Andrew, you are right on with your final point about absurdity.
Hopefully this vile top-posting will illustrate exactly why.
Sorry, I couldn't resist.
B.
Andrew Kohlsmith wrote:
On Thursday 17 June 2004 09:21, Troy Settle wrote:
However, my preference is for top posting. The reason, is that in
I should follow this up to accurately state that audio was not
operational in my test calls from the PDA. I have patched the
iaxclient library with the changes available from ZiaxPhone that word
align the IAX2 library on the ARM platform. I haven't finished
compiling a new binary to test
i'm new to asterisk and am having trouble placing outbound calls. i
Bug Grandstream so that they finally fix their buggy software.
The GS phone sends occassional SIP packets to port 0, not to port 5060, as
tcpdump or (better) ethereal will show you.
There's a page on this at voip-info.org.
We have a customer who is connected to our PSTN gateway using IAX and
noticing that even when the traffic from their site is modest their outbound
audio has short dropouts. Inbound audio is fine. (They have ADSL so it is
expected that outbound audio would be the first to experience problems.)
We
You're basically looking for hotline functionality. I'm using Sipuras
for my FXS ports, and they can be configured to dial a phone number upon
pickup. I played with that before, and the call was established so
quickly that I had to add a Wait instruction in there so the receiver
could make it to
Hi Everybody,
as a relative newby I´m just trying to get a Draytek Vigor Router (2600Vi)
connected to my Asterisk System (CVS-05/31/04). With X-Lite and a Cisco
Phone it is no problem, but the Vigor seems to have some problems with
Asterisk.
The first thing ist when I do a sip show peers on the
On Thu, 17 Jun 2004, George Pajari wrote:
Q1: Are there any statistics collected/available or diagnostics tools to
tell us how much of this can be attributed to packet loss and how much to
packet jitter and to measure quantitatively how bad this is?
Q2: Is jitterbuf working well enough to
Jeremy
I speak for myself, I have been testing with oh323 driver as well, because
in my case, your h323 driver is not working, it was working before, but then
when I started to upgrade to 0.7.0 version of asterisk and from that point
onwards (beginning of January), calls have had no audio. I
T. Chan wrote:
Jeremy
I speak for myself, I have been testing with oh323 driver as well, because
in my case, your h323 driver is not working, it was working before, but then
when I started to upgrade to 0.7.0 version of asterisk and from that point
onwards (beginning of January), calls have had no
call from PBX with analog FXS line to ISDN PRI T100P
if I use number analize exten = 452., dial call not working becouse
Asterisk get connect to analog line and analog line not proclaim all
number for call
if I useexten = 452XXX, Dial call working after pres on
analog phone all
Hello,
I would like to know if someone gets a doc which resumes what changes need
a reload and what changes need a restart of asterisk.
Thanks.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi all,
I'm experiencing problems with the TDM card
with 4 fxo modules. on all tests,
if the cards has 4 modules, I get
poopy kernel messages on the card.
The card works for sometime,then hangs
and a asterisk restart must be done,
along with kern modules unload/reload .
if I remove the first
Jeremy,
Yes, I felt that it was important to report my trouble and I did it three
times, reporting to the asterisk community, but for some reasons, I was not
being responded to at all. I thought my messages were embedded among the
hundreds of them and were missed out or everyone was having the
SNIP
On the other hand... Go take a look at all of the ~$100 wireless
router/firewall/print server/gateway boxes on the market, and you'll see
one thing that almost all of them have in common: they all run Linux.
Most of them are even based on the same small number of tools; things
like
Kannaiyan Natesan [EMAIL PROTECTED] wrote:
I got the dring value from the following call log.
-- Detected ring pattern: 337,0,0
Here is the configuration for my BT Line:
usedistinctiveringdetection=yes
dring1 = 367,0,0
dring1context = default
dring2 = 337,0,0
dring2context =
Hi,
I am trying to recice a fax with * using SpanDSP - but it doesn't create the output
file. (See the bottom of log file).
* Loads both app_rxfax.so and app_txfax.so fine.
Also I can't make * autodetect an incomming fax call (yes I have enabled
faxdetect=both in zapata.conf - though it's not
Michael Hamann wrote:
Hi Everybody,
as a relative newby I´m just trying to get a Draytek Vigor Router (2600Vi)
connected to my Asterisk System (CVS-05/31/04). With X-Lite and a Cisco
Phone it is no problem, but the Vigor seems to have some problems with
Asterisk.
The first thing ist when I do a
I can't compile Asterisk on a Debian machine.
What is wrong? :/
debian... :-(
I was only able to compile asterisk when I gave up on doing it by myself
and decided to use the debian package (.deb).
I've got Asterisk CVS running on at least 8 Debian machines - most current
at Testing level -
Hello
I have the following structure
SIPH323 (chan_h323)
SIP Phone Asterisk/H323
---
Hi,
i want to load the cdr into oracle using unixODBC.
I'm using RH 9 2.4.20-30.9smp, unixODBC 2.2.6, easysoft odbc driver for oracle 1.3.1.
My unixODBC is working well.
With isql i can connect to the database, do selects, inserts and so on.
I created the table cdr as described on the asterisk
Hi everybody,
any hints when the next version of bri-stuff will be released so that it
will work with the current CVS head? (Klaus-Peter? ;-) )
Regards
Julian Pawlowski
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Jun 17, 2004, at 10:18 PM, Brian K. West wrote:
g726 is 16,24,32 and 48k asterisk only does g726-32k. The iaxy
doesn't do
g726 it does ADPCM as g726 is too complex for the iaxy to do.
So in this case g711ulaw/alaw is all you have to choose from.
Okay, that's what it looked like. So the IAXy
On Jun 16, 2004, at 4:05 PM, Michael George wrote:
Following the installation directions on the wiki, I got festival
built and installed. However, when I hit it from my dialplan, I get:
Feature Token_Method not defined
I found only one reference to this error message in the archives and
there
Holger Schurig wrote:
i'm new to asterisk and am having trouble placing outbound calls. i
Bug Grandstream so that they finally fix their buggy software.
The GS phone sends occassional SIP packets to port 0, not to port 5060, as
tcpdump or (better) ethereal will show you.
There's a page on
for today we only have experience with BRI applications together with asterisk.
is the following scenario possible and stable enough for production?
FYI : We want to build a unified messaging application integrated with SIP.
We have an E1 connection in Belgium with 100 msn's
We would think
On Friday 18 June 2004 02:46, George Pajari wrote:
(b) other times we would experience no audio in one direction for between 1
and 4 seconds and then things would seem to work fine;
I just had this problem with my * setup:
KSU - Adit600 - T100P - IAX2(Office) - IAX2(Colo) - IAX2(Nufone)
The
We would think about having 2 servers :
Server A : Asterisk
PRI card (Digium TE410P)
Server B : Fax server
PRI card (Eicon PRI30M)
Call --- TE410P/1 --- Asterisk Extension ---
Voice ? --- Voicemail or Dial
Fax ?--- TE410P/2 crossover
i'dd like to but is it stable enough for production (receiving over 500 faxes a day ?)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: vrijdag 18 juni 2004 13:58
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] TE410P / Eicon
When we enabled jitterbuffer the sound quality seemed to improve but we
noticed some problems:
(a) sometime we would get only one-way audio;
(b) other times we would experience no audio in one direction for between 1
and 4 seconds and then things would seem to work fine;
(c) some times
On Fri, 18 Jun 2004, Rich Adamson wrote:
A google search of the asterisk-cvs list indicates there has been several
iax changes in the last several months. Iax2 with gsm is working very well
between * systems using the current cvs Head.
I was told specifically by Mark to include
All,
Experiencing some issues on my FXO lines. If a call comes in
on an FXO and then get transferred to another FXO (say to call someones cell phone), those two lines will stay tied
together indefinitely. This happens to us when we transfer an incoming call to
our on call guys after
I just saw that one of our techs posted the same
question - I apologize for the multiple
posts (as I put my asbestos suit on J ).
Greg
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
219-462-7200
Tim,
busydetect=yes
callprogress=yes
Set these to no and it should stop the random hang-ups.
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
219-462-7200
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Schlie
Sent:
I had a bit of a problem compiling CVS Asterisk on Debian-Woody, but
www.voip-info.org has a debian-specific page that lists the debian
packages you will need to apt-get:
http://voip-info.org/wiki-Linux+Debian
...after installing these, it compiled without a hitch!
Also, make sure you have the kernel-headers package that matches your
kernel-source package.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
The base problem, I presume is not that there is no documentation, but how
to combine all those defacto standards, from an user and an application
point of view.
An Active Directory implementation in Linux (for users and application)
for me starts with the standard PAM/NSS stuff but why not extend
i'dd like to but is it stable enough for production (receiving over 500 faxes a day
?)
i think it is. at least i know someone who is using it in production on
a Digium E1 card.
If everything else fails you can buy that eicon card later on in the
worst case.
best regards
Klaus
--
Hi,
you can integrate it via PRI or BRI.
Regards
Felix
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo
Sent: Friday, June 11, 2004 7:04 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Integration with
Experiencing some issues on my FXO lines. If a call comes in on an FXO and then get
transferred to another FXO (say to call someones cell
phone), those two lines will stay tied together indefinitely. This happens to us
when we
transfer an incoming call to our on call guys after
hours and on
or a lot easier:
Pull the patch i use for my cvs snapshot Debian packages:
http://loke.home.marlow.dk/dists/sid/asterisk/patches/01-debian-marlow.diff
Apply it to latest cvs.
chmod +x debian/rules
And compile.
Have fun.
Kind regards,
Martin List-Petersen
martin (at) list (dash) petersen (dot)
Billy, looking at this more closely, I have some questions...
On Jun 15, 2004, at 9:45 PM, Billy Huddleston wrote:
Yes, I use it. Here's a sample extension of how to use it.
exten = 1234,1,Answer()
exten = 1234,2,MailboxExists(1234)
exten = 1234,3,Dial(SIP/1234,20) ; Try to ring for 20 seconds, no
On Thu, 2004-06-17 at 09:11, Klaus-Peter Junghanns wrote:
Hi,
Actually, you the Geode CPU mentioned below is a 5x86 (486 platform) at
233 MHz. If you take Pebble (http://www.nycwireless.net/pebble/), which
is a downstripped Debian ( 64 MB) on a readonly ext2 filesystem, you
should be
All,
I'm having trouble getting the X100P working.
Lsmod shows :
zaptel179808 0
I did a .
# modprobe zaptel
and here is my zaptel.conf (comments omitted)
__SNIP__
fxsks=1
loadzone = us
defaultzone=us
__SNIP__
Here is zapata.conf
__SNIP__
[trunkgroups]
[channels]
On Fri, 2004-06-18 at 14:57, Adam Lewis wrote:
All,
I'm having trouble getting the X100P working.
Lsmod shows :
zaptel179808 0
I did a .
# modprobe zaptel
and here is my zaptel.conf (comments omitted)
__SNIP__
fxsks=1
loadzone = us
defaultzone=us
Don't you need a 'modprobe wcfxs' also?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adam Lewis
Sent: 18 June 2004 14:57
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problems with X100P
All,
I'm having trouble getting the X100P working.
Hi!
I trying to configure * in a way, that it uses a different CLIP (Caller-Id
in ISDN) per SIP user, when relaying the call from SIP to the ISDN. So far
always the main (1st) number of the number-block is sent to the ISDN.
I have a E100P from Digium and use the zapata stuff (chan_zap).
All SIP
That did it. Thanks!
Adam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Williams
Sent: Friday, June 18, 2004 10:08 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with X100P
At 09:57 18/06/2004 -0400, you wrote:
I did a .
#
I am trying to connect directly to ATT VoIP service
CallVanage. I have ATTs ATA (D-Link DVG-1120M). They use mgcp. I have
traces of the connects from the Dlink and hoping to setup Asterisk the same.
It looks like I need to have Asterisk be a MGCP endpoint (gateway). How do I
configure
I have multiple voiceage g729 licenses installed on a RH9 box, and have
a remote C7960 configured to use it (low bandwidth). In calls like:
Remote C7960 - g729 - asterisk - g711 - C7960
the audio is oftentimes rather choppy. Changing the remote 7960 to use
g711 seems to eliminate/reduce the
On Fri, 2004-06-18 at 15:16, Bernie Hoeneisen wrote:
Hi!
I trying to configure * in a way, that it uses a different CLIP (Caller-Id
in ISDN) per SIP user, when relaying the call from SIP to the ISDN. So far
always the main (1st) number of the number-block is sent to the ISDN.
I have a
Hi
Does anybody if the X100P works in Switzerland? We can't get a line to PSTN.
When I run zttool it shows me always a red alert. I can make and receive calls with an
anlog phone plugged in the phone connector.
I've compiled and configured the card according to the wiki. Everything seemed to be
On Thu, Jun 17, 2004 at 05:02:26PM -0500, Erick Perez wrote:
10 analog extension using conventional phones (lets say Panasonic kx-ts3
analog)
4 analog lines coming from our telco
So i will need 3 TDM40B (total 12 FXS and none FXO so i can have 2 extra FXS
ports for future)
and one TDM04B
Michael,
From the docs, it looks like MailboxExists() will add 101 to the
priority if the box *does* exist and goes to the next priority if not.
I think the show application mailboxexists documentation is wrong. I
believe it's the other way around. It does exits? Jump to next
priority. It
Gonzalo Gasca wrote:
Create the profile
And a new windows appears:
Profile name
File name
Profile type Calls through SIP proxy
Then in SIP proxy,
click the sip proxy option
enter the Ip address of the proxy domain port
user domain
and proxy for nat and also the port (5060)
be sure u
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Holger Schurig wrote:
|I should follow this up to accurately state that audio was not
|operational in my test calls from the PDA. I have patched the
|iaxclient library with the changes available from ZiaxPhone that word
|align the IAX2 library on the
If you would rather use HylaFAX instead of spandsp and have $10K to
throw around, then may I suggest hiring an Asterisk channel author to
write a T.38-supporting channel driver? That way you could just use
t38modem with HylaFAX, and you wouldn't need all the duplicate hardware.
Lee.
On
We're using Cisco phones running skinny protocol.
When I call other extensions I don't get a ringtone,
although the remote end does ring and when answered we get clear two way audio.
When I call a queue from a skinny phone then I don't
hear the announcements.
Likewise we don't hear
Which voicemail is current and latest?
Voicemail
or
Voicemail2
I thot it was voicemail2 but this link sort of indicates otherwise...at
the bottome of the page it says:
Old version:
. Asterisk cmd VoiceMail2
http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMail2#comments
--
respectfully,
better send the EUR 10k (not $10k... :) ) to the author of spandDSP.
Nobody needs HylaFAX for receiving faxes. Converting a tiff to pdf and
storing it somewhere is not rocket science. ;)
best regards
Klaus
Am Fr, 2004-06-18 um 17.08 schrieb Lee Howard:
If you would rather use HylaFAX instead
Well I'm slowly learning my way around asterisk although as yet I
haven't had the chance to actually hook the system up to an ISDN line.
I am going to migrate from an Argent Office setup. My only problem is
keeping costs down on the phones.
The Argent system is running about 30 POTS phones. Can
On Friday 18 June 2004 11:08, Lee Howard wrote:
If you would rather use HylaFAX instead of spandsp and have $10K to
throw around, then may I suggest hiring an Asterisk channel author to
write a T.38-supporting channel driver? That way you could just use
t38modem with HylaFAX, and you wouldn't
T. Chan wrote:
Jeremy,
Yes, I felt that it was important to report my trouble and I did it three
times, reporting to the asterisk community, but for some reasons, I was not
being responded to at all. I thought my messages were embedded among the
hundreds of them and were missed out or everyone was
What does your sip.conf look like? Always make sure that you have the
following codec order for G.729 pass-thru:
[general]
disallow=all
allow=g729
allow=ulaw
allow=alaw
you don't need to force your C7960 (SIP settings) to use G.729 with the
above config.
see also:
You don't even need spandsp - fax is dead, remember? ;-)
-d
- Original Message -
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 18, 2004 11:10 AM
Subject: Re: [Asterisk-Users] TE410P / Eicon PRI
better send the EUR 10k (not $10k... :) ) to the
On Friday 18 June 2004 11:10, Klaus-Peter Junghanns wrote:
better send the EUR 10k (not $10k... :) ) to the author of spandDSP.
Nobody needs HylaFAX for receiving faxes. Converting a tiff to pdf and
storing it somewhere is not rocket science. ;)
Incorrect. I've been unable to get spandsp
Anyway, it appears as though the two contexts you have listed below have
the exact same name in-internal,
sorry, my error in anonymizing the stuff. the dupe is not in
the real config.
randy
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
If you're already using POTS phones and want the flexibility of SIP you may
just want to get SIP adapters that you can continue to use your POTS phones
with. I recommend the Sipura SPA-2000 dual analog adapter(www.sipura.com).
You can get them for about $92(if you get more than 10 in one order)
On Jun 18, 2004, at 8:56 AM, Randy Bush wrote:
Err, it works for me, with a 7940 and 6.3. I've never bothered with
'NewCall' or 'Dial'; you can get around them if you can set up a
decent
dialplan.xml.
aha. ok. thanks. on to sorting out a dialplan.xml. any simple
one that sez just give it all
I am running asterisk on an old PowerComputing Mac clone running
YellowDog 3.0 (Red Hat clone for PowerMacs) I've decided to try adding
a generic winmodem card and compile zaptel-0.9.1 for it.
First I tried to just unpack zaptel archive and do make clean; make
install. Compiled fine, but
Hi,
We used 512meg compact flash running debian.
John Bittner
Simlab.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
W. Kevin Hunt
Sent: Thursday, June 17, 2004 8:54 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Soekris
We're thinking of doing the same with our argent office system at the
moment.
The Argent system is running about 30 POTS phones. Can someone suggest
the cheapest option? Should I get some kind of large scale FXS box or
would the cost of doing that on a large scale work out the same as
getting
Am Fr, 2004-06-18 um 17.53 schrieb Darren Nickerson:
You don't even need spandsp - fax is dead, remember? ;-)
Why do YOU sell hylafax servers then? ;)
best regards
Klaus
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi,
just found out about the great lingo.com service offerings.
Could this be used with Asterisk? I have a couple of Sipuras on the LAN
and would like to use * to route this to Lingo or my POTS adapter.
People report that Lingo is using SIP although they say it can only be
used with their ATA.
Andrew Kohlsmith wrote:
On Friday 18 June 2004 11:10, Klaus-Peter Junghanns wrote:
better send the EUR 10k (not $10k... :) ) to the author of spandDSP.
Nobody needs HylaFAX for receiving faxes. Converting a tiff to pdf and
storing it somewhere is not rocket science. ;)
Incorrect. I've
By reading the Wiki's I found out that an Asterisk server with many (1)
extensions and/or SIP users can become slow when reloading. But what happens when you
also have many contexts in extensions.conf? More precisely, one context for each SIP
user?
I need this because I will have users
my current, inherited, dialplan.xml is
DIALTEMPLATE
TEMPLATE MATCH=00,1.. Timeout=0 User=Phone /
TEMPLATE MATCH=00,* Timeout=5 User=Phone /
TEMPLATE MATCH=* Timeout=5 User=Phone /
/DIALTEMPLATE
the last of the three entries would seem to be the
Hi Philip,
Unfortunately, * speaks MGCP only as the Call Agent, rather
than as the Media Gateway. MGCP is a master/slave protocol,
and it would take some effort to make * work as the slave.
I have the same problem: Free Telecom here in Paris includes
MGCP service with their DSL. You can call
On 2004.06.18 08:34 Andrew Kohlsmith wrote:
On Friday 18 June 2004 11:08, Lee Howard wrote:
If you would rather use HylaFAX instead of spandsp and have $10K to
throw around, then may I suggest hiring an Asterisk channel author
to
write a T.38-supporting channel driver? That way you could just
Jeremy
I did not report that to the bug tracker, I did not even think that was a
bug, I just thought may be I did something wrong, and I reported my problem
3 times to this mailing list, trying to get some light to my problem, I did
not get any response.
This time, at least I got some response,
Why not use mysql as it should be faster I'd suspect
-Original Message-
From: Manuel Wenger [mailto:[EMAIL PROTECTED]
Sent: 18 June 2004 5:43 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Thousands of contexts?
By reading the Wiki's I found out that an Asterisk server with many
Hi,
I had a similar problem for a while in Ireland.
Eventually after much hair tearing I decided it must
be something to do with the phone socket and commenced
to make a direct conenction between the twisted pair
and the X100P socket. Low and behold it worked.
After more mucking around I found I
Klaus-Peter Junghanns wrote:
Am Fr, 2004-06-18 um 17.53 schrieb Darren Nickerson:
You don't even need spandsp - fax is dead, remember? ;-)
Why do YOU sell hylafax servers then? ;)
best regards
Klaus
Working with the dead never stopped undertakers making a living :-)
Regards,
Steve
On 2004.06.18 08:10 Klaus-Peter Junghanns wrote:
better send the EUR 10k (not $10k... :) ) to the author of spandDSP.
Nobody needs HylaFAX for receiving faxes.
Firstly, I'm not just talking about receiving faxes.
If my choices are between HylaFAX and spandsp and if I want outbound
queueing and a
On Friday 18 June 2004 12:37, Steve Underwood wrote:
The segfaults I have followed up on have all been due to libtiff
versions. Are you sure there isn't some other version of libtiff lurking
on your machine? If there isn't I would like to follow up with you and
find why this happens. Many
On Fri, 2004-06-18 at 11:06, Artur Jasowicz wrote:
I am running asterisk on an old PowerComputing Mac clone running
YellowDog 3.0 (Red Hat clone for PowerMacs) I've decided to try adding
a generic winmodem card and compile zaptel-0.9.1 for it.
First I tried to just unpack zaptel archive
- Original Message -
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 18, 2004 12:03 PM
Subject: Re: [Asterisk-Users] TE410P / Eicon PRI
Am Fr, 2004-06-18 um 17.53 schrieb Darren Nickerson:
You don't even need spandsp - fax is dead, remember? ;-)
Manuel Wenger wrote:
By reading the Wiki's I found out that an Asterisk server with many (1)
extensions and/or SIP users can become slow when reloading. But what happens when you
also have many contexts in extensions.conf? More precisely, one context for each SIP user?
I need this because I
Manuel Wenger [EMAIL PROTECTED] wrote:
By reading the Wiki's I found out that an Asterisk server with many
(1) extensions and/or SIP users can become slow when reloading. But
what happens when you also have many contexts in extensions.conf? More
precisely, one context for each SIP user?
On 6/18/04 11:27 AM, Andreas Schiffler [EMAIL PROTECTED] wrote:
Hi,
just found out about the great lingo.com service offerings.
Could this be used with Asterisk? I have a couple of Sipuras on the LAN
and would like to use * to route this to Lingo or my POTS adapter.
People report that
David J Carter [EMAIL PROTECTED] wrote:
Don't you need a 'modprobe wcfxs' also?
Not for an FXO device, such as the X100P, no.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
_/ _/_/ _/ _/ _/_/
I have 2 Grandstream HT-286 devices and an Asterisk server. The *
Server is not using NAT and has port 5060 opened up. One HT-286 is
using traditional NAT and the other HT-286 is behind a residential DSL
router/firewall. I have the HT-286 setup as the DMZ Host in the
router/firewall so that
On Fri, 2004-06-18 at 13:03, Randy Bush wrote:
if i go off hook and dial 666 from an internal sipura spa-x000
(at extn 141), it rings straight through to extn 666.
using the same dialplan, from a cisco 7960 with 7.1 sip code
(at extn 142), i have to
go off hook
hit NewCall
punch
On Friday 18 June 2004 13:20, Lee Howard wrote:
Well, if you don't like t38modem, then a really cool thing would be if
you wrote a T.38 driver for HylaFAX also. So then Asterisk and HylaFAX
could play together without t38modem, without the AT command-response
language limitations.
I wasn't
hello
i have a cisco 924 router (its a router with a cable modem
interface, ethernet interface hublet, two pots jacks/fxs and one
pstn jack/fxo). i am not using the cable modem interface. i
merely want to use it as an ata device, possibly just a fxs if
thats all that can be done.
as some may
Am Fr, 2004-06-18 um 19.56 schrieb Lee Howard:
Firstly, I'm not just talking about receiving faxes.
If my choices are between HylaFAX and spandsp and if I want outbound
queueing and a client-server interface for networked usage, then
spandsp will not cut it alone.
So yes, anyone who
Is there any reason you can't use the callerid=name number in sip.conf
instead of a ton of contexts to do this?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Bond
Sent: Friday, June 18, 2004 1:47 PM
To: [EMAIL PROTECTED]
Subject: RE:
Just in case anyone else is looking for this information, I'm posting the
answer here. I do need to double check if this works with asterisk = 0.9.0.
I've heard rumors that IAX1 support has been removed in newer versions.
-- Forwarded Message --
Subject: Re:
On Fri, 2004-06-18 at 12:46, Chris Bond wrote:
Why not use mysql as it should be faster I'd suspect
I doubt it would be faster as asterisk will keep it all in memory, only
changes might be slowed.
But the thought is correct, use a database to store the data and one
context that does a lookup
Folks,
Randomly, when the phone is taken off-hook, the the Iaxy produces a
irritating banshee scream as opposed to a dial-tone. Cycling the power
fixes the issue, sometimes it magically goes away by itself.
Has anyone experienced this issue potentially fixed it?
I'm using asterisk CVS
On Jun 18, 2004, at 10:57 AM, Jeremy Jones wrote:
From the docs, it looks like MailboxExists() will add 101 to the
priority if the box *does* exist and goes to the next priority if not.
I think the show application mailboxexists documentation is wrong. I
believe it's the other way around. It
1 - 100 of 159 matches
Mail list logo
