Hello--
I've done some coding for call screening in Asterisk. It's not in
Asterisk yet, mainly because we're waiting for
prompts from Allyson so it sounds like the rest of the system. But
patches, prototype sound files, etc, are all
filed at:
http://bugs.digium.com/bug_view_page.php?bug_id=7
Gonzalo,
Have you tried IAX, I see yo are behind NAT, and my experiences with IAX
behind NAT are much less painful :-)
I've FWD via IAX, receiveing calls (in fact, last time was a nearby
person in Portugal :-) that tested it).
One last thing, you mention dialup client, I guess she is not in dialu
> I am struggling with hardware choices to get started with. My options are
> narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G.
>
> of importance is:
>
> - functionality / integration with asterisk
> - headset functionality and use
> - voice quality
> - build quality
>
> Is t
Hi
I've bought the Wildcard TE110 some days ago but I'm unable to get it to work
with Siemens HiCom 300.
I've tried this so far:
1. I've used standard cat5 cable cut off on one edge and twisted wires 1 to 4
and 2 to 3 which is according to cisco a short circuit for E1 lines. Wildcard
takes a f
He's trying to use sip, not iax. It would appear he's got both a fwd
registration issue and an incoming fwd context issue. They don't appear
to be in sync (probably an understanding of context issue actually).
> Yes, of course you can do that. I have this very setup working for the
> office, wi
> I see from reading the mailing list theres a way to set audio levels on the
> zap channels but I'm wondering if there's a way to set audio levels on
> either sip or iax channels. I'm using some BT-100's and people are saying
> the audio levels are a little low and I would like to bring them u
Inline...
> Hi forum,
> I have been fighting days and days configuring FWD and asterisk with NO
> success
> I have the following scenario.
>
> My sister in Spain with FWD dialup client
> My question is if she can dial my FWD dialup number, which is registered
> in Asterisk and the call being f
OK. I now have call recording working for both incoming and outgoing
calls.
Now I want to make those wavs into mp3. I could launch a script from
cron that checks for new wavs and converts them. But that wouldn't be
so elegant.
Launching it from * on hangup would be nicer. How is it done?
[outg
On Sun, Dec 19, 2004 at 12:21:28AM -0600, Matthew Boehm wrote:
> I'm having a similar problem. Do you have "operator=yes" in your
> voicemail.conf under [general]?
Argh, thats it, solved!
Thanks a lot :)
...cut
--
Tho/\/\as
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Asterisk-Users mailing
how do you integrate Gnugk and Asterisk billing?
Are you using Asterisk's H323 channel?Voip Business <[EMAIL PROTECTED]> wrote:
I integrate Gnugk and a gnugk billing system working like a charm.regardsHAOn Sat, 18 Dec 2004 01:48:56 -0800, Inam <[EMAIL PROTECTED]>wrote:> HI Alll> > this is my first
> Is it possible to send the incoming PSTN caller ID to a Grandstream Budge
> Tone-100 SIP phone? I've configured the extensions.conf file and the log is
As Eric notes, the BT100 phones won't show letters. If a call comes in
without CID, asterisk sends a string like "Asterisk call" which the BT
w
> http://www.voip-info.org/wiki-RTP+Silence+Suppression
>
> http://lists.digium.com/pipermail/asterisk-users/2003-August/018670.html
>
>
> So I am confused. The first says that VAD is supported in RTP. Ok, I know
> that. The
second is kinda ambiguous and seems to imply that *
> doesnt su
Hi
I am struggling with hardware choices to get started with. My options are
narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G.
of importance is:
- functionality / integration with asterisk
- headset functionality and use
- voice quality
- build quality
Is there much of a differ
> > I have * running on Mandrake 10.1 and I to had similar problems in the
> > begging but as soon as I had ztdummy configured correctly everything
> > seemed to just fall into place and work with IAX and *, not that I have
> > got a perfect dialplan as that confuse's me but hey thats another su
HI;
I have an Asterisk with 10 "SIP" ip-phones, our pbx
features are now: Voicemail and Call Transfer.
How can I serve both "Call Waiting / 3 way calling"
for our SIP Phones.?/
Appreciate Any Help
Mohammad
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Asterisk-Users mailing
It seems that all my CDR is dumping into the Master.csv file. There is a way
to create per user/extension CDR but I have looked endlessly in the Wiki,
docs, README.CDR, mailing list archives etc.. I can't seem to find a way to
do this..
Any help would be appreciated.
Thanks!
--
Start Your Own I
Sorry
I mean the voice mail
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer
Sent: Sunday, December 19, 2004 2:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] call screening
Sorry, I don't follo
Hi,
Currently I am using a ISDN BRI PCI FRITZ card (works), would I get
any benefits switching to a HFC card? Or it would be a better choice to
switch to a ISDN with a DSP processor?
Currently I have echo on my CAPI channel when calling analog lines,
if call a cell phone, ISDN or
Sorry, I don't follow.
Dialing *98 will achieve what?
Up until the time I decide to take the call, I want to be able to hear
the person leaving a message interactively, so when I decide to pick up
the call he's still there.
Like a regular answering machine
> -Original Message-
> From: h
On Sunday 19 December 2004 10:13, hadi wrote:
> Yes
> U can do it with asterisk and by dialing *98 on your Ip Phone you can
> listen to your message
No, that's voicemail. ie: The caller leaves a message and hangs up, then you
retreive the message later.
The OP wanted to be able to hear the inc
Yes
U can do it with asterisk and by dialing *98 on your Ip Phone you can listen
to your message
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer
Sent: Sunday, December 19, 2004 1:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussio
Hi all.
Is there a way to use asterisk for call screening?
Meaning, a call comes in, asterisk answers with voicemail after I don't
pickup, and the voicemail prompt + the caller's message a played via the
sound card on asterisk. If I wan't to pick up, I do so by picking up the
phone and dialing so
Martin List-Petersen wrote:
On Sun, 2004-12-19 at 02:11, David Uzzell wrote:
Then the other thing if mem serves me you are running 2.6 kernel so why
not run ztdummy? With the 2.6 kernel this does not require any
specialist Hardware or anything!
Sorry, but maybe you should have read his posts mor
On Sun, 19 Dec 2004, Eric Bishop wrote:
> Apart from the the coolness factor can anyone explain to me in what
> situation one would use TDMoE rather than IAX for communication
> betwwen 2 Asterisk servers?
There are two advantages with TDMoE:
* low latency (prevents far end echo from going from
Yes, of course you can do that. I have this very setup working for the
office, with * aggregating voip and isdn incoming calls and forwarding
them to my laptop wherever I am.
just follow the instructions on the FWD website, and run "iax2 debug" from
the console to see what's happening in anyt
Bruno Hertz wrote:
On Sun, 2004-12-19 at 13:11 +1100, David Uzzell wrote:
I have * running on Mandrake 10.1 and I to had similar problems in the
begging but as soon as I had ztdummy configured correctly everything
seemed to just fall into place and work with IAX and *, not that I have
got a per
On Sun, 2004-12-19 at 02:11, David Uzzell wrote:
> Then the other thing if mem serves me you are running 2.6 kernel so why
> not run ztdummy? With the 2.6 kernel this does not require any
> specialist Hardware or anything!
Sorry, but maybe you should have read his posts more thoroughly. ztdummy
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