I downloaded the astcc calling card program. Thanks, it is very easy to
setup and works Excellent. Anyway, it says to use DeadAGI to run it
rather than AGI. I don't know what I am doing wrong. I just updated my
asterisk from cvs and rebuilt and reinstalled. I do not have an
application cal
On Tue, 31 Aug 2004, Storm D. J. Petersen wrote:
> I have a problem with jitter over a 2mb up 1mb down satellite connection. I
> call my friend over the satellite - I call perfect but they cannot make out
> a word I say. However if I leave him voicemail on his asterisk box, it
> records my voic
On Sun, 29 Aug 2004, Kris Boutilier wrote:
> Is timestamp information calculated purely from the relative timestamps of
> each frame of the current incoming stream or is there some degree of RTC
> synchronization expected between the two endpoints?
No sync is needed; its all relative.
> Simi
On 1 Sep 2004 at 17:15, [EMAIL PROTECTED] wrote:
> A customer of mine has 3 TDM400P cards in a box running asterisk. On
> each card he has four FXO modules.
>
> I have set up the dialplan to dial via group 1 for an outgoing call.
>
> Channels 1-12 are in group 1.
>
> If he plugs a telephone
Hi,
I'm trying to dial in from one phone and give it access to another
line (ie incoming on zap/1 and outgoing on zap/2)... how can I transfer
the call from channel 1 and give it the dial tone on channel 2? I can
use dial but that takes a phone number, which I want the user to be able
to s
A customer of mine has 3 TDM400P cards in a box running asterisk. On
each card he has four FXO modules.
I have set up the dialplan to dial via group 1 for an outgoing call.
Channels 1-12 are in group 1.
If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when
he dials out, it s
On Tue, 31 Aug 2004 15:58:16 -0500, B. J. Bomar <[EMAIL PROTECTED]> wrote:
> I use a Plantronics Supra H51 plugged straight into the headset port, and it
> works great.
>
> B. J.
Same here.
They're wonderful headsets.
-Shaun
___
Asterisk-Users mailing
Look up the word persist in the XML config file...
- Brent
On Tue, 31 Aug 2004, Reid A. Forrest wrote:
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Matthew Marlowe
> > Sent: Monday, August 30, 2004 12:55 PM
> > To: [EMAIL PROT
Chris Jensen wrote:
I am hooking up to a DMS500 (100&250 together) and wanted to see if
anyone had any experience with this. We have the GR-303 span up, the
IDT is built.
I have not yet heard of anyone doing this, but would be _extremely_
interested in your experiences. Please keep in touch with
i'm new here and i need help on how where can i get
software version 4.0.x of the mediatrix and how can i
install it...
mediatrix unit im using has a software version of
2.4.9.57. i would like to use H.323 not SIP...
please need help asap!... hope to hear from anyone of
you soon..
thanks in adva
Just out of curiosity,
What version of CVS and Polycom SIP software are you running happily?
Are you running SIP 2.3.0 yet? 2.2.0? 2.1.0?
I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1 with
poor results. Transferring did not work as expected. Using the # key to
do blin
Tobias Jönsson wrote:
Sorry, I did not know these american specialities. I just noticed in
Larry's PRI debug info that he received a STATUS message during the
waiting, so I thought that the waiting could lead to some kind of
timeout at the telco. In EuroISDN the callerid always come in first
SE
On Aug 31, 2004, at 8:42 AM, Steve Underwood wrote:
Chris Shaw wrote:
- Channel Support:
IAX2 in asterisk
IAX2 in libiax2
Other IP channels in asterisk (RTP-based ones, I guess are all that
is
left).
CNG/VAD and DTX in SIP is a must if * is to be taken seriously as a
complete
solution... As
I need to know how to setup the data side of the T1 on my Linux Box. I
have found information about configuring a PRI and HDLC but nothing
about the Frame-Relay type setup for data.
The following is information from our T1 provider.
Network T1:
Framing = ESF
Line code = B8ZS
Correct. TDM (time division multiplex) FXO is for analog ports coming from
the telco.
- Original Message -
From: "Marcello Lupo" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 31, 2004 1:52 PM
Subject: [Asterisk-Users] Analog lines and TDM card
> Hi,
> sorry to both
try steven sokol's iaxphone and see if you have the same problems dialing
his box while taking * out of the equation. same problem=network, no
problem = *
http://www.sokol-associates.com/IaxPhoneDownload.htm
- Original Message -
From: "Robert Rozman" <[EMAIL PROTECTED]>
To: "Asterisk U
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Matthew Marlowe
> Sent: Monday, August 30, 2004 12:55 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration
>
> I just got a Polycom soundpoint and I set
I set up my own STUN server and turned reinvite
off.
Lyle
- Original Message -
From:
[EMAIL PROTECTED]
To: '[EMAIL PROTECTED]'
Sent: Tuesday, August 31, 2004 8:53
AM
Subject: [Asterisk-Users] SIP
registration with public dynamic ip address
Hi, I'm trying
Maxim-
This will not work through a FWD DID as you suggest. BT requires each
telephone number to be registered in order to receive SMS messages. You
need a either an analogue, BRI, or PRI line that terminates in your asterisk
box directly. The way a line gets registered is that you must initiat
On Tue, 31 Aug 2004 15:34:51 +0200, Axel Eble <[EMAIL PROTECTED]> wrote:
> On Tue, 31 Aug 2004 15:22:26 +0200, Michael Labuschke
> <[EMAIL PROTECTED]> wrote:
> >
> >
> > Pick up mobile phone.. enter sms .. send it to the * phone number.
> > Done
> > On the * side.. follow the sms howto (voip-info.o
Luis Vazquez [EMAIL PROTECTED] wrote:
> Does anybody knows if it's posible or if there is some develoment in
> course to be able to use longer transmit packet sizes (as long as I know
> this is fixed in 20ms now) with the compressed voip codecs in asterisk
> (g729, g726, gsm, etc). I need to use as
Hi all!
I downloaded right mpg123, chabged path to mpg123 binary in
app_mp3.c, rebuilt app_mp3.so, and got MusicOnHold to work. But
MP3Player refuses to do properly:
-- Accepting AUTHENTICATED call from x.x.x.x, requested format =
1024, actual format = 1024
-- Executing Answer("IAX2/[EMAIL
Hi all,
Has anyone gotten custom ring tones to work using ALERT_INFO with the
Cisco 7940 SIP phone? I've read the wiki, but just can't get this to
work. I'm currently using the 7.2 SIP image.
Thanks,
Chris
___
Asterisk-Users mailing list
[EMAIL PROTE
The standard for loop start does not send answer supervision, so * and all
other telcom devices that do CDR records have to 'assume' that the call was
answered.
Lyle
- Original Message -
From: "Rodrigo P. Telles" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discu
Does anybody knows if it's posible or if there is some develoment in
course to be able to use longer transmit packet sizes (as long as I know
this is fixed in 20ms now) with the compressed voip codecs in asterisk
(g729, g726, gsm, etc).
I need to use asterisk to connect remote sip clients with 2
hi;
anyone can recommend a good TTS for the dutch language compat in
linux?
--
Best regards,
Danny mailto:[EMAIL PROTECTED]
belGOnet.com a Euro-pictures division - internet solutions
place princesse elisabeth 9/11 - 1030 Brussels - Belgium
Tel : +32-(0)2-215.67
I have Asterisk RC2 setup here with a Fritz ISDN card on Debian Woody.
I'm using chan_capi-0.3.5 and fcpci-suse8.2-03.11.02. Settings are
pretty much the default, except in order to get the fcpci module to
compile, I had to follow the instructions here:
http://www.voip-info.org/tiki-index.php?pa
The vm-password file is used else where, such as queues. If you changed
it, then you would change for all the other applications.
I guess, since we alter the source code often.. it's not that big of a
deal. We just create our own patch files and if we update from cvs, we
patch against the new sour
Hi all,
I have next configuration:
SIP Provider<--->ADSL router<---localnet
192.168.20.0--->ASTERISK<---localnet 172.24.240.0--->softphones
first localnet 192.168.20.0
second localnet 172.28.240.0
in second localnet we ha
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Nobody knows about that strange "behaviour" of Zap channels or
at least if is that right?
Thanks in advance.
Rodrigo P. Telles wrote:
| Hi,
|
| I'm using a TDM400 with one FXS and one FXO module (developer kit) and
| I've been testing termination fr
On Tuesday 31 August 2004 23:22, Umar Sear wrote:
> On Tue, 2004-08-31 at 10:53, Stephen Hon wrote:
> > Paul,
> >
> > What you can do is modify the source code for the voicemail application.
> >
> > Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the file
> > 'vm-password' to 'pls-
On Tue, 2004-08-31 at 10:53, Stephen Hon wrote:
> Paul,
>
> What you can do is modify the source code for the voicemail application.
>
> Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the file
> 'vm-password' to 'pls-enter-vm-password'.
>
> Recompile and install.
>
> The
I had always thought it was because an early clone of 'meridian
mail' was called 'chameleon mail' and 'comedian mail' is a really good
take off on 'chameleon mail'.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/list
Lol reverse hold!
I can't see that working ever though, I tried it once and the agent at the
other end hung up on me... I had to wait another hour in the queue...
- Original Message -
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 31, 2004 2:53
On Tuesday 31 August 2004 17:36, Kevin Walsh wrote:
> Spam-dialling should be made illegal. I, for one, wouldn't spend two
> seconds adding features to support this sort of usage.
I can think of at least one legitimate use for this -- reverse spam dialling,
or at least "real person" detection.
Chris Shaw [EMAIL PROTECTED] lazily top-posted:
> I've wondered that myself... obviously the writer has a sense of humor! :)
>
> I like the sound of "Digium Mail", it sounds cool...
>
I like the sound of, err, nothing.
Mine just prompts for "Mailbox?"
--
_/ _/ _/_/_/_/ _/_/ _/_/_/
On Tue, 31 Aug 2004 14:14:40 -0400, "Jon Bebeau" <[EMAIL PROTECTED]> wrote:
>Hello all,
>
>I'm working an a switchboard console for Asterisk and would like to investigate using
>IAX Client library to Asterisk. I don't seem to be able to find the source. I'm
>planning on a Win32 app. Guidance
hmm Meridian Voice Mail == Comedian Voice Mail:)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
> suppose I have agents waiting on a queue and I configure asterisk to dial
> out and to forward the call to the first agent enqueued. Asterisk will do
> it even if the answer to the call is "busy".
>
> Is it possible to configure asterisk to detect the busy signal and, in
> that case, dial anothe
I've wondered that myself... obviously the writer has a sense of humor! :)
I like the sound of "Digium Mail", it sounds cool...
- Original Message -
From: "Kevin Walsh" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, Aug
Kris Boutilier [EMAIL PROTECTED] wrote:
> Inquiring (management) minds want to know. I'm assuming it's because 'it's
> funny how simple it really is to write a really decent voicemail system'?
>
Perhaps it was written by someone with a red nose, oversized shoes and
a custard pie. I don't know eit
On Tue, 2004-08-31 at 15:58, Pliva, Josef wrote:
> Unfortunately, I am seeing great many missed IRQs continually...if in fact
> it is that which causes the loss of D-channel.
Then you need to find out why interrupts are being locked for long
enough to make the T100P miss interrupts. Common causes
On Tue, 31 Aug 2004, Benjamin Johnson wrote:
> I found the same with lots of headsets and my 7940, but I've just
> plugged the headset from my Norstar system into the *handset* port on my
> and it works perfectly. It's not ideal but it'll do for now!
Ah, yeah, didn't think of that - works fine.
O
H I guess from a troubleshooting standpoint to try and pinpoint the
problem what I would do is remove all cards from the system and then only
replace the cards that are absolutely necessary like your SCSI card and your
Video card and of course the T100P and then check /proc/interrupts to se
Inquiring (management) minds want to know. I'm assuming it's because 'it's
funny how simple it really is to write a really decent voicemail system'?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
___
Asterisk-Users maili
Hi,
Can anyone recommend a BRI card which works fine with asterisk and which
supports point-to-point mode? Software fax detection should also work.
Price does not matter. :)
Digium seems to sell only PRI cards, and the Beronet drivers for
the quad BRI cards seem to be in an early stage of devel
Hi Chris,
thanks for taking time to look this over.
T100P/* is connected to the Mitel IP-PBX/CU and it to telco - so I think our
setting is correct.
BTW, I did try 0 (as well as 2) without success, just for "fun", before I
came on a good explanation
of the sync source in this forum.
Unfortunat
I use a Plantronics Supra H51 plugged straight into the headset port, and it
works great.
B. J.
-Original Message-
From: Nate Carlson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 15:05
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] OT: Headset for Cisco 7960?
Sorry, I kno
> Zaptel.conf sets t100p to be the primary sync source for the only span, as
> suggested by many Asterisk users.
I'm trying to understand so please bear with me... The T100P is connected
directly to the Mitel? Or to the Telco through a T1?
What I mean is are calls coming into the Mitel from the t
Started a Wiki page here:
http://www.voip-info.org/wiki-Cisco+Phone+Headsets
Jim
James H. Thompson[EMAIL PROTECTED]
- Original Message -
From:
Edward Eastman
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Tuesday, August 31, 2004 10:28
Sorry, I know it's OT, but does anyone know of a relatively inexpensive
headset that is compatible with the Cisco 7960?
I've tried the headset off Norstar phones, doesn't seem to work with or
without the amp.
Plantonics S10 at Office Depot works fine.
___
GN-Netcom has a nice little headset for about US $120. As to the
pin-out,
I believe that the headset port uses pins 1&4 instead of 2&3.
Dan
-Original Message-
From: Edward Eastman [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 1:29 PM
To: 'Asterisk Users Mailing List - Non-Co
Hi
Last week I installed Asterisk (release1) with digium t100p single span T1
(wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is
currently the only user of the t1 line. All worked well for about a half a
day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I
consis
Cisco headset pinout is different from normal ones (grr)
If it's just for you, (ie nothing too professional ;) you can snip the lead
of an existing plantronics type headset and do some reordering - this will
give you the necessary info (sorry - can't remember exactly how I did it):
http://www.mml.
I found the same with lots of headsets and my 7940, but I've just
plugged the headset from my Norstar system into the *handset* port on my
and it works perfectly. It's not ideal but it'll do for now!
Cheers,
Benjamin
Nate Carlson wrote:
I've tried the headset off Norstar phones, doesn't seem to
Hi there
Background:
I want to add DDI and voicemail to users on an existing analogue pabx..
It does not support ISDN.
I have 10 DDI numbers via IAX which I am having sent to my Asterisk
box. I have 2 X100P cards connected to 2 analogue extension ports of my
main legacy analogue pabx. I have
I don't know that Plantronics stuff qualifies as inexpensive but I have
been using Plantronics H headsets with the adapter at this link.
http://store.yahoo.com/founderstelecom/dirconcabfor.html
I have two of these cables and they work very well.
Bryan
Nate Carlson wrote:
Sorry, I know it's OT, bu
Sorry, I know it's OT, but does anyone know of a relatively inexpensive
headset that is compatible with the Cisco 7960?
I've tried the headset off Norstar phones, doesn't seem to work with or
without the amp.
| nate carlson
Hi,
I'd like to implement scenario to send user to operator's queue by default
(if doesn't dial any extension) but only if there is operator agent logged,
so user could get response. If not I'd like to send it to voicemail...
Any quick advice ?
Thanks in advance,
Robert.
__
On Wed, 2004-09-01 at 04:39, Deon Rodden wrote:
> All of my phones use sip, their accounts are in the sip.conf file and
> they have the context of 'company' or whatever. These phones need to be
> able to call each others extension, as well as dial outside to the real
> world. So in that context
Hi,
I'm trying to get Asterisk working on P4 2.8 server behind NAT and Firewall
(all ports we're set according to instructions) on DSL line.
When pbx connects to Digium demo server( I'm located in Slovenia, Europe),
it gets first few words, then silence and then comes back when enumerating
dial p
Nope, Asterisk will not do this, at least not without some serious
busy-detect action going on and some tinkering with the dial and agents
code, in which case any call that is not busy will have to wait a second or
two for Asterisk to say that it isn't busy.
Another way to go is to look into what
[EMAIL PROTECTED] wrote:
> Hi,
> suppose I have agents waiting on a queue and I configure asterisk to
> dial out
> and to forward the call to the first agent enqueued. Asterisk will do
> it even if
> the answer to the call is "busy".
>
> Is it possible to configure asterisk to detect the busy si
AFAIK, this is not possible - but I'll throw it out there anyhow...
I subscribe to telco voicemail, for the event that all my pstn lines are
in use.
Telco gives me a stutter-tone dialtone when I have a message waiting.
Can a Zap card detect this stutter-tone and perform some action?
I'm using TDM
All of my phones use sip, their accounts are in the sip.conf file and
they have the context of 'company' or whatever. These phones need to be
able to call each others extension, as well as dial outside to the real
world. So in that context I put the outbound rules so that the phones
can call ou
Hello all,
I'm working an a switchboard console for Asterisk
and would like to investigate using IAX Client library to Asterisk. I
don't seem to be able to find the source. I'm planning on a Win32
app. Guidance on where the source is or who to "take" to is
requested.
Jon
_
Hi,
suppose I have agents waiting on a queue and I configure asterisk to dial out
and to forward the call to the first agent enqueued. Asterisk will do it even if
the answer to the call is "busy".
Is it possible to configure asterisk to detect the busy signal and, in that
case, dial another num
> We have a Dlink DVG-1120M and were surprised that it was able to handle 2
> simultaneous conversations to 2 seperate phones using only 1 MAC address and
> 1 IP address.
>
> So we asked ourselves..why can't other 1 MAC/1IP devices do this as well?
>
> I have a Grandstream 486 that has 1IP and 1M
Hi,
I installed asterisk-addons and configured it so that the cdr is done on a mysql
database. Everything was fine, until I originated outgoing calls using the
manager API. The call itself is performed perfectly, but when I hangup, I get
the following warning on asterisk CLI:
Aug 31 14:29:23 WA
Hi,
sorry to bother you, but i need to connect 8 standard analog lines to 2
asterisk servers (one in Italy (4 lines) and one in USA (4 lines)) and after
let this 2 systems to interact between them.
I was thinking to use the TDM400 card equipped with 4 FXO modules on both
sides.
Is it correct to do
The HT486 is a single-line device with a PSTN pass-thru. The only multiline
IADs I know of are the SIPURAs and the Cisco ATA-186...
What you do is you create 2 contexts, 1 for each line of the device and you
set the host name to the IP address (or host name if applicable) of the IAD.
Set the usern
It's very possible that the Polycom IP600 will work with this. As it is
just an implementation of a SIP standard for subscribing to the state of
other extensions.
As for the feature improvements you might see them from me, but not very
likely. It is easier for me to train my customers to hit *8
We have a Dlink DVG-1120M and were surprised that it was able to handle 2
simultaneous conversations to 2 seperate phones using only 1 MAC address and
1 IP address.
So we asked ourselves..why can't other 1 MAC/1IP devices do this as well?
I have a Grandstream 486 that has 1IP and 1MAC. But I don'
Why are you including your outbound context into your incoming context in
the first place? That doesn't make any sense?
I'm guessing that because you're using a number in your exten => you're
using an IP channel like SIP or H323? Is this correct? If you're using a
T1/PRI or POTS lines you need to
You should be able to do that, but of course always test, test, test to make
sure.
Lyle
- Original Message -
From: "Deon Rodden" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, August 31, 2004 11:24 AM
Subject: Re: [Ast
Thank you!
I took your advise and replaced the original vm-password.gsm file. Worked like
a charm.
Thanks again,
Paul
--- Jason Kawakami <[EMAIL PROTECTED]> wrote:
>
> - Original Message -
> >
> > Hello All.
> >
> > I'm just beginning with Asterisk and I have it all working now. I'm u
Claus:
One difference is that I'm using the slower ATA disk, not the SCSI.
Is the noise rhythmic (periodic) or constant? If periodic, what is the time
between noise bursts?
Do you hear the noise throughout a call, or just occasionally?
Regards
Scott Stingel
Scott M. Stingel
President,
Emerging
> After reading a retarded amount of docs I'm still unable to figure out how to
> get Asterisk to monitor my phone line and pick it up when the phone
> rings...Im using a voice/fax/data modem on ttyS2. Any tips/pointers to
> another stack of docs? Is this even doable without special hardware?
N
Hi there,
The disks are SCSI Raid hotswap disks 1 RPM, P4 2.8 gig CPU, 1 Gig. of
ram., and the server is running Red Hat 9.0.
The sound is just like hearing a disk just muffled (sounds like strange
static)..
If you have a number I can call you at then you can hear it yourself.
Kind Regards
If I put my outbound rules in a different context, and then "include"
them in my main context, callers who call in will be able to access the
extensions in the main context, but not the "included" (ie the outbound
extensions) extensions called from the outbound context?
Lyle Giese wrote:
You li
My DID is 303 as well.
Marty Mastera wrote:
I'm using RC2 and last weekend's changes from VoicePulse. Outbound
calling and dtmf works fine. However, an inbound call to my DID
cannot
send dtmf digits to the IVR.
Thoughts?
I have the same problem...my iax.conf is set up exactly as recommended
per
After reading a retarded amount of docs I'm still unable to figure out how to
get Asterisk to monitor my phone line and pick it up when the phone
rings...Im using a voice/fax/data modem on ttyS2. Any tips/pointers to
another stack of docs? Is this even doable without special hardware?
TIA,
Nic
You limit them by context. You put your outbound dialing patterns in a
context that inbound callers cann't access.
Lyle
- Original Message -
From: "Deon Rodden" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 31, 2004 9:05 AM
Subject: [Asterisk-Users] limit the length
On Tue, Aug 31, 2004 at 10:15:02AM -0400, Deon Rodden wrote:
> exten => _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
> exten => _1NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
> exten => _1NXXNXX,3,Congestion
This would dial the number twice..?
My config is
exten => _9.,1,Dial,IA
> I have been reading the RFCs and I'm a bit more familiar with how it works
> now although the algorithms are a bit over my head. I am somewhat new to
> RTP/VoIP, but I have a strong telecom/networking background so it makes
> things a bit easier to understand since they share a lot of common
> fe
> I'm using RC2 and last weekend's changes from VoicePulse. Outbound
> calling and dtmf works fine. However, an inbound call to my DID
cannot
> send dtmf digits to the IVR.
>
> Thoughts?
I have the same problem...my iax.conf is set up exactly as recommended
per the recent Voicepulse changes
> This is nothing to do with SIP. It is an RTP issue, common to everything
> which uses RTP - SIP and H.323 included.
I have been reading the RFCs and I'm a bit more familiar with how it works
now although the algorithms are a bit over my head. I am somewhat new to
RTP/VoIP, but I have a strong te
Claus-
This is a problem that interests me, as I'm about to deploy TEN of these at
a customer site, all with TE410P's.
I'm currently load testing one Proliant box (3GHz P4 processor) looping 59
calls out to 59 calls in (leaving one channel open) - ie: lots of load.
While I'm doing this, I call in
On Tue, 2004-08-31 at 10:37, Matthew Marlowe wrote:
> I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up
> fine on my 7960... W/ the name on top and the number below that.
>
> -- Executing NoOp("SIP/614-3ede", "Caller*ID is Matthew Marlowe
> <6092521155>") in new stack
>
>
I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up
fine on my 7960... W/ the name on top and the number below that.
-- Executing NoOp("SIP/614-3ede", "Caller*ID is Matthew Marlowe
<6092521155>") in new stack
When the phone rings, only 'Matthew Marlowe' would display. When
Hi everyone,
I'm having a little problem and was wondering whether anyone would have
any ideas or pointers for me.
I've been working on load-balancing asterisk and have had a pretty
successful setup using LVS and IP tunneling (plus a bit of iptables
nating).
I am only load balancing the SIP reg
Is there a solution for asterisk to send the calling costs
to a display of a grandstream Bt101 phone.
Does anybody know if there is a solution for this?
Greetings Han
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.dig
Put a NoOp(Caller*ID is ${CALLERID}) in your dialplan JUST before the
Dial to the Polycom. See if the correct name and number shows up on the
console when the NoOp runs. If it does, there's a problem in the
Polycom, if there is no NAME then you have a problem with your Asterisk
config.
On Tue, 2
John,
By chance do you know how to set a default ringer?
What I have done is the following:
As you can see, I want 7 to be the default ringer for line 1... For some
reason, it doesn't take these changes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
If that was possible, that would make my life easier as well :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Graves
Sent: Tuesday, August 31, 2004 10:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
I got most of the features of my phone working. Polycom TEch support
refuses to help or even talk to me. So I'll have to ask here again.
On incoming calls, only the NAME is displayed. I am trying to figure
out how to get the NAME & NUMBER displayed.
If anyone can help me do this it would be
1) The samples are empty? No, they have variables with settings. Maybe
I'm not understanding you.
2) I don't know how to dump the current settings to an xml file. You
might try increasing the log level, but I doubt you're going to get a
pretty looking xml file written to the log files. You'
On Tue, Aug 31, 2004 at 11:02:30AM +, Brian Wilkins wrote:
> I had that problem, but apt-get install did the trick.
Not to mention apt-get source and apt-get build-dep if you need to patch
existing packages
--
Tzafrir Cohen +---+
http://www.techn
On Mon, 30 Aug 2004 22:09:26 -0500, John Baker wrote:
>Hmmm...
>
>Hands Free might be:
>
>voice.gain.rx.digital.chassis="15" (15 is my setting)
>
>Call waiting? You can turn it off in sip.cfg - do not disturb settings
>I think. Don't know about gain for call waiting. You might try playing
>wi
- Original Message -
>
> Hello All.
>
> I'm just beginning with Asterisk and I have it all working now. I'm using
> Asterisk 1.0 RC1.
>
> My only question is this; when I check my voice mail the PBX simply says
> "password". I wanted to make it say "please enter your voice mail
password"
1 - 100 of 144 matches
Mail list logo
