If you use open-source software, you have to accept that sometimes
project need some times to be stable and have all features.
OH323 works - even if there are still a few bugs - and the people around
the project are working hard to make to work even better.
If you want something that work now,
Carlos Chavez wrote:
I have 2 Asterisk servers that communicate with IAX2 between them and
support multiple SIP clients each. Only one of them has Zap channels to the
PSTN. I've been having problems because the Zap channels do not hang up when
a sip client of the external server makes a call
On Tue, 22 Mar 2005, McQuiggan, Mark xt46480 wrote:
> I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an
> Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover
> cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of
> the office calls g
Hi All * lover.
This is not a question only this is a request to all SIP and Asterisk user .
I am also with asterisk last few month and providing callingcard soluation.
most of the SIP or IAX provider asking very high price which is really tough
to resell in market. but still there is some h323 p
I spent the better part of the day trying to figure out why my SIP
calls going through * were just going dead after 20 seconds. I was
sure it was a nat issue but now I'm not so sure anymore.
I have * on a public ip and clients behind a nat. I was using
simpletelecom to terminate my calls. I cou
Finally i installed the asterisk home , meetme2 is working perfectly in it.
Thanks a lot
anil
On Tue, 22 Mar 2005 20:34:41 +1100, PHP Mechanic
<[EMAIL PROTECTED]> wrote:
> ï
>
> User=guest, password=restricted.
> This account wil be open util friday.
>
> Nope:
> 220 Welcome to the Vink Cons
Release 0.68 of IPSwitchBoard is ready for download at:
http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA
Release Notes:
Import/Export extensions to "cidname" in Asterisk Server, integrate this in
your dial plan to see who's calling by using the Asterisk app.
"LookUpCidName()."
S
Yup that works on our end as well We assign 3 of the same lines to
the same phone and it works perfectly.
-
Dan Levine
CYTEXONE | Your Technology Specialists
t: 877.CYTEXONE x 810
l: 212.477.0990 x 810
e: [EMAIL PROTECTED]
http://www.cytexone.com
-Original Message-
From: [EMAI
[EMAIL PROTECTED] has this for it's incoming fax macro
--- start snip ---
[ext-fax]
exten => in_fax,1,GotoIf($[${FAX_RX} = system]?2:analog_fax,1)
exten => in_fax,2,Macro(faxreceive)
exten => in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf
${FAXFILE}.pdf)
exten => in_fax,4,syst
I get the results you want by assigning the same extension to multiple
lines on the phone.
Ben
Friend, George E. wrote:
I'm pretty new at this stuff, but I believe you will need to configure
two different extensions and then roll from one to the other. Without
that, it behaves like call-waiting
I'm pretty new at this stuff, but I believe you will need to configure
two different extensions and then roll from one to the other. Without
that, it behaves like call-waiting on one line.
George
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Phil
We've just started testing with Asterisk (CVS HEAD) and a pair of Cisco
7940G phones running the SIP 6.3 firmware.
One issue that we've run into is the ability to have multiple calls ring to
the phone. Our scenario is that the user is using an extension and another
call comes in for that extensi
Thanks.
In fact SIP <->SIP is fine as well as SIP<->FXS. Although I am not sure
what Codec Polycom is using I will try and force the Polycom to use AWAL.
Thanks for the advice. My machine is a Dual P4 as well with 1gig of memory.
-Scott
- Original Message -
From: "Adam Goryachev" <[EM
Just a guess, but you are using Postgres? When I started
working on/with the MeetMe2 gui I saw the same problem, found
in the archives that others were seeing it and that using MySQL
just worked.
I tested with Postgres and confirmed that sql updates were not
being written back to the database, bu
On Wed, 23 Mar 2005 16:12:42 +1100, Maron Kristófersson
<[EMAIL PROTECTED]> wrote:
> I transformed 12 phones the other day from SCCP 3.something to SIP. Had
> to upgrade to SIP 5 first and then SIP 7.
You didn't have to install 6.x somewhere inbetween? I have a 7940
thats got 5.3 but refuses to
Hi.
It matters hugely which version of firmware you're running on the phone.
The Cisco pages don't help you very much with this though.
I transformed 12 phones the other day from SCCP 3.something to SIP. Had
to upgrade to SIP 5 first and then SIP 7.
Well worth it, these are highquality SIP ph
How to find out why DTMF is not working???
I was running Gentoo versions:
*-0.0.9
*-1.0.3
*-1.0.5 and DTMF on my SPA-3000 was working normally.
I downgraded to CVS-stable and my DTMF is not working.
I'm using dtmfmode=inband in sip.conf and SPA=3000 is set to Auto.
Is there a way to troubles
HI
I installed ACTCC, when i enter the pin number it
says this call will cost 4.04 cents, it does not give a message like you have
100 mins. how do i get a message about the no of mins i have
Tks
Kanishka
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Will it get added to cvs-head?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Bockman
Sent: Martes, 22 de Marzo de 2005 07:39 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Chanspy is back !
> vote f
hi,
i was wondering is it possible to use asterisk manager to send a beeping
sound to a channel to inform the user that he/she has something behind it
to do?
thank you.
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I would appreciate any thoughts anyone
might have on configuring this registration. Thanks
I have a SIP account that I can successfully register with
XTEN and a Sipura-2000. I have yet to be able to get it to authorize with
*.
My XTEN looks like:
Username:
Hi,
how can I completely disable silence suppresion and echo cancelling in
asterisk (and zaphfc)
Thank you very much.
Sebastian
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Hello
We are running Asterisk CVS 22/12/04 and pwlib/oh323 pandora version to work
with our call agent.
Unfortunately **VERY** frequently, asterisk stops responding and goes to 99.9%
CPU. There is no debug output or other information that indicates there is a
problem...
Rather than continually
try using x-ten and see what happens
--- Chris Mason <[EMAIL PROTECTED]> wrote:
> I have installed my first Asterisk implementation
> using the [EMAIL PROTECTED]
> ISO. I am using the SJPhone software. Using the
> setup page, I have been able
> to configure two extensions. Whne I dial from one to
NeuStar also offers CNAM db services, but VeriSign pays you for your cnam
listings as they receive reciprocal compensation for their databases,
probably charging rbocs, clecs etc per query.. I'm not sure about NeuStar
or how they handle this, but I'm almost positive that they provide cnam
updat
On Wed, 2005-03-23 at 01:34 +, Paul Goodyear wrote:
> I'm all up for reading and looking round for people in the same boat
> to try and solve the issue together, but there appears to not be large
> community yet, just the asterisk mail lists.
Actually, the asterisk community is very large IMHO
On Tue, 22 Mar 2005 15:15:18 -0500, Jesse Guardiani <[EMAIL PROTECTED]> wrote:
> > FreeBSD had some issues with Asterisk.
>
> This should be "has some issues". I do not consider
> the FreeBSD zaptel support to be production quality
> in any way. I experienced reproducible system hangs
> (mostly a
On Tue, 2005-03-22 at 17:58 -0800, Scott Wolfe wrote:
> Hi there,
> I have suceseffuly installed Asterisk with a TDM22B and two sip phones
> (Polycom300 and Xlite). My provider is Broadvoice and I am having a heck of
> a time with the audio. All of my calls are broken up and sometimes really
> to
On Mon, 21 Mar 2005 13:15:22 -0500, Andrew Kohlsmith
<[EMAIL PROTECTED]> wrote:
> On March 21, 2005 01:07 pm, Kevin P. Fleming wrote:
> > Well, let's see.. 99.99% of the available VOIP hardware only support
> > SIP, MGCP and H.323, but not IAX2. Is that a good reason?
>
> No. 95% of the marketpla
I have a question which I'm sure has been asked before but my research has
yet to find it.
I have Asterisk running on a Linux server but in order to get it to connect
I needed to punch a hole in my firewall on port 5060 for it to receive the
registration responses from broadvoice.
If I run sjphon
Nice, will have to try it!
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of C F
> Sent: Tuesday, March 22, 2005 7:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Chanspy is back !
>
Tom Samplonius wrote:
I had be using a group of two PRIs for more than a year on a Nortel
PBX. After I started testing with Asterisk through a AS5300 gateway,
I quickly noticed that I could present any calling number.
Yes, we all know we can do that (and do it every day). The poster's
question
On Tue, 22 Mar 2005 19:38:41 -0700, Damon Estep
<[EMAIL PROTECTED]> wrote:
> Anyone planning on putting a variable or some other method to limit
> which channels can be chanspy'd?
>
> How hard would this be?
I believe it's there already.
Look at this:
http://bugs.digium.com/file_download.php?file_
Anyone planning on putting a variable or some other method to limit
which channels can be chanspy'd?
How hard would this be?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Senad Jordanovic
> Sent: Tuesday, March 22, 2005 6:44 PM
> T
On Tue, 22 Mar 2005 14:39:04 -0800, Kris Boutilier
<[EMAIL PROTECTED]> wrote:
> > -Original Message-
> > From: Robert Goodyear [mailto:[EMAIL PROTECTED]
> > Sent: Tuesday, March 22, 2005 1:21 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] Is
Hey, I'm currently using the GotoIf application to set it so if
certain caller ID's call my number, it will transfer it to my cell
phone, here is the code I have so far. I get an error message that
states "call rejected by 198.22.67.70: No such context/extention."
when I call the number from my hou
Hi there,
I have suceseffuly installed Asterisk with a TDM22B and two sip phones
(Polycom300 and Xlite). My provider is Broadvoice and I am having a heck of
a time with the audio. All of my calls are broken up and sometimes really
tough to hear.
My CLI> 'sip show peers' has me usually at OK(128
Jerry wrote:
On Mar 18, 2005, at 2:40 AM, George Pajari wrote:
Vicky Shrestha wrote:
The asterisk configuration and the channel bank configuration are
both set to esf and b8zs. Howerver I am still getting the framing
Error "Red and blinking". zttool shows there are no alarms.
According to the ma
[EMAIL PROTECTED] wrote:
> Guys'n'Gals
>
> vote for bug 3836 - Chanspy is back. Better than ever. Let's get this
> one into CVS.
>
> Julian
And where do you see it as "back" ?
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> vote for bug 3836 - Chanspy is back. Better than ever. Let's get this
> one into CVS.
>
> Julian
SWEET!! I'll test it!
Thanks,
Kevin Bockman
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Thanks Kevin.
I'd already checked the conf file and it looked ok. I recreated it from scratch (it's only got 4 lines!) and it now loads ok.
Cheers
Rob
Rob Gillan
Director
DZhoN Pty. Ltd.
[EMAIL PROTECTED]
www.dzhon.com
M:+61(402) 332 087
F: +61(2) 9383 8386
On 23/03/2005, at 12:26 PM, Kevin P.
I'm all up for reading and looking round for people in the same boat
to try and solve the issue together, but there appears to not be large
community yet, just the asterisk mail lists.
I got Asterisk working with X-Lite great now for internal calls and
also calling land line numbers etc. The two p
On Tue, 22 Mar 2005 19:36:26 +, cmould <[EMAIL PROTECTED]> wrote:
> I am forklifting a Merridian option 51c with 112 Nortel Digital Handsets
> and 400 analog units. For the analog units I have quotes for 9 ADIT 600
> 48 port fxs units and 17 Rhino 24 port FXS channel banks. I have used
> neith
Rob Gillan wrote:
Mar 23 11:42:44 WARNING[15003]: config.c:579 cfg_process: parse error:
No category context for line 14 of cdr_mysql.conf
This doesn't mean it can't find it, this means there is a syntax error
in the file and it cannot be parsed and loaded.
___
I am trying to figuare out how to connect my asterisk box to a server
that uses H323.
I have the did number I ust cannot figuare out how to register to this
proxy.
Http://www.fordvoice.org is the web site address.
Thanks in advance
___
Asterisk-Use
Hi,
Using Zack's -shared replacement posted earlier, addons now compiles. For some reason though, when trying to load it cannot find cdr_mysql.conf even though it's in the /etc/asterisk directory as it should be.
Anyone with any ideas? There's still references to _i386 files that are probably
Verisign, CNAM
http://www.verisign.com/products-services/communications-services/intelligent-database-services/cnam-calling-name-database/page_001662.html
Look there.
-m
-
"Yeah, we rocked the vote all right. Those little
bastards betrayed us agai
We offer that service for our termination customers,
however we can only provide it for (206) area code
numbers. So what we find is people who don't care as
much about the number and more about their callerid
lookup such as businesses and call centers opt to
utilize it. We can even change the name
Guys'n'Gals
vote for bug 3836 - Chanspy is back. Better than ever. Let's get this
one into CVS.
Julian
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To UNSUBSCRIBE or update optio
(Sorry, but my english is very bad)
Hi
I'm newbie with Asterisk, but i was able to install and configure
Asterisk with 3 budgetone 102 and 2 Handytone 206 and works fine for me.
I have a problem and i don't see answer in forums: DNS resolution:
First Day:
==
In configuration menu of t
Robert Rozman wrote:
I've compiled Asterisk cvs stable (few days ago) unde Suse 9.2 without
any problems. We're using te110p and wcte11xp module that is autoloaded
by Suse 9.2.
Card goes green after reboot, but this meesages appear in logs:
Mar 22 11:28:51 linux kernel: Zapata Telephony Interfac
Robert Goodyear wrote:
Does anyone know if there's a service out there to -- for a fee --
inject our DID into the LEC's CLI database so a called party gets
our associated name?
No, only if the LEC servicing the number offers it to you. It is the
responsibility of the operator running the switc
I am forklifting a Merridian option 51c with 112 Nortel Digital Handsets
and 400 analog units. For the analog units I have quotes for 9 ADIT 600
48 port fxs units and 17 Rhino 24 port FXS channel banks. I have used
neither. Which is the best choice? The price difference is not that
great. I
Robert Goodyear wrote:
Does anyone know if there's a service out there to -- for a fee --
inject our DID into the LEC's CLI database so a called party gets our
associated name?
No, only if the LEC servicing the number offers it to you. It is the
responsibility of the operator running the switch
Search the list -- it's virtually impossible, even if you can intercept
the SIP credentials with a packet-sniffer, they're using md5
authentication, so you're pretty much out in the cold. Get an X100P and
connect that... Or buy some IAX minutes from one of the wholesalers.
-Original Message--
I made the Vonage mistake too. Cost me a hundred bucks!
They will hit you with a Disconnect Fee, if you don't return their
equipment in the original box.
Sux, if you ask me.
Vonage does offer an attractive flat rate for 500 minutes.
You could, purchase a digium card, and plug the Vonage POTS
Using the D() option on a Dial, I get only the first digit of my string.
I'm running 1.0.5. I looked at the change log and didn't see anything that
referenced this.
Anybody know what I'm doing wrong?
exten => s,3,Dial(IAX2/[EMAIL PROTECTED]/13115552368,60,D(1234567890123456789))
Thanks in advanc
Check the list archive. This thread just happened a couple days ago.
A couple that I remember as supporting IAX:
voipjet.com
nufone.net
You also might try opbx.com
JD Austin wrote:
> Im a newbie to this list (joined today).
> Other than Broadvoice, what voip providers work well with Asterisk?
> Never had a problem with mine. I set my DHCP server to hand out a
> specific IP to the IAXy, too.
Works on some, but not all. It fails on Microsoft and Cisco DHCP
Servers. Just because it works for you doesn't mean it's implemented
correctly.
> What's the advantage over unplugging the unit and
On 22 Mar 2005 at 16:29, Calvin Hendryx-Parker wrote:
> I have been using the Siemens Gigaset 8825 with a Cisco and Motorola
> ATA and it includes a VMWI on the base and cordless handset. Works
So the VMWI works with both the built in a/m and telco voicemail? Hmm, cool.
Anyone else using on
How about sipXphone? They say STUN will be added within a week or two.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roman Zhovtulya
Sent: Tuesday, March 22, 2005 4:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Aster
Hey Everyone,
This is not an operational issue, and to my knowledge only effects the
look of the command, but when I issue a "sip reload" then a "sip show
peers" I see all of the actual usernames I have assigned in my sip.conf.
However, five minutes later I reissue the sip show peers and all of t
Gotcha
However that's not going to work for me, I need a 2 line analog system that
works for me now, the Asterisk part is slowly coming along, I have a machine
and a couple of the intel based modems that are supposed to act like the
Wildcat cards.
I also have a lot of large projects with clie
remove auth=md5 from your iax.conf and try again.
On Tue, 22 Mar 2005 21:38:15 +0100, Androtech <[EMAIL PROTECTED]> wrote:
> Dear All,
> I bought one IP PHONE from Integrated Networks which was showed to wiki too:
> http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks
> I have prob
> >I've improved the stability of my card by adding a capacitor on the
> >reset line. Hasn't taken a hit in over two weeks.
> >
> >
> Is this the E/F or revised H card? Where and what cap did you install?
My card reports as E/F; only have one, so not sure what the differences
are between the va
I'm calling from outside POTS Line.
Asterisk is connected via Sipura-3000.
I had functional *-1.0.5 installed via Gentoo ebuild and everything was
working. I de-installed Gentoo version and I've updated (or downgraded
to stable CVS version using command):
cvs co -r v1-0 asterisk
The only thing I
On 2005/03/23, at 5:38, Androtech wrote:
Dear All,
I bought one IP PHONE from Integrated Networks which was showed to
wiki too:
http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks
I have problems with the Asterisk authentication. It does't want to
LOG IN to Asterisk; it always says
Words of a user, ... what can I make better?
Most of the calls had a little delay. People on the other end of the
phone said it sounded like cell phone with little stop during the
phone. So, it seems voip is not a good quality pstn phone yet. But I
wondered my classmate that using Dynasky callin
I’ve got a Linksys PAP2 on my Vonage
account with unlimited usage, but my softphone-addon account only has 500
minutes. Anyone ever try to setup their * to mimick the Linksys PAP2 ? Any
comments or suggestions on what problems I might run into if I try?
___
This will turn the lights on but not off~
- Original Message -
From: "Brian S. Adelson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Cc: "David Brodbeck" <[EMAIL PROTECTED]>
Sent: Tuesday, March 22, 2005 3:17 PM
Subject: Re: [Asterisk-Users] Setting M
Hi Androtech,
Androtech wrote:
Dear All,
I bought one IP PHONE from Integrated Networks which was showed to wiki
too:
http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks
I have problems with the Asterisk authentication. It does't want to LOG
IN to Asterisk; it always says "LOG ON F
On Tue, 22 Mar 2005 13:27:29 -0700, Gordon Anderson
<[EMAIL PROTECTED]> wrote:
> Quick description of scenario:
>
> Would like to be able to plug in an analog line to a Digium Wildcard X100P
> FXO card in an * server. Whenever that line receives a call, we would like
> it to automatically ring a
Install asterisk * home.. it will quickly allow you to set things
up... as well as creating the HG.
On Tue, 22 Mar 2005 13:27:29 -0700, Gordon Anderson
<[EMAIL PROTECTED]> wrote:
>
>
> Hello all,
>
> Quick description of scenario:
>
> Would like to be able to plug in an analog line to a Di
ttachment was scrubbed...
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64af90/attachment-0001.htm
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To U
Hi,
I've compiled Asterisk cvs stable (few days ago) unde Suse 9.2 without any
problems. We're using te110p and wcte11xp module that is autoloaded by Suse
9.2.
Card goes green after reboot, but this meesages appear in logs:
Mar 22 11:28:51 linux kernel: Zapata Telephony Interface Registered on m
Robert Goodyear wrote:
Does anyone know if there's a service out there to -- for a fee --
inject our DID into the LEC's CLI database so a called party gets our
associated name?
No, only if the LEC servicing the number offers it to you. It is the
responsibility of the operator running the switch
Wei Su wrote:
I understand Asterisk is more like a B2BUA. But when this INFO request is
sent to asterisk, asterisk is supposed to bridge the request to the other
endpoint, right? In what situation, it decides to send a reply; in what
situation, it decides to bridge the request?
It is not required t
I'm trying to figure out if this is a nat problem.
I have a private network behind a freebsd nat box. The * server is on
a static nat, with a private ip of 10.139.10.165. I'm connecting with
sjphone as the client from 10.139.10.159.
I am calling out using simpletelecom. When connecting directl
> -Original Message-
> From: Robert Goodyear [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, March 22, 2005 1:21 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's
> CLIDB?
>
>
> Does anyone know if there's
David Mandelstam
On the road, hence using Hotmail
reply to [EMAIL PROTECTED]
From: Rich Adamson <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IRQ headaches
Date:
Many thanks. Following the advice fro Martin, the permissions issue is
resolved but now I have a new problem. The number is now dialled and
connected (i.e. the call is placed to the PSTN) but it is immediately
disconnected and I get the following message on the console:
Starting Zap/3-1 at from-
On Tuesday 22 March 2005 3:55 pm, Kristian Kielhofner wrote:
> Jesse Guardiani wrote:
> > Kristian Kielhofner wrote:
> >
> >
> >>Jesse Guardiani wrote:
> >>
> >>>Hello,
> >>>
> >>>Can anyone tell me what the "normal" number of
> >>>interrupts per second is for an X100P card?
> >>
> >>1000 / card
On Mar 22, 2005, at 2:34 PM, Harondel J. Sibble wrote:
1) The base station/corded phone is not wall mountable (not critical)
2) The unit doesn't have WMWI (visual message waiting indicator) for
Telco
provided voicemail, the VMWI only works with the units onboard
answering
system. :-(
I have been
Consider what I've done -- I picked up three separate 5.8GHz
single-handset phones and run them as individual extensions. Multi-line
isn't that interesting in this context, as I can always use distinctive
ring or my cid_rewrite utility to identify which incoming line I'm
dealing with.
> -Orig
Hi Dan,
This sorta works for me. The only thing that doesn't work is the actual
admin functions (changing mode of users from Listen to Listen and Talk,
or Kicking users).
I can see whose in the conferece and see if they are a user or admin though.
kRis
Dan Austin wrote:
> I've had 50+ people d
Hello!
I wanted to make sure that, in addition to my complaints, I make it very
clear: Digium's support is excellent. The jury is still out on the
usefulness of the TDM products. However, Digium has worked very hard to
make sure that this issue is resolved. I actually got an e-mail from
so
And some kind of password to protect the config, so not any intelligent
person with a prov. Utility can change your iaxy!
Back on topic, I much prefer iax personally, however the lack of hardphone
options is a real pain. Obviously depending on the situation, I use SIP
hardphones on internal LAN, I
I have a question which I'm sure has been asked before but my research
has yet to find it.
I have Asterisk running on a Linux server but in order to get it to
connect I needed to punch a hole in my firewall on port 5060 for it to receive
the registration responses from broadvoice.
I
Title: Message
Kerry,
Thanks
for the reply. I THINK I have it figured out and set up properly now after
going through the AMP interface, but we'll have to see once we get the card we
ordered from eBay today. I ended up setting all incoming calls to go to a
call group with all the extensi
[EMAIL PROTECTED] wrote on 03/22/2005 03:56:22 PM:
> On March 22, 2005 03:08 pm, [EMAIL PROTECTED] wrote:
> > The phone in question is what I would consider to be a good-quality GE
> > two-line cordless telephone. Digium's guess is that it is "putting
power
> > on the telephone line and the card
Does anyone know if there's a service out there to -- for a fee --
inject our DID into the LEC's CLI database so a called party gets our
associated name?
/rg
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I am using a SIP
softphone (X-lite, SIPPS or Firefly) connected to an Asterisk v 1.0.3 PBX.
The PBX is also connected via a ISDN-PRI crossover cable to a Avaya Definity
Generic 3 PBX via a TE405P card. All outside of the office calls go
through the Definity. Here's the issue:
Calls to i
Rich Adamson wrote:
>> I'm running Asterisk 1.0.6 with zaptel 1.0.6 on Gentoo
>> Linux with a 2.6.11-gentoo-r2 SMP kernel (but no SMP
>> hardware) and mpg123 0.59s-r9.
>>
>> When I leave a voicemail message via my X100P, the
>> message is way too quiet. I can barely hear it.
>>
>> I googled this
Hmm. Well, I'm not sure what to try, but it works fine if I use GSM or
G711 or even G729. Such a shame, because I liked iLBC.
--
Dana
On Tue, 22 Mar 2005 22:16:17 +0100, Roman Zhovtulya
<[EMAIL PROTECTED]> wrote:
> I'm using the mute switch on the Plantronics headset 90 with SJPhone, so
> never h
Rich Adamson wrote:
If you draw a schematic of the fxo module on the TDM card, its almost
exactly like the tech note schematic for the Silicon Labs chipset.
First guess is that was the starting point for whoever built the
card and modules for digium.
It appears on the surface that whoever did
[EMAIL PROTECTED] (GliTcH) writes:
> I'm trying to investigate going to a different manufacturer, but I
> don't like the Cisco ATA-186's very much and they're too pricey, so I
> don't know where to go next. voipsupply has a pretty big collection,
> maybe I'll order 1 of each for testing.
Grandstr
That seems great, except for some reason, its not working. I'm able to
dial any extension on the system, not the ones I'm trying to define in a
context. Here's what I have that accepts the incoming calls:
[did]
exten => 9995,1,Answer
exten => 9995,n,Background(welcome)
exten => 9995,n,WaitExt
Good point, I forgot about using the softphone add-on account.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Tuesday, March 22, 2005 1:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Use
Did you check SJPhone?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Scott Bussinger
> Sent: Montag, 21. März 2005 22:22
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] why even use SIP
>
>
>
Use ## it's in features.conf
On Tue, 22 Mar 2005 07:42:27 -0700, Damon Estep
<[EMAIL PROTECTED]> wrote:
> Looking for a liitle help if anyone has dealt with this;
>
> The options on dial and queue of t (allow called party to transfer call)
> and T (allow calling aprty to transfer call) seem to w
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