James wrote:
I've seen something with the X101P that lead me to think so: I have two
cards and two lines. I also own a small UPS that happend to have a jack
for a phone line, to act as a power cleaner and I've put the line that
goes to one of these cards there.
Surge arrestors used for POTS
On Sat, 1 Jan 2005, Steve Murphy wrote:
What I'd like to do:
Use IAX softphones running on computers, in Auto-answer mode, with sound
going to speakers, as a sort of public announcement system.
What isn't working:
Well, my first experiment was to set up the MeetMe system described on
the
Dear members,
I have 2 locations with broadband internet. There will be an * on one
location and Wifi stations on both locations. The locations are to far away
to have the Wifi covering the distance. I want to use about 12 2000W
Portables phones on both places. The Wifi stations will not carry
Hi,
I wonder if you're willing to share your setup with autoanswer mode...
Regards,
Rob.
- Original Message -
From: Steve Murphy [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, January 02, 2005 7:50 AM
Subject: [Asterisk-Users] Announcements via IAX phones
On 2 Jan 2005, at 10:04, Satchid wrote:
Dear members,
I have 2 locations with broadband internet. There will be an * on one
location and Wifi stations on both locations. The locations are to far
away
to have the Wifi covering the distance. I want to use about 12 2000W
Portables phones on both
Using 1.0.3 and spandsp 0.0.1k.
Having spoken to coppice, I built spandsp-0.0.2pre6 on his suggestion.
He made a change that detects asterisk version and handles the cid
structure member difference.
However, no difference is seen in our receive results. Sending
software still says
Getting
Wilson Pickett wrote:
Using 1.0.3 and spandsp 0.0.1k.
Having spoken to coppice, I built spandsp-0.0.2pre6 on his suggestion.
He made a change that detects asterisk version and handles the cid
structure member difference.
However, no difference is seen in our receive results. Sending
software
Can it be that the MeetMe application is not installed by
default even if there is a meetme.conf ?
pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for
extension (from-sip, 550, 4)
Regards,
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I have installed asterisk on a server and everything is fine. I get asterisk
to play a sound when a call arrives.
However
If I switch off the server, when it boots I get an annoying very loud buzz
from the speakers.
So I switch off. Remove the PCI sound card. Enable the on board sounds card
- Original Message -
From: Gary Ruddock (Swift Drinks) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, January 02, 2005 11:56 AM
Subject: [Asterisk-Users] Sound Card Buzzing
I have installed asterisk on a
On Sun, 2 Jan 2005, Serge Schumacher wrote:
Can it be that the MeetMe application is not installed by default even if
there is a meetme.conf ?
pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for extension
(from-sip, 550, 4)
The meetme application is built if the make script finds
I have installed asterisk on a server and everything is fine. I get
asterisk to play a sound when a call arrives.
However
If I switch off the server, when it boots I get an annoying very loud
buzz from the speakers.
P.S.The loud buzzing stops if i kill asterisk.
add these to modules.conf.
What about the wisip VOIP handset?
And how do I proceed?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of tim panton
Sent: Sunday, January 02, 2005 12:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Serge Schumacher wrote:
Can it be that the MeetMe application is not installed by default even
if there is a meetme.conf ?
pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for extension
(from-sip, 550, 4)
It is not installed if you haven't got a Zaptel timer. See the Wiki docs
on
Hi all,
Some people are successfully using my MFC/R2 support for *, while a
couple have reported that outgoing calls foul up, and they hear strange
noises. unicall-0.0.2pre1 should fix this problem. You will need to
replace all the unicall code, as the APIs between the libraries have
changed.
Hi ALL;
I installed ASTCC, but the sound files(*.gsm)
cannot be opened.All of them are located in :
/usr/share/asterisk/sound.
Asterisk says:
Jan 2 06:01:54 WARNING[9759]: file.c:475
ast_openstream: File astcc-tone does not exist in any formatJan 2
06:01:54 WARNING[9759]: res_agi.c:436
Dear Group,
Is the Cisco Wireless IP Phone 7920 compatible with *
Is it gooed?
Thanks,
Willy
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I have the following line in my extensions.conf which when I dial 100
(from my BudgeTone 100, and then wait a few seconds) I will get an
outside line.
exten =100,1,Dial(Zap/1,20)
What do I need to put in the extensions.conf file so that I can dial 9
(and then a number) and then Asterisk
On Sat, Jan 01, 2005 at 07:23:58PM -0500, Jim Van Meggelen wrote:
[...]
What if, for example, the TDM400 issues were a cumulative thing? If you
had over 6dB of attenuation on the PSTN loop, coupled with greater than
5V potential on the neutral-ground of your elecrical receptacle,
compounded
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, January 01, 2005 9:12 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Qs about FXO/FXS cards
[...]
I'm running older, but solid hardware and
richard wrote:
I have the following line in my extensions.conf which when I dial 100
(from my BudgeTone 100, and then wait a few seconds) I will get an
outside line.
exten =100,1,Dial(Zap/1,20)
What do I need to put in the extensions.conf file so that I can dial 9
(and then a number) and then
You can do something like this:
[directdial]
ignorepat = 9
exten = 9,1,Dial(Zap/1/)
this will give the channel 1 on zap, it justs gives you the Dial tone
on it and then you can dial.
or you can do this:
[localdial]
ignorepat = 9
exten = _9NXX,1,Dial(Zap/1/w${EXTEN:1})
this will dial the seven
I have almost the same problem (but in reverse). When I make a call the
other person can hear me but I can't hear the other person.
I believe it has something to do with the router i'm using to connect the
asterisk to the net, because if I connect my pc to that same router to an
ethernet port of
On Friday 31 December 2004 09:41 pm, Charles S. Antrim wrote:
I have a stable server and want to upgrade. How do I upgrade to the
latest version of * ?
Try (wget or http) szmidt.org/asterisk/asterisk-update.sh
Run it without any parameters to see your available options.
It will perform a
Gary,
IP trading is the vendor of this... Does it have any distributor in Europe?
Or website to check for informations?
Thanks
Helder
- Original Message -
From: Gary [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
The AGI looks in
/var/lib/asterisk/sounds for its sound files with names not fully
qualified.
This looks like a bug in
the ASTCC makefile or the astcc.agi should refer to the soundfiles in the
/usr/share/asterisk/sounds directory explicitly.
Karl
Putz
-Original Message-From:
One Sigurd Karlsbakk, there's only one Sigurd Karlsbakk, one Sigurd
Karlsbakk, there's only one Sigurd Karlsbakk.
Ta la.
- Original Message -
From: Roy Sigurd Karlsbakk [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
[...]
I'm running older, but solid hardware and not seeing any
issues. I'm using a Compaq Proliant 1850R Gen1 dual PII 400
with 512MB ram, GB ethernet, and SATA Hardware RAID. Cheap,
efficient, redundant. And for a Debian box, good enough.
[...]
I just have to add my $0.02
C F wrote:
You can do something like this:
[directdial]
ignorepat = 9
exten = 9,1,Dial(Zap/1/)
this will give the channel 1 on zap, it justs gives you the Dial tone
on it and then you can dial.
or you can do this:
[localdial]
ignorepat = 9
exten = _9NXX,1,Dial(Zap/1/w${EXTEN:1})
this will
Hi All,
I've read J.R. Richardson's paper Create an Embedded Asterisk Server
which outlines making a Debian server that boots from a compressed disc
image on a CF card. I'm really interested in this as I want my * server
to be more like an appliance than a PC. However, the paper is only an
Michael Graves wrote:
Hi All,
I've read J.R. Richardson's paper Create an Embedded Asterisk Server
which outlines making a Debian server that boots from a compressed disc
image on a CF card. I'm really interested in this as I want my * server
to be more like an appliance than a PC. However, the
Hi,
Steve Underwood schrieb:
Recent versions no longer require libtiff internals.
Cool!
Recent versions of Debian should have the libtiff bugs fixed. Some
people trying to work with spandsp got them in there :-)
:-) Debian packages of spandsp and app_dtmftotext, app_rxfax and
app_txfax are built
You can use it with SCCP channel. We have tried and it works well. But I
don't recommend this phone it's battery is not enough. It has a 3-4 hours
time of standby.
Best Regards.
Yusuf Alakavuk
Teknik Danisman - Technical Consultant
Grid Bilisim Teknolojileri A.S.
Kustepe Mahallesi Leylak
This has been an interesting discussion. I'll chime in with my
experience here.
I have two servers. One with the cheapest motherboard and athlon
processor I could find on Newegg.com. The other is a 1999 era
motherboard with a Via C3 processor, again a bargain basement special.
The Athlon
On Sun, 02 Jan 2005 09:47:22 -0800, Steven P. Donegan wrote:
Michael Graves wrote:
Hi All,
I've read J.R. Richardson's paper Create an Embedded Asterisk Server
which outlines making a Debian server that boots from a compressed disc
image on a CF card. I'm really interested in this as I want my
Hello,
X-Lite in German:
http://www.globalipphones.com/xlite
Gunnar
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Afraid not (yet). I contacted the maintainer of chan_sccp (Jan Czmok) and
I sent him 2 tcpdump files one with the skinny and succesfull registration
and the other with chan_sccp.
This was just before x-mas though and I haven't received a reply yet,
probably due to the holidays.
The IP600 will
Serge Schumacher wrote:
Can it be that the MeetMe application is not installed by default even if
there is a meetme.conf ?
pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for extension
(from-sip, 550, 4)
If you don't have zaptel installed Astrisk won't build Meetme.
You should be able to boot a full system from a 32M card without a problem.
I've read J.R. Richardson's paper Create an Embedded Asterisk Server
which outlines making a Debian server that boots from a compressed disc
image on a CF card. I'm really interested in this as I want my * server
to
Hello,
X-Lite in German:
http://www.globalipphones.com/xlite
Thank you! That is exactly what I was looking for.
Thanks,
Adi
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To
I am looking at some lower cost phone to use with Asterisk. What is the
ArtDio IPF-2000 or the Sipura SPA-841 like? Also, I see voipsupply.com has
an ArtDio IPF-1000 listed, is this a new or an old model? I cannot find
any information on it.
Adi
___
[EMAIL PROTECTED] wrote:
This has been an interesting discussion. I'll chime in with my
experience here.
I have two servers. One with the cheapest motherboard and athlon
processor I could find on Newegg.com. The other is a 1999 era
motherboard with a Via C3 processor, again a bargain
Dorn Hetzel wrote:
On Sat, Jan 01, 2005 at 07:23:58PM -0500, Jim Van Meggelen wrote:
[...]
What if, for example, the TDM400 issues were a cumulative thing? If
you had over 6dB of attenuation on the PSTN loop, coupled with
greater than 5V potential on the neutral-ground of your elecrical
Michael Graves wrote:
Yeah, I'm familiar with this. I even own a Soekris box for my firewall.
However, I need something with more CPU power and the ability to take a
TDM400 card. So I'm really trying to make the plain vanilla PC that I
presently use into something that's more robust. I'd like to
One thing to look at is the proximedy to the powersupply of your audio
devices. Some mobos have their chipset integrated in very closely to
their power supply pins. With an unclean power source the fluxuations
would be enough to add some of the white noise which would give you the
whine.
I'm hoping someone can help me with a problem I've been having for a while
now. I've googled and wiki'd to no avail.
Whenever I place an outbound call from * to a PSTN through a SIP or IAX
provider (e.g. Voicepulse or Broadvoice), the first 1/2 to 2 seconds of the
remote call are clipped
Hi list!
I have been experiencing a lot of mystical hangs on my box. At first I
suspected bad memory or motherboard but I found out that the mystical
hangs disappear when I do not load the drivers for my x100p.
The box will only hang under high load (copy operations with 30Mb/sec to a
raid
I am wondering how Asterisk selects codecs between devices. For example,
in my sip.conf I have:
disallow=all
allow=ulaw
allow=alaw
allow=g729
Does the order matter? Does it mean it will try each codec in succession
and use the first that both endpoints support?
Thanks,
Adi
Hi to all,
I recently installed a Fritz PCI ISDN BRI card in hope that the echo
problems would go away, but unfortunately it didn't solve the problem when
calling analog lines, when calling cell phones and PBX with a E1/T1 it works
great.
Can anyone suggest what would be a way to
Victor Rini wrote:
This has been an interesting discussion. I'll chime in with my
experience here.
I have two servers. One with the cheapest motherboard and athlon
processor I could find on Newegg.com. The other is a 1999 era
motherboard with a Via C3 processor, again a bargain basement
I have more FXO ports on TDM400's than I have PSTN lines available for
testing. When all the lines were used up (the FXO ports are all in
zap group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial
succeeded even though there is neither line voltage nor dial tone.
Can at least
Victor Rini wrote:
This has been an interesting discussion. I'll chime in with my
experience here.
I have two servers. One with the cheapest motherboard and athlon
processor I could find on Newegg.com. The other is a 1999 era
motherboard with a Via C3 processor, again a bargain
The order does matter.
You want to offer ulaw, alaw and g729. I want to offer g726, g729 and
gsm.
In the order you have them listed in your stanza you offer them to me.
In the order I have them in mine I offer them to you. In this case only
g729 matches our lists and so we choose g729 for the
Adi, spend your money on something else. I've tried it (Zyxel) and it is
simply bad. If you want to go cordless - go VOIP-DECT. That seems to be the
best working solution at the time. Check wiki for reference.
Vel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi there, what phones are available that have two ethernet ports?
I want to do some cabling at a new installation and i heard there are
such phones (SIP i guess) out there. That way i dont have to run two
cat5 to the user desktop.
I think 3COM had one but can't find the web site reference for the
Update:
found the 3Com® 3101 Basic Speaker Phone
Provides dual port 10/100 switched Ethernet for one-wire connectivity
between the phone and a PC
any others not so expensive? does these 3com sip phones work with * ?
On Sun, 2 Jan 2005 16:35:12 -0500, Erick Perez [EMAIL PROTECTED] wrote:
Hi
Yes, this makes a lot of sense and gives me the information I am looking
for!
Thanks,
Adi
The order does matter.
You want to offer ulaw, alaw and g729. I want to offer g726, g729 and
gsm.
In the order you have them listed in your stanza you offer them to me.
In the order I have them in
Some of the Cisco phones do (7940,7960 etc, also the 7910+SW but this is
skinny only).
Steve
-Original Message-
From: Erick Perez [mailto:[EMAIL PROTECTED]
Sent: 02 January 2005 21:35
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] phones with two ethernet ports
Hi
Rich Adamson wrote:
Have you noticed that a TDM with fxo modules is more/less stable then
a TDM with only fxs modules?
Gut feeling (no reasonable analysis at all) from various postings tend
to suggest the TDM with fxo's is less stable. Would you agree or not?
Yes.
Also, could you share the driver
I have more FXO ports on TDM400's than I have PSTN lines available for
testing. When all the lines were used up (the FXO ports are all in
zap group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial
succeeded even though there is neither line voltage nor dial tone.
Can at
Erick Perez wrote:
any others not so expensive? does these 3com sip phones work with * ?
http://www.grandstream.com
The BT102 has 2 ethernet ports.
Doug
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On Mon, 2005-01-03 at 08:35, Erick Perez wrote:
Hi there, what phones are available that have two ethernet ports?
I want to do some cabling at a new installation and i heard there are
such phones (SIP i guess) out there. That way i dont have to run two
cat5 to the user desktop.
I think 3COM
After some reading I see that DECT requires dedicated base station
hardware. I am looking for something to work with established 802.11
infrastructure. I've used a Cisco 7920 on Cisco CallManager. It works
quite well. Unfortunately its price is way out of line for my purposes.
Adi
On Sun, 2 Jan
Hi,
I'm reading that spandsp works only with zaptel channels. What are my
options if I want to receive faxes through ISDN Fritz card with Asterisk and
possibly forward it as emails ?
Regards,
Rob.
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Thanks, I've applied the fix to two systems here. Guess I'll have to
wait for recurrence as well. But, hopefully I'll have to wait
forever. :)
Rich
Have you noticed that a TDM with fxo modules is more/less stable then
a TDM with only fxs modules?
Gut feeling
Hi;
In IAX, because both signaling and rtp ports are
uniqe, so Asteriskis always in rtp path. Am I right???/
If yes, is there anyway to bypass Asterisk from rtp
traffic?
Regards
Mohammad
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More on this problem.
Sometimes I get the following error message instead of the original one
that I quoted:
# ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
2 channels
After setting the pridialplan=unknown I seeing the Called Number TON change
to Unknown Number Type but not the Calling Number TON. Should both be
following this parameter or not. If not is their another option to change
the Calling Number TON?
Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT
I've heard good reports of the Senao Wi-Fi SIP phone.
Mike
On Sun, 2 Jan 2005 16:12:29 -0600, Adi Linden [EMAIL PROTECTED] wrote:
After some reading I see that DECT requires dedicated base station
hardware. I am looking for something to work with established 802.11
infrastructure. I've used
Have a problem which can't find solution to on WIKI..
Trying to get * to use UK based indication tones. i.e. british ring, dial
tone, busy signal.
Have changed the indications.conf file to default to UK. However this seems
to have no affect. What am i missing. Am using 1.0.3 stable.
Many
Any idea where it can be purchased.
Adi
On Sun, 2 Jan 2005, Mike Dent wrote:
I've heard good reports of the Senao Wi-Fi SIP phone.
Mike
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The manufacturers are chinese,
IP Trading is in Sydney Australia, they sell the units as a distributor
(I believe) and thats where i have been getting mine.
I've had unsolicited emails from different chinese manufacturers but
they have yet follow up on my replies/questons.
Gary
On Sun, 2 Jan
On Mon, 2005-01-03 at 09:59, Andrew Brown wrote:
Have a problem which can't find solution to on WIKI..
Trying to get * to use UK based indication tones. i.e. british ring, dial
tone, busy signal.
Have changed the indications.conf file to default to UK. However this seems
to have no
Hi,
Can someone see what's wrong here please ?
I've installed the ztdummy driver to enable meetme, put his in my
extension.conf
exten = 550,1,Answer
exten = 550,2,Wait(1)
exten = 550,4,MeetMe(18|Md)
exten = 550,5,Hangup
this in my meetme.conf
[rooms]
;
; Usage is conf = confno[,pin]
;
conf =
Maybe some light finaly.
The only thing the English and Americans don't have in common is the
Langauge. I wish they would change their PSTN signaling to the
american way so this statement becomes true. Would you be the one to
start?
It's Sunday and I need a break so I decided to go thru the list,
Jim Van Meggelen wrote:
If I may, I'd like to ask you some general questions about the
environment these systems are running in.
- How are these systems powered and grounded?
Not optimally by a longshot. On the Athlon machine, my main machine, all
the equipment is plugged into 2to3 prong
On Mon, 2005-01-03 at 10:16, Adi Linden wrote:
Any idea where it can be purchased.
Adi
On Sun, 2 Jan 2005, Mike Dent wrote:
I've heard good reports of the Senao Wi-Fi SIP phone.
You could try their web site senao.com.tw if you can get past the
stupid Chinese M$ IIS 404 page.
Mike
Randy MacKay wrote:
I tried Firefly in WinXP and it works fine. I tried it on Win98, it looks
it up. Anyone experience this? Any ideas?
Was running fine here until the Windows 98SE box blew itself up...all
Linux now...bummer I've lost CounterStrike though!
But seriously, where does it crash?
Try making sure that the numbers match the one in meetme.conf.
I don't know if this makes a difference but since in my config this is
how it is setup and it works, just give it a try.
On Mon, 3 Jan 2005 00:34:25 +0100, Serge Schumacher [EMAIL PROTECTED] wrote:
Hi,
Can someone see what's
On Sun, 2 Jan 2005, Glenn Dalgliesh wrote:
After setting the pridialplan=unknown I seeing the Called Number TON change
to Unknown Number Type but not the Calling Number TON. Should both be
following this parameter or not. If not is their another option to change
the Calling Number TON?
Jim Van Meggelen wrote:
Frankly, what is most interesting is the fact that your systems are
trouble-free. Certainly if you were to ask if such systems could be put
into production, you would probably be advised not to expect much.
One last thing. I have a somewhat special PSTN connection. I
On Sun, 2 Jan 2005 16:35:12 -0500, Erick Perez [EMAIL PROTECTED] wrote:
Hi there, what phones are available that have two ethernet ports?
Check out the Polycoms. IP500 and IP600.
-Chuji
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On Mon, 2005-01-03 at 10:38, C F wrote:
Maybe some light finaly.
The only thing the English and Americans don't have in common is the
Langauge. I wish they would change their PSTN signaling to the
american way so this statement becomes true. Would you be the one to
start?
Two nations divided
Can someone see what's wrong here please ?
exten = 550,1,Answer
exten = 550,2,Wait(1)
exten = 550,4,MeetMe(18|Md)
exten = 550,5,Hangup
The priority 3 is missing ... 1, 2 then timeout...
--
Joel Vandal
ScopServ Inc.
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Hi,
I'm trying to get my Agent onto an extension when they finish a call.
Actually i'm doing that way:
Call enter the the context 'cc-incoming' get into a Queue and then, when
someone hangup (caller or callee) I launch an AGI which initiate a call to
my Agent and then ask for a report code.
Adi - Yes I believe the IPF-1000 is formally discontinued, although I
believe we still have some stock of them, and we will certainly warranty and
support them.
You can verify stock status by emailing one of our sales associates,
[EMAIL PROTECTED]
SPA-841 still has not shipped from Sipura, we
Howard Lowndes wrote:
It's Sunday and I need a break so I decided to go thru the list, so I
had to laugh to myself when reading this and decided to write this I
hope someone elae finds this amusing as well.
Why would we want to move away from the UK dial tones? They work well
there and in
When I start Firefly, I see the little firefly box appear (flash on the
screen), then the screen turns blue with an Error #44. I am unable to do
anything, but reboot the computer.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matt
Sent: Sunday,
Serge Schumacher wrote:
exten = 550,1,Answer
exten = 550,2,Wait(1)
exten = 550,4,MeetMe(18|Md)
exten = 550,5,Hangup
and when I call 550 I get this error and the MusicOnHold (exten =
550,4,MeetMe(18|Md)) also doesn't work:
-- Executing Answer(SIP/ses-0730, ) in new stack
-- Executing
mohammad wrote:
In IAX, because both signaling and rtp ports are uniqe, so Asterisk is always in rtp path. Am I right???/
IAX and IAX2 use the same port for signaling and audio. IAX and IAX2 do
NOT use RTP.
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Hi List,
Don't deal with this guy (C F shmaltz at gmail.com), he is misleading
people
and just using the replies to show that what ever he is selling to his
clients is popular.
After a couple of exchanges:
C F wrote:
Thanks for your replay, at this time I'm just waiting for a commitment
from
I like the quality of the uniden UIP200. They just relesed new firmware and
I don't yet have it workin behind a nat, but I am working on that. Uniden
didn't post new docs for the new firmware yet.
Lyle
- Original Message -
From: Erick Perez [EMAIL PROTECTED]
To:
Randy MacKay wrote:
When I start Firefly, I see the little firefly box appear (flash on the
screen), then the screen turns blue with an Error #44. I am unable to do
anything, but reboot the computer.
Is the computer clean from viruses and spyware/adware?
Is it running any firewall software which
On January 1, 2005 04:09 pm, Rich Adamson wrote:
b. don't ever post anything to the -dev list regarding a TDM card as
that is NOT the forum for digium cards or drivers,
Eh? If you're hacking on the code for wctdm, -dev is most certainly an
appropriate place to post. If you're just going
On January 1, 2005 06:24 pm, Steven Critchfield wrote:
1. Power alarms. WTF does that mean? Wish I had some support docs.
2. On bootup, Excessive leakage module x, ProSLIC failed Auto
Configuration. Again, WTF? Reboot and it's ok. But, just a reboot
after driving 100+ miles to the
Not sure that's an accurate subject line, but...
I've started looking at Asterisk as a possibility for a small PBX here.
I'm thinking of an ISDN (BRI) card for connection to the telco, with some
analogue converters (Sipura) for existing phones too.
I'm pretty sure that the functionality I want
Well
disallow=all
allow=ulaw,alaw,g729
That works in cvs-stable and head now. For sip order can/does matter on
global and per user/peer.
More people need to test and comment on this:
http://bugs.digium.com/bug_view_page.php?bug_id=0002971
Don't say we didn't ask you to comment when it goes
Karl,
Naughty, you know better than to do this in a public forum. If
you're directing comments like this at a person you're leaving your ass
flapping in the wind since it could be considered slander, defamation or my
favorite libel. If you don't have hard facts to back something like
Mr Brose.
I don't see why you did this? why was this needed?
All I wanted was to show my client that * is popular. Without this
none of us can sell this if we want. Nobody buys a pbx they can't get
support on (we are talking about a $50,000 installation). Therefore I
had to show my client that it
On Sun, 02 Jan 2005 18:15:53 -0600, Eric Wieling aka ManxPower wrote:
mohammad wrote:
In IAX, because both signaling and rtp ports are uniqe, so Asterisk is
always in rtp path. Am I right???/
IAX and IAX2 use the same port for signaling and audio. IAX and IAX2 do
NOT use RTP.
I answer
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