I?m looking to install a couple of SIP phones into a small/medium company.
The easiest way would be to simply add the phones on the LAN network. But
what would happened if someone make a huge file transfer: will it make trouble
on the Sip connections ? I think so, that?s why I?m asking you if
On Sun, 2004-08-29 at 21:44, Steve Maroney wrote:
I have some problems with my extensions.conf. When a call from pstn comes
in, the call gets put into the [from-fxo] context. From there the caller
is able to dial sip extensions that are included from the [sip-extenions]
context.
When a sip
On Fri, 27 Aug 2004, Larry Shields wrote:
Thanks for the reply. I tried that initially and it did not work. To
verify I went back and tried again. It answers and still no sound is
heard.
-- Accepting call from '8541' to '2688' on channel 0/2, span 1
-- Executing Wait(Zap/2-1, 3) in new
On Sun, 29 Aug 2004, Kris Boutilier wrote:
Is it possible to make cmd_dial() bridge the audio going out to the network
back to the calling party as soon as dial() starts? Put another way, is it
possible to have the caller hear the outside dialtone and subsequent DTMF
digits? I notice that
On Sun, 29 Aug 2004, Greg Blakely wrote:
I am wondering if it is possible for an extension that is served by a
zaptel device to revert to dial tone once a call disconnects.
For instance, if I make a call to another extension, talk with them, and
THEY hang up, can I then be presented with a
I finally found out the second card was in RED Alarm by running zttool
(cat /proc/zaptel/2 also works). The analog extension in our Merlin
Legend was bad. Plugging the X100P to another available extension
solved the problem.
On Sun, 22 Aug 2004 10:53:46 -0500, spectro [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
The G.729 monopolists have made enough money out of their week's
work, so why give them more?
A better idea is to use a different codec, such as GSM, iLBC or even
ulaw (if you have the bandwidth), and ignore G.729 completely. You
can add several choices in a list
Hi,
In the CDR when a call is placed using X100, the saved duration is the one
starting with the Zap channel connection, not related to the other part
answer.
There is any possibility to know when the other part has answered the call
placed over X100?
I want to know the real call duration in order
I've read the manual, for the most part. I've set the volume xml tags
all to 1 but when the phone first boots the volume is still not on the
highest.
The only other thing that might do the trick is the gain options, which
I don't understand so that's simply beyond me because I don't want to
I've taken a look at:
http://www.voip-info.org/tiki-index.php?page=Getting+MWI+on+Polycom+Phon
es+to+work+with+Asterisk
I've followed those directions but when I press my message key it
doesn't dial the number specified... It actually dials 614p which is the
user I register with.
My MWI
I don't have echo problems on my X100P (at home) but that won't stop
me from dumping it in favour of a Sipura SPA-3000 next month, once it
gets full UK support in firmware (caller ID etc.).
It may not be a big deal, but other considerations are:
There is another box to manager.
Another
I think I've found the problem... I don't think my phone is loading the
configuration file for some reason. The mac address is correct, it's
all lowercase. Everything seems to be set right but I don't think it's
reading it. I don't see anything in the log about it.
Can someone give some
I'm also interested in this, as the other Steve knows.
Anyone in the re-worked jitter-buffer/PLC/DTX crowd besides me going to
be at astricon?
We can at least start working there on requirements. I think I've
wrote this before, but here's what I'd _really_ like to see as
requirements for a
Daryll Strauss [EMAIL PROTECTED] wrote:
On Mon, 2004-08-30 at 09:07, Kevin Walsh wrote:
Asterisk will remain in the loop if you have specified t or T in
your Dial() command, as it will need to listen for the hash key. It
will also remain in the loop if you're recording the audio stream
Hello,
I have been wanting to try a Sayson 480 ADSI phone with our * box, but I haven't had much luck getting the phone to use *'s built in ADSI script.
Does anyone know if there are any how-to's out there for this? Or, could anyone enlighten me?
Thanks,
Craig
Do you Yahoo!?
Read only the
- 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 much as we all hate it's complexity and wish that everything
Hullo over there. i'm trying to link an asterisk box
with a legacy PBX system with a four wire trunk line.
the legacy PBX has 21 analog phones connected to it
and i would like to route calls to another site via
the asterisk box. i would like to use EM signaling
over this line. my question is how
Brent Franks [EMAIL PROTECTED] wrote:
I don't have echo problems on my X100P (at home) but that won't stop
me from dumping it in favour of a Sipura SPA-3000 next month, once it
gets full UK support in firmware (caller ID etc.).
It may not be a big deal, but other considerations are:
At 1:23 PM -0500 on 8/30/04, David Hinkle wrote:
The snom 200 and 220 have five programmable buttons. Each button has a
led that can be used to indecate if an extension is idle, in use, or
ringing. A button pannel for the 220 is also comming out soon that will
have 20'ish programmable buttons on
Thanks. That did the trick.
This is what I ended up with (on extension 45)
exten = 45,1,Dial(Zap/44r1,30,g)
exten = 45,2,System(test ${DIALSTATUS} = NOANSWER)
exten = 45,3,GotoIf($[${DIALSTATUS} = NOANSWER]?4:6)
exten = 45,4,voicemail(u10)
exten = 45,5,Hangup
exten =
On Mon, 2004-08-30 at 16:26, Kevin Walsh wrote:
Brian Wilkins [EMAIL PROTECTED] lazily top-posted:
Point is that unfortunately many systems do use G 729 so it is necessary,
in order to be compatible with existing gatekeepers, to use that codec.
I'd love to use GSM but the existing systems
On Mon, 30 Aug 2004 10:28:59 -0400, Michael Graves wrote:
On Mon, 30 Aug 2004 16:21:04 +0100, Kevin Walsh wrote:
Rich Adamson [EMAIL PROTECTED] wrote:
3. If impendance mismatch is the (or a major contributing) factor, can
we not devise some interface circuit which will allow a variable rate
Hi All,
I'd really like to see a show of hands with regard to people's
experience with FXO interfaces. I own a few X100p cards and have had
nothing but problems with them.
I also took part in Sipura's beta program, for the SPA-3000. While it
can be an improvement over the X100p, it presently
Just a brief reminder to everyone who wishes to attend AstriCon 2004: We
need your registrations ASAP, especially if you plan on staying on-site at
the conference hotel. We have to present the hotel with a solid count of
rooms on Wednesday, so please take a few minutes and sign up at:
Michael Graves wrote:
Hi All,
I'd really like to see a show of hands with regard to people's
experience with FXO interfaces. I own a few X100p cards and have had
nothing but problems with them.
I also took part in Sipura's beta program, for the SPA-3000. While it
can be an improvement over the
Hi!
Have a look at the following entry. I solved this problem:
http://enrico.todo.de/weblog/item/asterisk-oh323-compile-error
Regards
Enrico Stahn
smime.p7s
Description: S/MIME Cryptographic Signature
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From my tests, it doesn't work.
- Original Message -
From: Eric Bart [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 27, 2004 11:12 PM
Subject: does agi wait for digit work in a meetme room ?
I'd like to monitor key press in a meetme room.
Is it possible when
This is the message I posted to the asterisk mailing list detailing how
to configure asterisk to drive the snom programmable buttons.
David
---BeginMessage---
The snom 200 and 220 have five programmable buttons. Each button has a
led that can be used to indecate if an extension is idle, in use,
Nevermind, DUH, I was reading it wrong, it states that they DO NOT contain
CNG algorithms, it describes a way to send CNG on codecs that do not contain
CNG algorithms natively...
-Chris
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Just a little correction. The link to the company's home page should be
http://www.arcturusnetworks.com. But all y'all already figured that out, didn't
you?
(http://www.arcturus.com)
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Hello,
1-2 sec pauses happen while * plays (streams) messages/prompts. We get
reports about that from users and experience it ourselves randomly.
Cannot reproduce it for debugging though, so need to figure out some other ways to fix
it.
1. It's not silence recorded within or pauses between
now that I have finally figured out what I was doing wrong with my
polycom phone and got it to read the configuration file Im changing some
gains.
I successfully changed the gain for the ringer... It was too low for me.
Does anyone know which gain would be for the call waiting and which tone
I'm not certain I understand your question, however: If you have one four
wire EM trunk interface then you can only handle one call over it at a
time. EM is a handshaking protocol, not a multiplexing protocol (such as
the protocols used by T1 circuits, which give 24 channels on one pair).
Hope
Greetings all,
I have been watching Asterisk for a while now, but haven't had the
nerve to jump in and start playing until now... I'm fed up with our phone
system (or lack thereof) at my office, so I decided to start seriously
looking at Asterisk... Mostly as a plaything to get my hands on
I recently dug into this, from what I've seen, the best bang for the
buck out there is going to be Polycom's. A local vendor has Polycom
IP500 phones for $174 shipped to me. IP500 would be comparable to a
7940G I'm assuming. I ran into the same problem with pricing, don't want
grandstreams, but
Hi,
I am running Asterisk CVS from 8/27/04, and since about 8/17/04 Asterisk
crashes on reload. I did remove support for h323 (as it crashes my * at
random, and I don't need it currently).
Here is a cut-out of the last lines when I give a reload command...
== Parsing
It definitely sounded sarcastic :P
Lethol
On Mon, 30 Aug 2004 08:21:06 -0400, Leif Madsen [EMAIL PROTECTED] wrote:
On Mon, 30 Aug 2004 10:21:55 +0800, Joseph Shi [EMAIL PROTECTED] wrote:
Steve Underwood Wrote:
Just wait for the simplified Chinese version to appear in Shenzhen's
Book City.
Heya Kelvin,
Are you using the latest asterisk download from voicetronix webpage.
I got most asterisk features working with an OpenLine4 but I still
have some bugs/incompatibility issues to resolve.
Make sure you download the latest driver and asterisk and make. After
installing the
Benjk,
I dont have an answer to your problem, but I am currently using the
same asterisk CVS HEAD found in voicetronix webpage. Most features
are working OK and I am currently trying fo fix a voicemail problem
but it appears not to be related to loopdrop. Are you sure the card
works fine?
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
with some of the variables in ipmid.cfg under
ringType
John
Matthew
I have recently played with and did get a working copy of Asterisk functioning in the
lab environment.
Then when I moved it out of the lab and onto the outside of our firewall it functions
as before except that I get a
Connected to Asterisk CVS-08/21/04-12:16:04 currently running on 63
chan_skinny refers to support for the cisco Skinny Client Control Protocol
which is used by Cisco IP phones in a Cisco Call manager environment. It
sounds like the PIX is forwarding stuff to port 2000 on your asterisk box.
If you are not using SCCP then you can prevent the module loading by
Is there anyone out there who has VoicePulse Connect working with DTMF?
I've been unable to get it to work from the start, and the recent
VoicePulse updates
did not help.
A caller to my DID's hears Asterisk, but pressing DTMF does nothing:
On call setup iax2 debug shows:
-
Tx-Frame
Bryce Nesbitt (mailing list account) wrote:
Is there anyone out there who has VoicePulse Connect working with DTMF?
I've been unable to get it to work from the start, and the recent
VoicePulse updates
did not help.
I use VoicePulse connect, have similar configs (although I only use iLBC
with
I have discovered something that seems to be backwards. When pressing
flash to end a three way call Between me, Party A and Party B,
Asterisk will drop Party A instead of Party B. My Telcos version of
three calling will drop Party B when ending the three way call.
In my testing, Party A is a
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