PBX and replacing the Stratagy voicemail with A
Problem was with asterisk.. Mark had made a change in chan_sip.c that
affected noncodec capabilities, it's been fixed.
Thanks,
Billy
- Original Message -
From: "Tenorio, Leandro" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent:
I got a AS5350, and change the config to reflect your config as follows
AS5350
dial-peer voice 50101 voip
preference 1
destination-pattern xxx
voice-class codec 1
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx
dtmf-relay rtp-nte
Etc
And in *
sip.conf
[context]
type
Quoting Jerry Geis <[EMAIL PROTECTED]>:
> Cepstral offers Linux versions.
> Just contact them.
>
> http://www.cepstral.com/cgi-bin/downloads?page=voices
Note that you can not download any Linux versions from that page.
They changed something a while back. Released a new TTS engine for Windows
I just purchased 30 of these after testing one for a few months and would like
to quickly purchase another 40.
We really like these phones: good sound quality, good echo control (no echo in
speaker phone), power over ethernet support, 10/100 switch, 8 programmable keys.
Unfortunately we missed on
What I did with Gentoo, which may work similarly with Debian, was first
I emerge'd the Asterisk 0.9.0 version that Gentoo offered. The benefits
of this was it downloaded and installed all the dependencies and then
installed Asterisk. I then used CVS to download the latest (with all the
bug fixe
Bill,
There was an FWD provisioning problem a week or so ago
where after signing up for FWDIAX, you were able to register using IAX,
but calls were not forwarded from SIP to IAX.
This problem was fixed, but I'm not sure if the affected accounts
were repaired. As you found out, simply re-activ
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Mike Roberts
> Sent: Thursday, September 09, 2004 2:30 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Debian Sarge -- cvs vs. apt-get
>
> After reading the wiki, I am still confu
I've also had problems receiving inbound IAX calls from FWD. And without
any outbound problems. I've three numbers and fortunately one worked while
two did not. The one working number encouraged me to keep trying to work
out a reason for the difference.
In the end (today) I re-activated IAX for
I am using CVS-HEAD-08/29/04-22:41:39
I have notransfer=yes in my iax.conf
I have been on the phone most of the day...dropped twice so far.
Paul Seniuk
-Original Message-
From: Kris.Boutilier [mailto:[EMAIL PROTECTED]
Sent: September 9, 2004 2:43 PM
To: asterisk-users
Subject: RE:
After reading the wiki, I am still confused as to whether I should install
zaptel, libpri, and asterisk from cvs or use apt-get to retrieve the Debian
packages. I am running Debian sarge with kernel 2.4.26. What is the
prevailing opinion out there about using cvs vs. apt-get for the various
compone
>>
>> Are the FXOs on the 2x on ports 1-2 or 3-4? Maybe it has to
>> do with *any* FXO on port 1...
>>
>> Please get back with the list with your findings.
>>
>
>
>My experience led to a replacement from Digium, but the card is a
>TDM400P with 4 FXO...now that I think of it, during troubleshoot
what are these error messages: Unknown IE 40 (cs6, Unknown Information
Element)
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> Works for me, follow the instructions closer. :)
>
> Storm D. J. Petersen wrote:
>
> >Hi,
> >I cannot seem to accept incoming calls from FWD using IAX2.
> I followed the
> >directions posted at www.fwd.pulver.com/advanced/iax I can
> make outgoing
> >calls fine using IAX via FWD. When s
Any advice on a good conference phone that works with
Asterisk? I like the Cisco line and was wondering if anyone has used the 7935
or 7936 phones. From what I can tell they don’t have a sip load. Has
anyone verified this or gotten an ETA from Cisco?
Chad
___
What are ZOMBIES and why am I getting them?
fs-1*CLI> show channels
Channel (ContextExtensionPri ) State Appl. Data
IAX2/[EMAIL PROTECTED]/2 (exten-access 2997 1 ) Up ZapScan
(Empty)
SIP/2921-700d (toll-access 1 ) Up Bridged
ï
do you got a number I can call to take a
listen?
- Original Message -
From:
Steve Murphy
To: [EMAIL PROTECTED]
Sent: Thursday, September 09, 2004 12:06
PM
Subject: [Asterisk-Users] Festival Speech
Synthesis 1.95:beta July 2004 Eval
Hello--In the interests
The Store only lists Windows and Windows CE.
They are releasing version 3Sept 12.. I called them.
Eric Wieling wrote:
On Thu, 2004-09-09 at 14:59, TELUX wrote:
How do you get Cepstral working, they only offer windows versions. do I
have to complie it to linux?
http://www.cepstral.com
On Thu, 09 Sep 2004 15:49:48 -0500, Eric Wieling <[EMAIL PROTECTED]> wrote:
> On Thu, 2004-09-09 at 14:59, TELUX wrote:
> > How do you get Cepstral working, they only offer windows versions. do I
> > have to complie it to linux?
> > http://www.cepstral.com
>
> They have a linux version for purchas
On Thu, 2004-09-09 at 14:59, TELUX wrote:
> How do you get Cepstral working, they only offer windows versions. do I
> have to complie it to linux?
> http://www.cepstral.com
They have a linux version for purchase on their web site.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> Sent: September 9, 2004 1:07 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] IAX2 dropping call?
>
>
>
> Hello all,
>
> I updated from CVS 3 days ago and now my IAX2 gateway is dropping
> calls without wa
Works for me, follow the instructions closer. :)
Storm D. J. Petersen wrote:
Hi,
I cannot seem to accept incoming calls from FWD using IAX2. I followed the
directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing
calls fine using IAX via FWD. When someone calls me from FWD I ge
Seth Mattinen wrote:
Does anyone know how (if possible) to do three way calling on the
UIP-200?
The UIP-200 currently doesn't support this, which is a shame.
I typically create a meetme room for every sip extension (ie: 8XXX where
XXX is the exten number). Users can then transfer callers to 8XX
Cepstral offers Linux versions.
Just contact them.
http://www.cepstral.com/cgi-bin/downloads?page=voices
Jerry
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Hi All,
I am setting up two boxes with asterisk. Box A has a T100P.
It is working I can put calls in the outgoing spool directory
and call extensions or outside numbers and do a playback of
demo-congrats (not sexy but good for an example).
Now I have a second Box b. Box B has no hardware just
runni
Hi,
I posted to this
list couple of days ago, that my astcc is not writing the card balance to the
mysql database.
http://lists.digium.com/pipermail/asterisk-users/2004-September/061645.html
I just want to ask
this question one more time before creating a bug note in "mantis". Since th
On Thu, 09 Sep 2004 13:59:20 -0600, TELUX <[EMAIL PROTECTED]> wrote:
> How do you get Cepstral working, they only offer windows versions. do I
> have to complie it to linux?
> http://www.cepstral.com
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
It sounds like Skinny couldn't load- doesn't sound fatal.
On Thursday 09 September 2004 08:01 pm, [EMAIL PROTECTED] wrote:
> Greetings,
>
> I've just installed Asterisk RC2 on a new RedHat 9 box (fully patched).
>
> The build and install went fine, but after starting Asterisk, I get the
> followin
Hello all,
I updated from CVS 3 days ago and now my IAX2 gateway is dropping
calls without warning.
It happens right in the middle of a conversation with no pattern. I
never had this
Problem before and am usually talking 2-3 hours a day.
Is their a bug? Should I rollback?
Cheers,
Paul Seniu
Greetings,
I've just installed Asterisk RC2 on a new RedHat 9 box (fully patched).
The build and install went fine, but after starting Asterisk, I get the
following messages in /var/log/asterisk/messages. Restarting Asterisk
produces the same errors:
-
Sep
How do you get Cepstral working, they only offer windows versions. do I
have to complie it to linux?
http://www.cepstral.com
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Is anyone familiar with the Trollphone's LCR package?
There is a field in the egress table labeled substitue. Placing a $1 there
results in the correct dial extension being passed. However how is this
field used to substitute replacement dial extensions... in other words as
an example, lets sa
Hank,
IP500's are 199$ USD at voipsupply.com
and
Cisco 7940 are 295.99$ USD at voipsupply.com
Thanks,
Matt
hank smith wrote:
what is the price range in us dollars?
- Original Message - From: "Jody N. Rudolph" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
A local vendor here carries IP500s for sub $200. Right now they are out
of stock, but he has more coming in. If you want his contact info msg me
off list.
-Tim
-Original Message-
From: Ty Purcell [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 09, 2004 2:45 PM
To: Asterisk Users Mail
In July I bought one from CDW for $280.75.
Ty
-Original Message-
From: hank smith [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 09, 2004 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940
what is the price
Ah... Looks like I have the 500's...Sorry.
Ty Purcell
-Original Message-
From: Derek Listmail Acct [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 09, 2004 12:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycon IP 300 SIP vs Grandst
Does anyone know how (if possible) to do three way calling on the
UIP-200? There doesn't seem to be much info about this phone, but all
the feature lists I've read says it can do conference calls. I can't
seem to do it, though. Any help would be appreciated.
--
Seth "et lux in tenebris lucet" M
Hi,
The big problem was with Ethernet name on bridge. U can find the real name
of your Ethernet on Cooperative Linux Console. Take a look @
http://www.techselesta.com/astwind.jpg and U will see a printscreen of my
error.
I have a Intel(R) PRO/100+ Alert on LAN* Management Adapter on my box.
This
Huddleston, Robert wrote:
Okay - read it... my configuration works... what I want
exten => XX,1,Wait,2
exten => XX,2,Dial(OH323/XX)
I want it to pass the 10 digits to the DIAL string... I'm not sure I
understand the macros
can I just put the ${EXTEN} in there??
Of course. Th
On Sep 9, 2004, at 8:53 AM, Marcello Lupo wrote:
we have a community of people on an * box that use SIP softphones to
talk each
other. Can you suggest me the quickest and simple way to let someone
know who
is online without have to call one by one the persons to look if they
are
present or not??
Hello--
In the interests of playing around and wasting time, I've installed the latest version of the
Festival stuff, 1.95beta.
And, in the interests of future Asterisk-Festival connectivity, I applied the 1.4.3 patch to put in the
asterisk related routines. I did it by hand, but, it looks li
*** Astricon 2004: Over 250 Asterisk professionals!
Astricon, the first Asterisk user's and developer's conference
is a success and we now have over 250 people registred.
Thank you for all your support of this event and please have
patience with us as we're trying to handle all details
with payment
what is the price range in us dollars?
- Original Message -
From: "Jody N. Rudolph" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Thursday, September 09, 2004 11:32 AM
Subject: RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940
- Original Message -
>
> Hey all,
>
> Did I see something on here about using an AGI script to do reverse
> lookups via anywho.com? I have a PRI that only gets caller-id number and
> no Alpha.
I have thought about this as well. Should be totally possible but first off
you could just cha
Okay - read it... my configuration works... what I want
exten => XX,1,Wait,2
exten => XX,2,Dial(OH323/XX)
I want it to pass the 10 digits to the DIAL string... I'm not sure I
understand the macros
can I just put the ${EXTEN} in there??
-Original Message-
From: Mic
-I../asterisk seems to be the key
cd /usr/src
cvs checkout asterisk asterisk-addons
cd asterisk-addons
make
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo
> Sent: Thursday, September 09, 2004 12:05 PM
> To:
can you post the information on how you got that thing working?
thanks
hank
- Original Message -
From: "Chris HARIGA" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Wednesday, September 08, 2004 8:55 PM
Subject: RE: [Asterisk-
Huddleston, Robert wrote:
Due to the format of the message coming from the H323 channels included w/
Asterisk we were unable to use our gatekeeper.
For a quick solution we tried the OH323 channel drivers and can receive
inbound calls from the parent gatekeeper.
We are trying to do a dial to gatekee
The Polycom IP500s do support customized ringtones and can use a customized
ALERT_INFO for all of them. One thing that is worth noting in this
comparison is that the IP500 doesn't support the XHTML microbrowser that the
IP600 does. Since they both use the same SIP application I am hoping they
enabl
Hello,
Does anyone know how to
conference a call on the SNOM 200 phone? Whenever I push the cnf/tran button it
just hangs up on the active call. The manual says you have to push the cnf
function key but it doesn't appear in the lcd on my phone.
Thanks
-Matt
__
Hey all,
Did I see something on here about using an AGI script to do reverse
lookups via anywho.com? I have a PRI that only gets caller-id number and
no Alpha.
TIA,
--
Daniel Jimenez
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digiu
Due to the format of the message coming from the H323 channels included w/
Asterisk we were unable to use our gatekeeper.
For a quick solution we tried the OH323 channel drivers and can receive
inbound calls from the parent gatekeeper.
We are trying to do a dial to gatekeeper...
I am trying
exten
On Sep 9, 2004, at 9:53 AM, Matt G wrote:
I've been asked to determine which phones our organization should go
with. And I've narrowed it down to the Polycom IP500 or the Cisco
7940.
From my travels through google, it's hard to find a definitive
comparison of the two phones. So I thought I woul
>
> Are the FXOs on the 2x on ports 1-2 or 3-4? Maybe it has to
> do with *any* FXO on port 1...
>
> Please get back with the list with your findings.
>
My experience led to a replacement from Digium, but the card is a
TDM400P with 4 FXO...now that I think of it, during troubleshooting
th
Does anyone know how to do this with the OH323 channel driver?
I want the local (7 digit dialing) to go out an h323 that I have registered
to a gatekeeper...
can I do something like
exten => _7.,2,Dial(OH323/ipofgatekeeper)
-Original Message-
From: Begumisa Gerald M [mailto:[EMAIL PROT
On Thu, 09 Sep 2004 12:52:56 -0400, John Kington <[EMAIL PROTECTED]> wrote:
> What about sip softphones that use STUN? I am especially interested in UK
> because my daughter is going to study in London.
If she is going to be on a residential ADSL, that shouldn't be a
problem. I have friends in the
> I plugged it in, configured it, and it works great. I really like the
> polycom phones. They have a superb speakerphone.
> (you can hear quiet whispers and people tapping pens on the desk.)
Just a note here... the IP300 doesn't have a mic on the speakerphone, it's
listen only.
_
Sorry for the OT, but I was simply wanting some rough guidelines from
someone who has bought one.
I resent having to fill out a form (every damn polycom "partner" I've found
in the uk will not publish a price) just to find out how much. If I have to
ask, is it too expensive ;)
And, from where
seems that asterisk isn't installed
Il gio, 2004-09-09 alle 18:48, Michael Workman ha scritto:
> Well this is what I am getting
>
>
>
> [EMAIL PROTECTED] asterisk-addons]$ make
> ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c`
> cdr_addon_mysql.c:17:29: asterisk/confi
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kenneth Shaw
> Sent: Thursday, September 09, 2004 12:56 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Legacy Toshiba Phones
>
> I found some postings from Google (notably from
I found some postings from Google (notably from Mark Spencer) about
successful integration of a legacy Toshiba Strata system and Asterisk.
I am also facing that current dilemma. The general legacy solutions that
I can come up with is very easy -- either making Asterisk a "proxy" (or
frontdoor) to
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