If you can plug it into a regular analog phone line and have it work,
then it will work with Asterisk.
On Wed, 2003-08-13 at 10:42, Chris Hale wrote:
Anyone know if the ATT 964/954 series phones have any issues with
Asterisk? We have 5 phones and would like to reuse them if possible.
Any
Hi.
I'm trying to do some custom call logging, and I want to call an AGI
script from a hangup handler to log call durations and things. Although
the script executes, it isn't retrieving variables from the AGI
interface. Looking closer, I realised the variables are actually getting
unset
I inquired to Grandstream about their resellers and they pointed me to
an establishment that never got back to me with a quote, even after
multiple reminders.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
James Sizemore
Sent: Wednesday, August
yes
On Wed, 13 Aug 2003, Chee Foong wrote:
Hi,
I manage to solve the problem. I just change the span configuration in
zaptel.conf to E1 configuration. Unload zaptel driver and load it again. It
seems to work fine.
I would like to know if RFC2833 is equavalent to out of band DTMF?
Foong
Are the VoiceAge people generally unpleasant to work with and geniunely
uncaring, or do they just fail to respond?
Matt Hardeman
PaperSoft
- Original Message -
From: Mark Spencer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 12, 2003 10:16 PM
Subject: Re:
Steve == Steve Underwood [EMAIL PROTECTED]:
Steve Kim C. Callis wrote:
I was reading on www.vovida.org/applications/downloads/G729A/ (home
of VOCAL) pages, and that there is a free license use for
non-commercial for G.729A. Is that usable under Asterisk or strictly
a Vovida offering?
I don't understand the reasoning here so could
somebody please help me out?
chan_h323 is causing a segmentation fault when
trying to connect a call.
I tracked the problem back to chan_h323.c in the
oh323_new() function.
the code is: tmp = ast_channel_alloc( 1
);
After this point, tmp-pvt
For PRI-*-fax over FXS
It's as simple as having the fax extension the the incoming context
associated with the PRI channels. With PRI channels we can hear the fax
before we even answer (in most cases)
regards
Martin
On Tue, 12 Aug 2003, Adams, Gavin wrote:
From: Martin Pycko [mailto:[EMAIL
Same thing. It will make sense to try
Register = FWDnum@fwd.pulver.com:FWDpass@fwdnat.pulver.com:5082
but in that case Asterisk sends
REGISTER sip:fwdnat.pulver.com SIP/2.0
which is not right. It should be sip:fwd.pulver.com but sent thru
fwdnat.pulver.com:5082
BR Borut
-Original
Thanks, I didn't realize that was the code for the codec. So, How do I know what codec
that code
translates to?
--
Paul
WipeOut . wrote:
At the console while a call is in progress run sip show channels and look in the
format column..
How can I tell what codec a SIP session is using?
--
On 12-Aug-03 Dave Cotton wrote:
What actually _read_ a manual, only wimps do that. Easier to ask someone
else. :)
He could try
*CLI help CR
But then you'd have to read.
Hi Prakash -
As a slightly more sympathetic new * user... here are a few things I've
discovered in the last 2
Chee Foong wrote:
Firewall shouldn't be a issue since the call works fine with ztdummy loaded.
I debug the chan_h323 and it uses the right codec G729 from digium.
H.323 does NOT deal with NAT or Firewalls without a smart edge device.
chan_h323 does not use ztdummy whatsoever, so that has no
Hi,
Quick question to all the electronics gurus out there..
I unpacked my second GS phone yesterday (had it for about a month!) and set it up..
This morning the power supply is dead..
I have looked for a new one online (In the UK using Maplin let me know if you know a
better place.) becasue
I've been taking another approach to this codec/bandwidth
problem. Instead of trying to get more codecs into
Asterisk (which is always hard due to licencing) I've been
trying to get vendors to implement GSM in their products.
SNOM do GSM.
D-Link gave me the good old, we have plans to
Title: ANI/DNIS call routing
Can someone in Asterisk'land subscribing to 800 service explain to me how to setup extension.conf to route calls based on the incoming DNIS/ANI. For example I want to route 3 incoming 800# coming across a trunk group to all land in the same queue. So I guess I am
On Tue, 2003-08-12 at 15:29, Steve Lane wrote:
Would the firewall pose a problem? I thought Asterisk had the solution
for working behind a firewall?
If is has it's one of the best kept secrets.
--
Dave Cotton [EMAIL PROTECTED]
___
Asterisk-Users
Can anyone explain why this is
happening?
I have a server attached to a phone line that will
play a .wav file, then play all the dtmf digits (after it answers the call). If
I place a call from a SIP device (like a Cisco 7960 phone) through Asterisk and
on to the test server, via PSTN, the
hi,
i have a external usrobotics modem, i want to use it with asterisk to
interact with the pstn,
what i have to do?
thanks,
--
santiago jos ruano rincn
administracin servidores y servicios de internet
red de datos
universidad del cauca
-BEGIN PGP MESSAGE-
Version: GnuPG v1.0.6
Generally speaking, something like this would require a regulated power
supply,
as the internal components are most likely needing 5VDC, instead of the
phone
taking over 5V and regulating the power itself (which drops the voltage in
the
process, hence taking over 5V...). I'm guessing that you
These list messages might be useful:
http://lists.digium.com/pipermail/asterisk-users/2003-June/013918.html
http://lists.digium.com/pipermail/asterisk-users/2003-June/013946.html
On Tue, 2003-08-12 at 13:22, Steve Lane wrote:
I am trying to do the same thing you are doing. I am new to asterisk
my point was your logic regarding 'calculating magic/metric' for
extremely long call times shouldn't be part of the 'logic' it
SHOULD be 'inhouse' policy where the mgr gives agentA a nice long
chat about how to sell/service the client more effectively.
i know there is at least one other out there
Try adding:
exten = fax,1,Dial(blah)
Where Blah is the zap or SIP port your fax machine is connected to.
On Mon, 2003-08-11 at 15:26, Eduardo Goncalves wrote:
On Mon, 11 Aug 2003 15:15:08 -0500
Tilghman Lesher [EMAIL PROTECTED] wrote:
On Monday 11 August 2003 02:46 pm, Eduardo Goncalves
Did you try:
register = FWDnum@fwdnat.pulver.com:FWDpass@fwd.pulver.com
On Tue, 2003-08-12 at 13:24, Borut Senicar wrote:
Hi All,
Is there any way to connect (register, initiate and receive calls) with
Asterisk to FWD through NAT? Since I own my router port forwarding is not a
problem.
Hi Mark,
Short of taking my board out of my * box is there any way to check what revision of
the TDM400P I have? It was purchased in May of this year. Is the pricing likely to be
the same or similar to the add-on FXS ports? Does this also mean that as we'd be able
to get away with not using
Yes, all does.
regards
Martin
On Mon, 11 Aug 2003, Michiel Betel wrote:
Simple Q but I can't find the answer in the archives (and am too lazy to
look in the source, but then its 32 Celcius here...
Do all digium cards provide the zapata timing? e.g. also the analogs
(including the X100P) or
On analog ports you need to Answer
Ringing
Wait,2
and then do something .
That should detect faxes.
regards
Martin
On Mon, 11 Aug 2003, Tilghman Lesher wrote:
On Monday 11 August 2003 03:26 pm, Eduardo Goncalves wrote:
On Mon, 11 Aug 2003 15:15:08 -0500
Tilghman Lesher [EMAIL
Does anyone have any info on weather or not GS plan to support some of the open
codecs, Mainly iLBC, GSM and Speex??
Having to use G.711 with Asterisk sucks.. its just too hungry on the bandwidth..
--
__
http://www.linuxmail.org/
Now with e-mail
We do the replacement adapters for £12+VAT if interested. You'll still need
the US-to-UK adapter though. Contact me offline if you need one.
You could go to cpc as was suggested. www.cpc.co.uk
Tan
telappliant.com
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL
I'll admit it. I'm a asterisk newbie (but no stranger to telephony).
The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone
specials) on a private segment calling to a Linux box acting as the
segment's firewall with a leg on our public network. The phones are
setup as SIP/phone1
At the console while a call is in progress run sip show channels and look in the
format column..
How can I tell what codec a SIP session is using?
--
Paul
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
So in other words... Asterisk can do what the Avaya Conversant can do if
you have a full understanding of it? Please pray tell how to do these
things. I am all ears.
Steve Lane
Vision Communications
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
On Sun, 2003-08-10 at 21:25, Andy Hester wrote:
Perhaps there is another way to cut down on increased traffic...
Specifically, I would go back to the suggestion of a collaborative website
for documentation. Collecting info and organizing into Howto's would reduce
the number of times people
Hi All,
Is there any way to connect (register, initiate and receive calls) with
Asterisk to FWD through NAT? Since I own my router port forwarding is not a
problem.
I tried with
Register = FWDnum@fwd.pulver.com:FWDpass@fwd.pulver.com
but since Asterisk still use internal IP in some SIP fields
Check out X-Lite. http://www.xten.com
Only the older (pre 5.x) versions of Messenger support
SIP. If u can get a copy of v4.7 that will work fine.
Instructions on setup can be easily found by doing a
search of this mailing list. (look for the keywork 'MSN').
I do however suggest X-Lite, for
what hassles?cvs update
Jeremy McNamara
Linus Surguy wrote:
Hi all,
We're using an older version of *, built a couple of months ago and before
we go through all the hassle of updating source files and checking latest
dependancies on other kernels etc, I'd like to know if the following is a
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armand A.
Verstappen
Sent: Sunday, August 10, 2003 5:29 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Iconnecthere
Hi Andrew,
On Sun, 2003-08-10 at 19:39, Andrew Joakimsen wrote:
-Original
All,
Has anyone had any thoughts/discussion on providing a malicious call trace feature
within Asterisk. Most legacy PBX's support this feature which allows a handset user to
indicate using DTMF during a call that it's a malicious call which instructs the PBX
to send a specific Q931 message
It works now ...sorry but it was my linux box ... I had Sip express router
installed on this machine :-\
So my ip phones loged on S.E.R and not on asterisk ;)
My voice mail works fine :)
Just a last question, if i configure G723 in my ATA, i can't call the
voicemail for exemple. I've seen
W:
checked the disk space and there is plenty of room
What sequence did you follow for debugging?
where does * put the E-mails before transmitting?
URiel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Simon
Woodhead
Sent: Saturday, August 09, 2003 4:02 PM
The chipset used in the X100P - at least the one I have - is designed for
the US/Japan market only. The reference design in the datasheet for the
chipset does not include facilities for the detection of line voltage
reversal. Hence the only way to detect caller ID sent before ringing would
Hi WipeOut,
softdtmf=1 will use asterisk's dsp functions to detect and
generate the DTMF tones. softdtmf=0 will use your capi controller
to do the detection/generation. unless you have an active
card i would suggest to use softdtmf=1.
early B3 connects will let you hear the inband call progress
Hi,
We have our * box configured to receive inbound SIP calls from FWD and
enter into an autoattendant where someone can enter an extension
directly.
However, the inbound DTMF is not being correctly detected in most
cases. Entering 8050 results in a correct detection, but enter 9003
results
Hi,
There is any Actiontec IPW owner on this list who can give me a
specification for the device drivers?
I mean just for the special functions like: ring, dial, tone, etc?
I intend to integrate it with the WinIAX client which for me works a lot
better than any other one.
Thanks,
Dan
You shouldn't run the gatekeeper and asterisk on the same machine.
Jeremy McNamara
Why is that?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Can someone find this?
On Sat, 9 Aug 2003, Dave Cotton wrote:
On Sat, 2003-08-09 at 15:07, Siggi Langauf wrote:
Well, I'd say: just strip the date and time off!. You made that call on
2003-08-08, 10:06h local time, didn't you?
My question was how to change the code.
As an experiment
On Sun, 2003-08-10 at 10:18, Andrew Joakimsen wrote:
On Sun, 2003-08-10 at 09:02, Andrew Joakimsen wrote:
Does anyone have Asterisk working with Iconnect here for incoming
and/or
outgoing calls?
have a look at:
http://www.loligo.com/asterisk/example-configs.2003-04-24/extensions.con
did you happen to run ztcfg after you setup the configs?
On Sat, 9 Aug 2003, Barry Porch wrote:
I am attempting to set up an Asterisk box which I am only concerned with
getting a single T1 working. I have this T1 connected to my PBX and I
am looking at using Asterisk as a conference bridge.
Quoting Paul Cheng:
Hi,
We have our * box configured to receive inbound SIP calls from FWD and
enter into an autoattendant where someone can enter an extension
directly.
However, the inbound DTMF is not being correctly detected in most
cases. Entering 8050 results in a correct detection,
Thank you. I am sorry. I've fixed the problem
Serge
From: Brian West [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Need help with installation of H323 chanel
driver
Date: Sat, 9 Aug 2003 20:19:54 -0500 (CDT)
Apparently you didn't read the
Well the code that actually blows up is the rawwriteformat assignment,
but yes the channel pvt struct somehow is getting destroyed or corrupted
somehow
Jeremy McNamara
John Fortman wrote:
I don't understand the reasoning here so could somebody please help me
out?
chan_h323 is causing a
Hi Bruce,
On Wed, 6 Aug 2003, Bruce Ferrell wrote:
[...]
Is there a difference between what asterisk is and a softswitch? Can
someone explain it in small words and phrases for me?
Sure, there is. First of all, * is much cheaper.
But technically, * does much more than a soft switch:
AIUI, a
try this:
http://www.loligo.com/asterisk/current/
Lubo
Adelino Baena wrote:
Dear Colleagues
I am a newbie on Asterisk and am having difficulties to find
documentation about how to configure the H323 and Sip services. Could
somebody of you have the kindness to send me functional samples of
Hi! I am new to Asterisk too, I got the similar problem and I would like to
know how to get * to work behind NAT.
When I have the SIP Debug turn on, I got the error 479 from FWD when * try
to register with FWD, it looks like * is using the local IP (192.168.x.x) in
the Contact field. I have
Dear Colleagues
I am a newbie on Asterisk and
am having difficulties to find documentation about how to configure the H323
and Sip services. Could somebody of you have the kindness to send me functional
samples of conf files to my personal e-mail ?
Im testing two VoIP clients: H.323
On Fri, 8 Aug 2003, Dave Cotton wrote:
The x100p does get the CID in France. It is now a question of how to break it down.
I changed callerid.c line 278 to :-
ast_log(LOG_NOTICE, Got this:- %s\n, cid-rawdata);
and the result on August 8 at 10:06 from 0490233081 was:-
File callerid.c,
Hi,
It seems that even the protocol is similar (for some European countries),
the frequency used for FSK modulation differ between US and Europe.
See the link for France:
http://matthieu.benoit.free.fr/freqcli.htm
They are:
Receiver FSKMin./Typ/.Max./Unit.
Transmission Rate
Sometimes you need to start asterisk from
/usr/src/
or /usr/src/asterisk/
and also in console mode for g729 code to work.
But I've heard that safe_asterisk also works.
regards
Martin
On 8 Aug 2003, Eric Wieling wrote:
I'm getting the following message when I start Asterisk:
WARNING[1024]:
Hello Michael,
Here is the BackTrace of the program which i forgot
to attach
BACKTRACE OF Asterisk -vvc
#0 0x42074d60 in _int_realloc () from
/lib/tls/libc.so.6
#1 0x420738c4 in realloc () from /lib/tls/libc.so.6
#2 0x47c7da89 in PAbstractArray::SetSize(int) () from
Mark,
if the capability for line reversal detection is in the hardware (X100P) then does
this mean that the detection of DTMF style caller-id as used in the following
countries would ber trivial? or am I hoping too much...
Finland, Denmark, Iceland, the Netherlands,India, Belgium, Sweden,
Can anyone else that is trying to get Agents Queues
into production shape report all issues to bugs.digium.com
use the new Project ACD so that we can see all issue
related to this new code
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Eric-
Thanks for answering. Maybe a Stupid Question. If you have two IAX
softphones connecting to the same destination, how does it manage it on the
receiving end. It seems like that would work great for point to point
connections, but if you have more than one session up (and I would think
Title: RE: [Asterisk-Users] queue / agent documentation
http://www.digium.com/asterisk_handbook/agentlogin_queues.html
And you should take a look at queues.conf for some comments detailing the various queue distribution algorithms, ringall, roundrobin, leastrecent so on so forth.
-Matt
Jim
check and make sure logger.conf logs debug to the log file
start * with -vvvgnc
get it to seg fault, find the name of the core.pid file like core.1234
cd /usr/src
asterisk -vvvgnc
gdb ./asterisk/asterisk ./core.1234
bt
paste the bt output a new bug rpt to bugs.digium.com
along with the
First of all I would like to thank Mark for getting roundrobin to go
roundrobin. Good job.
Now we have some options here for leastrecent and fewestcalls strategy. It
needs some work on the logic and Mark recommend that I ask the list and
get some input before he makes any changes to it.
Hi,
How can I play Music On Hold on a channel for just a limited period of time.
The Musiconhold application plays indefinitely.
Thanks,
Dan
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
On Fri, 8 Aug 2003, Maik Schmitt wrote:
[...]
I just tried to use it with our 7960 (sip-version).
I've set the services_url in SIPDefault.cnf to
http://xxx.xxx.xxx.xxx/xmlservices/vm/index.phpamp;user=1234amp;pin=1234;
It didn't work with ?user=...pin= cause the phone then tried to
get
Try first to stop it :
asterisk -rx stop now
then killall -9 asterisk
On Fri, 8 Aug 2003, Jim Friedeck wrote:
How do I stop asterisk when it is in a bad mood? It keeps dialing
extensions and won't listen! I tried kill PID. No go. I don't want to
have to reboot again. Thanks.
Jim Friedeck
You might look into PAT to forward your RTP traffic to the asterisk
boxes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut .
Sent: Saturday, August 09, 2003 2:37 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] X-Lite - No sound +
On Fri, 8 Aug 2003, Richard Scobie wrote:
I have been running busydetect=yes, using BUSYDETECT_MARTIN and am
having hangups during calls.
If you use also BUSYDETECT_TONEONLY then you can detect tones that are
irregular, eg: 200 ms of tone, 200 ms of silence, 200 ms of tone, 500 ms
of silence,
We have clients coming in from all over the metropolitan area with
dial-up networking issues. The problem I have is we may set their
computer to dial 1 to reach their ISP, the client gets home DUN
doesn't work because my guy forgot to remove the 1. Bad, very bad :-(
Just bought a channel
I agree...a php/forum based solution like for example
http://www.woltlab.info/en/produkte.php would be more effective and easier
to manage.
Ricardo Villa
http://www.telesip.net
- Original Message -
From: James Taylor [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, August 09, 2003
Hi,
Did you get something as caller id from the X100P card?
With the original code, in my situation the callerid information taken by
the X100P card was my own PSTN number.
With a very small change in the source code (commenting one line in
callerid.c source file) everithing now is ok.
This is
Thanks Mark.
Any plans on implementing full redirect functionality?
Michael
On Thursday 07 August 2003 06:06 pm, Mark Spencer wrote:
He should treat the first part as a local extension.
amark
On Thu, 7 Aug 2003, Michael Ulitskiy wrote:
Hi,
Does anyone know if asterisk can handle
Thanks for that,
I was looking at the extensions.conf, particularly the line in the general
section which is
TRUNK=SIP/???
Using this method would be easier.
How do you tell asterisk how many lines are available at the gateway
Dave
- Original Message -
From: Martin Pycko [EMAIL
Quoting WipeOut:
Hi,
I have just been playing with the latest X-Lite.. It works fine
with Asterisk..
As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only
one that didn't work.. not sure why..
Did you get Speex working? I've tried, but although I can get it to
Hi Dan,
I use panasonic DECT phones, when plugged into a TDM20B (2 port FXS card from Digium)
I get caller id passed through (name AND number) although i can't get callerid via the
pstn at the moment (located in nl) i do get it for VoIP calls. Plus when a pstn call
comes in and there is no
I am having trouble with the AgenCallBackLogin app. I can't seem to
define a context for the queue.
Here is the relevant configs:
queues.conf:
[general]
[default]
[q_lo_1]
music = default
strategy = ringall
context = c_in_1
timeout = 15
retry = 2
maxlen = 0
member = Agent/@3
agents.conf:
*This message was transferred with a trial version of CommuniGate(tm) Pro*
Dunno what I'm doing wrong here but I just did an upgrade to the latest
version and now I get no audio at all!
I havn't changed a single thing. Is there anything special I need to do
to get this to work again?
I
It is done. Let me know if there is anything I can do to expedite all
this. Thanks!
Jim Friedeck
--
TC wrote:
Jim
check and make sure logger.conf logs debug to the log file
start * with -vvvgnc
get it to seg fault, find the name of the core.pid file like
Ok I'm convinced..one last question will a dual PII-266 500Meg RAM have
enough horse power?
McAughan, Matt wrote:
Chris:
Try not to be so worried about sound card,
analog (FXO/FXS), digital (ISDN, BRI, PRI) and what is available by
connecting device. The channel drivers take
I run with asterisk-oh323 0.5.4 from inaccessnetwork.
Thanks
Rattana
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 04, 2003 5:39 PM
Subject: Re: [Asterisk-Users] H323 CallerID
Which H.323 channel driver are you running?
I
Well,
I've made a startup script for Debian working from init.d, for redhat
there's a script in the source tree somewhere if I remember well...
I addedd this functions on the script: Start, stop, reload, force-reload
(restart now), and console to enter the *CLI
If someone needs it, feel free to
Not better A LOT BETTER...
;-)
Dan
- Original Message -
From: William Flanagan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 06, 2003 8:21 PM
Subject: Re: [Asterisk-Users] Windows IAX soft phone
Hey all,
Why does the IAX client have better reliabilty than a
-= On Wed, 06 Aug 2003 13:38:58 -0700, George Pajari [EMAIL PROTECTED] said:
My question for the list is who will be the first to report on the
compatibility and usability of the SIPphone with Asterisk?
Well, I put in my order for the SipPhone 2-pack this morning. It
seems to be the cheapest
Dave Alan Caruana wrote:
hi ..
I have an asterisk system with three TDM100P (single port FXO) cards
and 10 Grandstream 100 phones connected to it ..
1st question:
when i phone out
or receive a call from one of the SIP phones onto the PSTN, there is
a LOT of local echo in the handset .. the
At 2:29 PM -0700 8/6/03, Steve Haehnichen wrote:
-= On Wed, 6 Aug 2003 14:18:00 -0700, John Todd [EMAIL PROTECTED] said:
There was nothing in the FAQ that really referenced anything about
using the phone with other systems other than the IPTel-powered
platform that they apparently are using.
Quoting Michael Manousos:
Michael Ulitskiy wrote:
Michael,
With all due respect to both of you, it's not related to h.323 driver.
The result is the same whether h.323 channel participates in the
call or it's pure sip-to-sip call.
Did you try it without the ztdummy and zaprtc?
I
these are taken as db right? 3.0 db = 100%
but, in some cases we've had to do txgain=9.0
is that bad, martin? are there any hardware limitations on this?
does zaptel really accept %? if so, then it should be taken as a
percentage, not pseudoDB (tm)
- wasim
On Tue, 5 Aug 2003, Martin Pycko
Does asterisk work with Vonage? I see all this talk or are you guys just
plugging it into an FXO port?
bkw
On Wed, 6 Aug 2003, Steve Meyers wrote:
On Wed, 2003-08-06 at 13:39, John Schmerold wrote:
I've canceled my Vonage service because of the requirement to prefix
every call with a 1.
Hi,
I have just been playing with the latest X-Lite.. It works fine with Asterisk..
As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only one that didn't
work.. not sure why..
But the bigger problem is that when I call another extension that is using a Snom200
the call connects
0.2.4. just upgraded to 0.2.4a. This was supposed to be fixed in some pre
0.2.4 afacr.
Snapped from #asterisk on irc.oftc.net (see levon's answer below):
RoyK ERROR[243751]: File chan_capi.c, Line 1078 (capi_request): no free b
channel on controller 2! will continue searching.
RoyK
On Tue, 2003-08-05 at 20:00, Chris Hirsch wrote:
Ok I'm convinced..one last question will a dual PII-266 500Meg RAM
have enough horse power?
I don't know what everyone else thinks, but I've run a test unit on a
PIII 400 with 128M, and at the moment there's a 133 with 92M beside me
with just
Hello all,
I have a cisco AS5300 which is register with a
gatekeeper and a Asterisk server also register with the gatekeeper.
PSTN AS5300 Gatekeeper
Asterisk
I set up a conference room on the Asterisk sever
(Room No 1234).
I try to call from PSTN to AS5300, The AS5300 will
They work great, I have 3 up and running all with mixed fxo-8 and fxs-8
cards.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Tinchev
Sent: Tuesday, August 05, 2003 4:56 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Someone used ADIT 600
I have attached the output. It is just one test call that goes to
voicemail. You can see the NOTICE message several times.
There is one thing interesting to note. If I start * from the console
asterisk -cvvv on the server I can repreduce it almost always. But if I
start it from a remote
What if someone adds your number to that list ?
Someone would have to moderate it.
regards
Martin
On Tue, 5 Aug 2003, McAughan, Matt wrote:
Does anyone keep a known telemarketer caller id database? If not has anyone
proposed an Asterisk community project to share this information? Sort
my error .. the cards are X100P which is why I wrote FXO.
The Grandstream phones are on a LAN, the * server connects to the phonelines
via the X100P cards. When I call from the Grandstream phones onto the PSTN
there is a VERY big amount of echo, ie. I can hear myself in the earpiece.
cheers
Dave
We use it, but with no caller id.
Tan
telappliant.com
- Original Message -
From: Dave Wilson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 05, 2003 1:53 PM
Subject: [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?
Hi all,
I can't seem to find any
Helo WipeOut,
I have found a solution for sending dtmf after
dial.
I use spooling. Take a look at the sample.call file
inside asterisk dir. You need to edit this file and dump it in
/var/spool/asterisk/outgoing. Asterisk will precess this file
automaticlly
Icreate the sample.call do
From: John Todd [mailto:[EMAIL PROTECTED]
exten = 4001,1,Dial(SIP/gadams,10,r)
Hmm, the syntax below is what one of the Digium guys put in. Tried your
format, getting a ring followed by the unavailable message.
For now I have the 2 SIP phones in the ATL context to test phone to
phone
401 - 500 of 501 matches
Mail list logo
