On Thu, 2004-05-20 at 01:05, Jay Milk wrote:
Good call -- write cycle life of 10^3-10^4 are probably not much of an
issue in a digital camera, but would probably die quickly if used as a
HD replacement. Linux would have to run w/o swap. CFII+ harddrive?
You're talking about a microdrive,
On Wed, May 19, 2004 at 05:06:09PM -0500, Steven Critchfield wrote:
On Wed, 2004-05-19 at 15:12, Jay Milk wrote:
Since this is related... Does anyone have Asterisk working on a
Flash-drive? I was considering this as an alternative to having a
harddrive in my machine, thus keeping down
Hi,
-Original Message-
Please, don't store your dynamic sound files (voicemail) on CF. You
can only write to CF so many times, so the card may start suffering
failures. You can easily add a small harddrive or more
memory and ramdisk software.
If people really want to use CF,
Yes, I've read and implemented all the stuff on IAX. It's the local SIP
connection and its RTP streams that's the problem. For instance I noted
the strange timestamp behaviour from * on local traffic earlier.
Iain
--On Tuesday, May 18, 2004 1:56 pm -0600 Rich Adamson
[EMAIL PROTECTED]
--On Tuesday, May 18, 2004 1:42 pm -0500 Nik Martin
[EMAIL PROTECTED] wrote:
Out of context, this isn't much information. Is your network connection
OK?
Yes, AFAIK - I'm running all the traffic shaping / prioritisation stuff
mentioned on the list
Is your broadband provider having troubles?
--On Tuesday, May 18, 2004 1:43 pm -0500 brian [EMAIL PROTECTED] wrote:
Strange I do 7960 = * = IAX all day long without one jitter or any bad
audio. Now if both ends are NOT running the very latest(within the last
month or so) CVS-head for example if you have say a 2 month old
chan_iax2.c on
Hi,
IF i use a sip softphone or a iax softphone with
asterisk, i get a lag of about 1 second.
The two phones were on 2 different pc's near me.
When I speak on one, i hear it on the other after about 1 second.
I tried using iaxComm, Xten Xlite, etc.
Same.
FYI: The codec used was GSM.
Using
I have ethereal installed and I'll do a full call trace. The Catch 22 is I
don't have access to access to a source of repeatable (ie recorded) content
accessed through IAX. That would help in producing traces for the ATA and
7960 for comparison. I mainly use IAX for non-critical
This is normal for all VoIP communication there is nothing to wory about
and the lag is not heard in normal use.
Jason
At 13:50 20/05/2004 +0530, you wrote:
Hi,
IF i use a sip softphone or a iax softphone
with asterisk, i get a lag of about 1 second.
The two phones were on 2 different pc's near
Hi!
Ineed to pass the call duration and Bill Sec after a successfull
call to an AGI script. Is there a way to do this ?
- check out asterisk-addons from CVS
- enable the CFLAGS+=-DMYSQL_LOGUNIQUEID in the Makefile
- catch the unique ID of the call and pass it along to your AGI script
- let
Hi!
I am trying to find remote call forwarding feature in asterisk. I don't know
is it possible or any one had already done it.
The Wiki is your friend:
http://www.voip-info.org/wiki-Asterisk+call+forwarding
Cheers, Philipp
___
Asterisk-Users
Hi!
Should be
mailbox = [EMAIL PROTECTED]
Watch out - don't confuse an extension.conf context with a
voicemail.conf context! Go to /var/spool/asterisk/voicemail
and check the names of the directories (=voicemail contexts)
present.
Cheers, Philipp
Hi All,
I decided to have a go at installing Asterisk on FC2 which now runs on
Kernel 2.6..
Unfortunately I didn't get very far..
When trying to build zaptel it required me to link /usr/scr/linux-2.6 to
the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess
thats still the RH
I have a 7940 running 6.3 SIP firmware and make the following type of
calls:-
7940 = * = IAX2 = * = Digium X100P = Nortel CICS Analog FXS
Both local and remote asterisk's run CVS-02/24/04 (built about 30mins
apart). The IAX2 connection is over a VPN, and both sites are running
1500k/256k ADSL
Hi!
Should've been more clear, what I was referring to was the ability to select
the first or last message as a starting point when reviewing vm's.
You are referring to (4)(4) -- first msg and (6)(6) -- last msg.
However I don't think this plan made it into reality.
Cheers, Philipp
Does anyone have any experience with making DISA work?
In my extensions.conf I have the line:
exten = 11/2,1,DISA,|dialout
The way I understood it to work is if a call comes in to 11
from 22 then if is pressed it goes to dialout, else it moves on.
Hi!
I am trying to dial a mgcp extention from my sip phone and i am getting this
error message. anyone got any idea?
Do a mgcp show endpoints at the CLI and watch the output.
May 19 22:30:01 NOTICE[1251156800]: chan_mgcp.c:1104 find_subchannel: Gateway
'10.0.1.150' (and thus its endpoint
Hi!
A question: is there any way to get * to answer certain DTMF sequences
entered on an extension with a stutter tone?
Record the stutter tone in a .wav or .gsm file and use Playback() or
Background() to deliver it to the user.
See also:
http://www.voip-info.org/wiki-CLASS
Cheers, Philipp
Philipp von Klitzing wrote:
Not sure if also the csv version of the CDR records does include the
unique id.
It does, although not by default:
cdr_csv.c:
/* #define CSV_LOGUNIQUEID 1 */
/* #define CSV_LOGUSERFIELD 1 */
Apollon Koutlides
___
Hi.
(B
(BI want to connect Cisco7960 phone using IOS CME to Aasterisk VoiceMail system.
(BBut, DTMF relay is not work well, so I am able to hear VoiceMail intro and I am not
(Bable to contorl Asterisk VoceMail system.
(BDid anyone connect CME to Asterisk VoiceMail system?
(B
(BAsterisk
Dear all
I am just getting started with AGI
so I wrote the following script as a simple test
but all that happens is silence before it times out and hangs up
can someone help to get me started?
yet if i use the agi-test.agi script everything works I don't see the
difference
Thanks
php -q
?php
'SAY NUMBER 123 #* isn't a valid * command - it should be saynumber
followed by a valid string of arguments. Do a show application saynumber
in *.
Iain
--On Thursday, May 20, 2004 7:14 am -0400 Jer [EMAIL PROTECTED] wrote:
Dear all
I am just getting started with AGI
so I wrote the following
Hi Jay,
I am working on this. I am using a 256MB CF. I will keep you informed.
Daniel
Jay Milk wrote:
Since this is related... Does anyone have Asterisk working on a
Flash-drive? I was considering this as an alternative to having a
harddrive in my machine, thus keeping down noise and
snip
But as I've mentioned before, this isn't the whole story. There are other
repeatable scenarios that still cause problems, and to which some large
progressive providers also see as an issue and won't accept termination becuase
of it:
GW - SIP - * - IAX2 - * - SIP - 79X0
Now, if
Iain Stevenson wrote:
'SAY NUMBER 123 #* isn't a valid * command - it should be saynumber
followed by a valid string of arguments. Do a show application
saynumber in *.
In the meantime, you might as well try a show agi yourself :-)
Apollon Koutlides
Running asterisk on gentoo 2004.0, kernel 2.6.5, 2.8 Ghz hyperthreading
CPU 1G RAM. I decided to use kernel 2.6 after reading about problems
with hyperthreading and asterisk in 2.4 on this list. So far I've only
connected to VOIP service providers and everything has been working very
well. I
Hello,
We have been running a system based around the quadbri card from
www.junghanns.net for around 3 weeks now. For the first two weeks
everything was stable and ran well. In the last week a issue has
appeared, described below:
Someone attempts to call us, * sees an incoming call (it is
Also, since installing 0.0.2 we see this occasionally in the logs:
May 19 18:59:26 WARNING[16400]: Ring requested on channel 1 already in
use on span 1. Hanging up owner.
May 19 18:59:32 WARNING[16400]: Ring requested on channel 2 already in
use on span 1. Hanging up owner.
May 19 19:00:10
On Thu, 20 May 2004, Iain Stevenson wrote:
The Catch 22 is I don't have access to access to a source of repeatable
(ie recorded) content accessed through IAX. That would help in
producing traces for the ATA and 7960 for comparison.
The payload (i.e. audio) of the RTP stream is not relevant,
Hi,
I try to use IConnect on my MultiTech MVP 200 VoIP Gateway
and didnt workL.
I try thru my asterisk box and everything works fine
The MVP200 is behind the Nat and my * is connected directly
on Internet exactly like IConnect.
Thanks in advance for any
help.
Chris HARIGA
I'm currently doing it successfully using the Monitor command. There's
a really good example on the Wiki (www.voipinfo.org) (just search for
'Monitor'). After following the Wiki example, I then use the
SetCDRUserField to store the recording filename in the CDR (using MySQL)
so I can display it
have managed to establish voicemail functionality using voicemail /
voicemailmain applications
the documentation on these applications from digium.com suggests that
voicemail greetings are customizable (as one would be expect), but am not
able to find any supporting documentation
can anyone
OK, but I have AGI working and you don't - so please allow me the error
since it's a while since I worked on this, Of course, it would help if *
used consistent syntax for identical commands in extensions.conf and AGI,
but that's another debate.
Why not check the logs for php and * and post
Hi!
have managed to establish voicemail functionality using voicemail /
voicemailmain applications
the documentation on these applications from digium.com suggests that
voicemail greetings are customizable (as one would be expect), but am not
able to find any supporting documentation
does any one has that can give it to me RADIUS acc module for Asterisk. Im
realy interested on an account an billing manager.
Best regards
Hekuran Doli
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
The Catch 22 is I don't have access to access to a source of repeatable
(ie recorded) content accessed through IAX. That would help in
producing traces for the ATA and 7960 for comparison.
The payload (i.e. audio) of the RTP stream is not relevant, at least in my
experience. All the
Hi!
I am just getting started with AGI
so I wrote the following script as a simple test
but all that happens is silence before it times out and hangs up
can someone help to get me started?
Look here:
http://www.voip-info.org/wiki-Asterisk+AGI+php
php -q
?php
fputs(STDOUT 'SAY
At 08:47 AM 5/20/2004, you wrote:
-vvvc mode I see
*CLI
-- Executing Wait(Phone/phone0, 1) in new stack
-- Executing Answer(Phone/phone0, ) in new stack
-- Executing AGI(Phone/phone0, test.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
-- AGI Script
It is a royal pain in the butt to manually walk through 2,000 packets
calculating timestamp differences, inspecting sequence numbers, etc. I'm
in the process of writing a small app to read the ethereal packet capture
files and do that stuff on request.
Or simply import the trace in to a
As a test, I was trying to use Iaxcomm and Iaxphone to connect to
Asterisk and dial out to my other line. Using either of these soft
phones, I can connect to Asterisk and listed to audio just fine. I can
even connect across the net to another asterisk server and hear audio
just fine, however,
Hello!
I've been reading through the archives on this list for the last 8-10
months. There are some reports on success with tftp autoconfiguration
with a given cfg.txt format but really vague. Has anybody successfully
done this without using GAPS, or has anybody got a correctly formatted
www.bkw.org/~web/parse.txt
That should parse and show ALL lines where the timestamps slip.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ray Burkholder
Sent: Thursday, May 20, 2004 8:11 AM
To: [EMAIL PROTECTED]
Subject: Re:
Good call -- write cycle life of 10^3-10^4 are probably not much of an
issue in a digital camera, but would probably die quickly if used as a
HD replacement.
i have a cigarette-sized freebad box using only flash for disk and
swap. runs for years under load. a bunch of us use them, though
the
http://www.bkw.org/~brian/parse.txt
Its still early. :P
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of brian
Sent: Thursday, May 20, 2004 8:58 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] AArgh, * and the 7960
I think I have it configured properly. ztcfg -vv shows it as channel 1 and
zttool shows it as OK. But I can't dial out.
When I try, it shows it arrive in teh right stack, but then issues the
following errors:
channel.c:1676 ast_request: No channel type registered for '{PSTN-1}'
app_dial.c:554
Can anyone share any advice / documentation on how to configure a Tellabs
2572 T1 echo canceller? I connected one between a T100P and an Adtran
TA750 FXO/FXS channelbank, but when echo cancellation is active I get
a LOT of snap-crackle-pop (and other problems) on the line.
The 2572 has a bunch
On Thu, 2004-05-20 at 09:44, Pats1776 wrote:
channel.c:1676 ast_request: No channel type registered for '{PSTN-1}'
app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}'
= = Everyone is busy at this time
[to-pstn]
exten = _1NXXNXX,1,Dial(${PSTN-1}/${EXTEN})
exten =
Thanks for the syntax error fix, but I'm still having the same problem.
Funny thing was, I never caught that syntax error because so far I was only
trying with the preceding '1'.
I can't seem to find this error relating to the x100p cards via google, the
asterisk mailing list archives, or the
Hi, to all!!!
I can't download asterisk-addons...I try with CVS,
but i can't.
How can I do???
Thank you
Fabio
On Thu, 2004-05-20 at 05:12, WipeOut wrote:
When trying to build zaptel it required me to link /usr/scr/linux-2.6 to
the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess
thats still the RH infulence.. :)
After than I tried again but the page rolls with errors and finally
I just grabbed my fresh Fedora Core 2 final release.
Untared zaptel-0.9.1 dir make linux26 and I get errors on
the compile. Anyone else tried this yet and been sucessful?
I did have to ln -s /usr/src/linux-2.6.5-1.358 /usr/src/linux-2.6
But I still get errors after that... about
asm/linkage.h
Hello to all.
I have a couple of budgetones connected to Asterisk
server. I can establish calls using budgetone with no
problem, but when I hang up a Budgetone, Asterisk
does not detect the hangup. It seems that the
communication goes on in spite of budgetone's hangup.
My sip.conf:
[general]
Has anyone seen this before? This channel is consistently present on
both of my asterisk servers. Sometimes they disappear for a few seconds
and then come back. It always has the same Call ID.
voip1*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter
Jerry Geis wrote:
I just grabbed my fresh Fedora Core 2 final release.
Untared zaptel-0.9.1 dir make linux26 and I get errors on
the compile. Anyone else tried this yet and been sucessful?
I did have to ln -s /usr/src/linux-2.6.5-1.358 /usr/src/linux-2.6
But I still get errors after that... about
Post your zapata.conf and zaptel.conf
Nik
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pats1776
Sent: Thursday, May 20, 2004 9:45 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] x100p card + dailing out
I think I have it configured
What address is that? Is it a phone (or address of a PC with a softphone?)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Dolloff
Sent: Thursday, May 20, 2004 10:41 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Mystery SIP channels
Karl Brose wrote:
I think when you have this setup you need to keep the media path going
through Asterisk at all times.
Your SIP is binding to both ports, internal and external, but that
doesn't correctly set it up for either scenario, localnet calls and
external calls. It won't keep the
I had the same thing come up on mine when I was having codec issues
with one of my phones.
Kyle
Nik Martin wrote:
What address is that? Is it a phone (or address of a PC with a softphone?)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve
I need to connect a bunch of analog telephone sets. Does anyone have
any comments about this channel bank? Disconnect supervision? Echo?
ADSI problems? The price is right @ $995 new and $695 refurbished.
Thanks,
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
Hi there,
Here at my company we are willing to use the asterisk IVR system.
The problem we are having rigth now is that all our GWs use G729.
I've read that in order to asterisk be able to make transcoding from the GSM
audio files to G.729, it is necesary to purchase a license from
You might want to try removing the hyphen. It could be misinterpreting
it? Might want to try simplifying things a bit too for testing
purposes. Take out the PSTN-1 and put in the ZAP/1 directly into your
dial plan to verify that * can access the ZAP channel correctly.
-Original
I always get that with a softphone, but not with a hardphone.
Grandstream BT100 is only $70 so Im gonna get those for most of the
people. And a few higher end phones for the execs.
Kyle
Navnit Chachan wrote:
Hi,
IF i use a sip softphone or a iax softphone with asterisk, i get a lag
of about 1
Joshua M. Thompson wrote:
On Thu, 2004-05-20 at 05:12, WipeOut wrote:
When trying to build zaptel it required me to link /usr/scr/linux-2.6 to
the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess
thats still the RH infulence.. :)
After than I tried again but the page rolls
I don't actually know. All of the users are behind NAT, so the channel
list doesn't match the peers list.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nik Martin
Sent: Thursday, May 20, 2004 10:48 AM
To: [EMAIL PROTECTED]
Hi there,
Here at my company we are willing to use the asterisk IVR
system.
The problem we are having rigth now is that all our GWs use G729.
I've read that in order to asterisk be able to make transcoding from the
GSM
audio files to G.729, it is necesary to purchase a license
If you are in a hurry then you should call Digium
On Thu, 2004-05-20 at 10:58, pesb wrote:
Hi there,
Here at my company we are willing to use the asterisk IVR system.
The problem we are having rigth now is that all our GWs use G729.
I've read that in order to asterisk be able to
Hi,
Im playing with the CVS head time limiting at Dial application, it
just works fine but the only problem is that the caller isnt hearing
the warning message. Im using a Cisco 7960 as the caller and a Polycom
500 as the callee. The audio is passing through Asterisk:
-- Executing
Here you go:
zaptel.conf
fxsks=1
loadzone=us
defaultzone=us
zapata.conf
[channels]
language=en
echocancel=yes
echocancelwhenbridged=yes
context=from-pstn
signalling=fxs_ks
callerid=asreceived
channel=1
Scott
- Original Message -
From: Nik Martin [EMAIL PROTECTED]
To: [EMAIL
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Anon wrote:
| On Thursday 13 May 2004 11:57 pm, Jason A. Pattie wrote:
|
|-BEGIN PGP SIGNED MESSAGE-
|Hash: SHA1
|
|Steven Critchfield wrote:
|| On Tue, 2004-05-11 at 17:33, Jason A. Pattie wrote:
||Our problem ended up not being with Asterisk
This is discussed at length in the Wiki, on several pages, including:
http://www.voip-info.org/wiki-Asterisk+G.729+licensing
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason Williams wrote:
| This is normal for all VoIP communication there is nothing to wory about
| and the lag is not heard in normal use.
|
| Jason
|
| At 13:50 20/05/2004 +0530, you wrote:
|
| Hi,
| IF i use a sip softphone or a iax softphone with
WipeOut wrote:
Joshua M. Thompson wrote:
On Thu, 2004-05-20 at 05:12, WipeOut wrote:
When trying to build zaptel it required me to link
/usr/scr/linux-2.6 to the default source dir which is
/usr/src/linux-2.6.5-1.358.. I guess thats still the RH infulence.. :)
After than I tried again but
Hi,
- Original Message -
From: Tor Houghton
You can enable the key beep in DIAX, but what's the reason to get a DTMF
type of feedback?
The beep is not enough?
For some people, maybe? I just find it more natural to hear the DTMF when I
hit a number. It means that if I am dialling a
Yes this is correct, you need too purchase licenses, but the number of
licenses you buy mast be proportional to the size of the cpu's processor
you have.
Jorge
Hi there,
Here at my company we are willing to use the asterisk IVR
system.
The problem we are having rigth now is
On Thursday 20 May 2004 11:04, WipeOut wrote:
Joshua M. Thompson wrote:
You'll need to configure the source tree before zaptel will
compile. The config files are in
/usr/src/linux-2.6/configs...copy the one that matches what
you're running to /usr/src/linux-2.6/.config and then run make
On Thu, 2004-05-20 at 12:04, WipeOut wrote:
Thanks for the try but its didn't work.. Got exactly the same result..
Apparently the FC2 2.6.5 kernel has another issue, one that I didn't
start seeing until 2.6.6 (I build my own kernel RPMs.) There are a few
files that are auto generated by the
Is anybody else out there using VoicePulse Connect and having problems
this morning? I just noticed that they have absolutely no contact
information in their website.. just want to make sure I didn't break
something in my asterisk configs.
-fedl
___
Jeff Noxon wrote:
Can anyone share any advice / documentation on how to configure a Tellabs
2572 T1 echo canceller? I connected one between a T100P and an Adtran
TA750 FXO/FXS channelbank, but when echo cancellation is active I get
a LOT of snap-crackle-pop (and other problems) on the line.
The
It seems like it might be nice to have a mailing list to talk about (and
to) voip providers for Asterisk users.
It would be a good place to share info about config, pricing news,
customer service, local numbers, transient outages, etc. Providers would
be encouraged to contribute sales info.
I'm finding I can't run two festival commands in the same connection. Given
the following:
exten = 555,1,Answer
exten = 555,2,Wait(1)
exten = 555,3,Festival(mary had a little lamb)
exten = 555,4,Wait(1)
exten = 555,5,Festival(she also had a duck)
exten = 555,6,Hangup
Calling 555 gets the first
I removed the PSTN-1 variable reference and started referencing it as Zap/1
and also ZAP/1, without any difference - same errors.
I believe the hyphen you were talking about was the one in PSTN-1.
Scott
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Joshua M. Thompson wrote:
On Thu, 2004-05-20 at 12:04, WipeOut wrote:
Thanks for the try but its didn't work.. Got exactly the same result..
Apparently the FC2 2.6.5 kernel has another issue, one that I didn't
start seeing until 2.6.6 (I build my own kernel RPMs.) There are a few
files
On Thu, 20 May 2004 14:18:16 +0100, Andy Farnsworth [EMAIL PROTECTED]
wrote:
As a test, I was trying to use Iaxcomm and Iaxphone to connect to
Asterisk and dial out to my other line. Using either of these soft
phones, I can connect to Asterisk and listed to audio just fine. I can
even connect
Hi,
I've looked through the archives and seen references to placing calls on
hold on a snom 200 (any version of the firmware but we have the latest:
2.05e.)
Basically, we can't place calls on hold on the snom 200! The manual
talks about the Flash button (which is really the R button, as far as I
Does anyone have experience setting up * to accept anonymous
sip UAs and the dumping the call into IVR? Im thinking this would be a
good way to have customers call us without creating an extension. So for my tests
have been focused on providing internal functionality.
Thanks,
Chad
Maron Kristófersson wrote:
Running asterisk on gentoo 2004.0, kernel 2.6.5, 2.8 Ghz
hyperthreading CPU 1G RAM. I decided to use kernel 2.6 after reading
about problems with hyperthreading and asterisk in 2.4 on this list.
So far I've only connected to VOIP service providers and everything
has
I am having an issue with voicepulse also.
John Bittner
Simlab.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
C. Sullivan
Sent: Thursday, May 20, 2004 12:52 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] VoicePulse broken?
Is
I thought that is what the Asterisk-Biz mailing list was for.
http://lists.digium.com/mailman/listinfo/asterisk-biz
On Thu, 2004-05-20 at 12:04, Reed Wade wrote:
It seems like it might be nice to have a mailing list to talk about (and
to) voip providers for Asterisk users.
It would be a
Philipp,
I already have that call-forwarding feature set into asterisk.
What I am looking is how to set that feature remotely by calling into your
voicemail or any given no. so that person can set call-forwarding remotely.
Few of our sales people want this kind of feature, because if they are
At 1:04 PM -0400 on 5/20/04, Reed Wade wrote:
It seems like it might be nice to have a mailing list to talk about
(and to) voip providers for Asterisk users.
It would be a good place to share info about config, pricing news,
customer service, local numbers, transient outages, etc. Providers
Title: Avaya Partner Phones to SIP?
I remember someone posting here some time ago about commercial offerings for taking channel banks of Avaya partner phones and turning them into asterisk compatible (SIP?) devices, but I can't seem to find a reference to the hardware manufacturer or specific
Maron Kristófersson wrote:
Hello!
I've been reading through the archives on this list for the last 8-10
months. There are some reports on success with tftp autoconfiguration
with a given cfg.txt format but really vague. Has anybody successfully
done this without using GAPS, or has anybody got
On Thu, May 20, 2004 at 11:17:55AM -0500, brian said:
I've seen that licenses are purchased on a per-channel basis. Could we
make
some sort of agreement on having a no-limit channel license? Even, we
would
like to have the possibility of installing it on how many machines we wish
to
First, try moving back to 2.05c or earlier. 2.05e has a few problems
(remember, it's beta quality) that could be causing this. Second, are you
sure that the disconnect on hook or transfer on hook settings are the
way you expect them to be. That caught us for a while since we were putting
people on
Inbound is working here, no problems that I know of.
Scott
- Original Message -
From: C. Sullivan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, May 20, 2004 12:52 PM
Subject: [Asterisk-Users] VoicePulse broken?
Is anybody else out there using VoicePulse Connect and having
Chad Brown wrote:
Does anyone have experience setting up * to accept anonymous sip UAs and
the dumping the call into IVR? Im thinking this would be a good way to
have customers call us without creating an extension. So for my tests
have been focused on providing internal functionality.
Just
The Wiki is a bit wrong.. you can record raw g729 streams to disk, what do
you think format_g729.c is?
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Scott Stingel
Sent: Thursday, May 20, 2004 11:25 AM
To: [EMAIL PROTECTED]
HAHAH why do they ever work! Take this to the -biz list please!
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C. Sullivan
Sent: Thursday, May 20, 2004 11:52 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] VoicePulse
Hi all,
Unable to find translation path. How to fix ?
May 20 18:22:49 NOTICE[1224059824]: channel.c:1508
ast_set_read_format: Unable to find a path from G729A to ULAW
May 20 18:22:49 NOTICE[1224059824]: channel.c:1478
ast_set_write_format: Unable to find a path from ULAW to G729A
Matthew Branton wrote:
I remember someone posting here some time ago about commercial
offerings for taking channel banks of Avaya partner phones and turning
them into asterisk compatible (SIP?) devices, but I can't seem to find
a reference to the hardware manufacturer or specific experiences.
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