richard Coco [EMAIL PROTECTED] wrote:
Peter Svensson [EMAIL PROTECTED] wrote:
On Sun, 19 Dec 2004, Jens Kübler wrote: I've bought the Wildcard TE110 some days ago but I'm unable to get it to work with Siemens HiCom 300. I've tried this so far: 1. I've used standard cat5 cable cut off on one
Hello all,
I'm trying to make phone calls from a softphone through an ISDN line. The
problem I have is that when I try to make a call (outgoing) my ISDN card
does not respond.
The point is that i am being able to make phone calls from an ISDN phone
connected to a ISDN-PBX (the same ISDN-PBX
I came across an interesting problem, which maybe I have missed the
solution ?
I need to limit incoming calls on a PRI to a particular number.
EG: if one number is assigned to a meetme conference, I need to limit
the number of participants in the conference to say 20 members,
otherwise to many
You missed it :-)
2 possibilities
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MeetMeCount
or
http://www.voip-info.org/wiki-Asterisk+cmd+GetGroupCount
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
Sent: dinsdag 21 december 2004
Hi,
Is there a easy way to disable callwaiting ?
I had tried incominglimit in sip.conf, but it seems not to work.
/hhandresen
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To
Saw those, but they wont let me simply define a maximum number of
channels...
Actually, in my immediate case, I need to set a max for inbound pri.
IE: I had a complaint today that all our inbound was busy... yep, a
larger number than expect on a dialin number, which was a meetme
conference (80
Nathan Alberti wrote:
Duane,
My apologies if I have misunderstood but is this an error ?
Dialing a 1300XX, number would make it 611300XX, then jumping to
StripMSD(3) would make it 300XX ?
I must update that, I completely re-wrote it and the more up to date
version is at:
bugger,
I really didn't read those links close enough
thanks, problem will be solved !!
Gary
On Tue, 21 Dec 2004 19:23:49 +1000, Gary wrote:
Saw those, but they wont let me simply define a maximum number of
channels...
Actually, in my immediate case, I need to set a max for inbound pri.
I have a problem trying to install two avm fritz cards on one asterisk
machine. I am using fcusb2 driver. 1 card works perfectly.
I tried to recompile driver like it is described on this page but with
no success.
http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO
I think the problem is that i
Michael,
Success! Managed to get internal calls working no problems!!
Now... I have set up the IAX settings... still if I call an external number
it doesnt work... now any idea how can I get the pbx to route out to
someone like voiptalk, and then to the big wide world?
I'm documenting my
Has anyone successfully connected a Digium T100P to a Zhone Z-Plex 10 24
S/O? I have been unsuccessful in getting the T1 to sync up. I have
searched the documentation and concluded that a cross-over cable and
ESF/B8ZF configuration on both hardware should have cleared alarms but that
does
Hi,
When
I have registered on Asterisk server already, then I have called to Asterisk
server but appear error:
Name/username
Host Dyn Nat
ACL
Mask
Port Status
2002/2002
192.168.1.55 D
N 255.255.255.255
5060 OK (2 ms)
2001/2001
192.168.1.9 D
N 255.255.255.255
5060 OK (10ms)
On Tue, 21 Dec 2004 09:45:34 -
Paul Brock [EMAIL PROTECTED] wrote:
Michael,
Success! Managed to get internal calls working no problems!!
Now... I have set up the IAX settings... still if I call an external number
it doesnt work... now any idea how can I get the pbx to route out to
Hello,
I have X101P card.
But it seems to be dead. Always
app_dial.c:803
dial_exec: Unable to create channel of type 'Zap' (cause 0)
I've add the
line:exten = 999,1,Dial(Zap/1). But calling to 999 show the same
error.Zap show channel, lspci etc show everything is normal.
Could you tell
Hi all,
Tiny, but very important question for me: what it can be when standard
agi-test.agi script, like:
print STDERR Testing 'waitdtmf'...;
print WAIT FOR DIGIT 1\n;
my $result = STDIN;
checkresult($result);
on the call from KPhone application does NOT return any DTMF code back (I use
Hi everyone!
I wonder whether the following would be possible:
Can Asterisk show the country from which a call originates on the
display, along with the phone number?
Regards,
Evert Meulie
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Hi All,
The link http://www.voip-info.org/wiki-Asterisk+Data+Configuration
talks about how a Digium PRI card can be setup for data, and
also be used by asterisk to handle voice.
Is the same or similar possible with BRI ?
There are a few links on the Wiki to BRI cards, and
that BRI can be supported
Hi All,
I searched the mail archives, and found a few older posts about
people interested in using asterisk as a telephony platform
for VoiceXML using OpenVXI or other browsers.
Has anyone been successful with any sort of intigration ?
I have looked at trying to integrate OpenVXI with other
Title: RE: [Asterisk-Users] Troubleshooting Asterisk
Great :-)
If you use context=from-sip in sip.conf, you should include the [voiptalk] context into your [from-sip] context. (in the extension.conf)
eg.
[from-sip]
include = 2001
include = 2002
include = voiptalk
This way the Cisco's
Those suggestions were just for testing, obviously.
iax.conf
[2000]
callerid=My Name 2000
sip.conf
[2002]
type=friend
username=2002
Dialing 2002 from 2000
-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, SIP/2002|20|tT) in new stack
2000 appears on the BT100
On Tue, 21 Dec 2004 11:44:15 -
Paul Brock [EMAIL PROTECTED] wrote:
Looking for 01934830055 in from-sip
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
That actually tells why it doesn't work. :-) It can't find anything in
[from-sip] that matches the number you are trying to call.
Hello!
How do I handle calls when they reach a queue that has no members? Currently,
the callers are thrown out, because of the autofallthrough. The message is
app_queue.c:2094 queue_exec: Unable to join queue 'queue-name'
== Auto fallthrough, channel 'Zap/3-1' status is 'UNKNOWN'
It seems
Hello, Im trying to implement a function
available on the PSTN net here, if you dial a number which is busy and you
press 5, you will be called back when the busy party hangs up.
Figuring out if a SIP user is busy isnt to
hard, ${DIALSTATUS} produces a BUSY message, however, how can I
Andreas Roedl wrote:
Hello!
How do I handle calls when they reach a queue that has no members?
Currently, the callers are thrown out, because of the
autofallthrough. The message is
app_queue.c:2094 queue_exec: Unable to join queue 'queue-name'
== Auto fallthrough, channel 'Zap/3-1'
Hello
Has anyone ever come across a ...
Intel Dialogic DMV 1200 4E1 CPCI
Just been offered some and being rather a newbie not sure what if any
use they are.
Best Regards
Simon
---
This message has been scanned for viruses and
dangerous
Andrew McRory wrote:
safe_asterisk used to work fine but with v1.0.3 I am getting all kinds
of permission errors, intermittant failures, etc. Even with file
permissions relaxed and ownership set to asterisk it craps out. Seems
to work fine when run as root. Comments???
Andrew,
Sorry about the question, but did you reload asterisk after making the
changes to your conf files?
-Original Message-
From: Philipp Ebneter [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 21, 2004 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I'm confused, should I be using SetCallerID(${CALLERIDNUM}) or
SetCIDNum(${CALLERIDNUM})? Also, I don't think it matters but I'm trying to
forward the CID coming in from the PSTN line. I know Asterisk sees the CID
because its shows up in the logs. I think I've tried just about every
combination
I reloaded using asterisk -rx reload.
-Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer
Sent: Tuesday, December 21, 2004 7:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Grandstream
[EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield Sent: Saturday, 4 December 2004 1:44 AM
To: Ed Greenberg; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Ouch, part
E. Versaevel wrote:
Hello, Im trying to implement a function available on the PSTN net
here, if you dial a number which is busy and you press 5, you will be
called back when the busy party hangs up.
Cron job, eighter parse every 10s both peer statuses, and create a call
file that is in the
On Mon, 2004-12-20 at 13:45, Me wrote:
What does t mean in a CDR entry?
The 't' probably means that the call ended up in the timeout extension.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
I was doing a daily make update for asterisk. On the 19th the new version
compiled fine. I installed it. All of my ata 186's can call out to pstn etc.
All inbound calls, the phones ring but when you pickup, just silence both
local and remote with no complaints in the cli. I backed down to the r
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield Sent: Saturday, 4 December 2004 1:44 AM
To: Ed Greenberg; Asterisk Users Mailing List - Non-Commercial
Discussion Subject: Re:
Milos Kocbek wrote:
I have a problem trying to install two avm fritz cards on one asterisk
machine. I am using fcusb2 driver. 1 card works perfectly.
The multiple fritz hack works only with pci cards (and, of course, it's
a hack). Avm decided to not allow multiple installation for fritz
Does anyone know of any obscur reference for detecting an incoming fax.
I currently have AMP running and everything else is working great.
Installed the spandsp patches and software... using the default AMP
extensions.conf, I start sending a fax, I hear it pick up and transfer
to voicemail after
It's not all that obscure. :)
http://www.voip-info.org/wiki-Asterisk+fax
Look for Zap fax detection
On Tue, 21 Dec 2004 08:52:11 -0500, Sean Cook [EMAIL PROTECTED] wrote:
Does anyone know of any obscur reference for detecting an incoming fax.
I currently have AMP running and everything else
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John Hill
Sent: Tuesday, December 21, 2004 8:40 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] upgraded source now ata's ring but stop silence
oninbound calls
I
On December 21, 2004 08:29 am, Jim Van Meggelen wrote:
5 Watts per FXS
They also told me each FXS port would support a REN (ringer equivalence
number) of 5.0, which means that you should be able to to ring five of
the old electromechanical telephones simultaneously off of each FXS port
on
It is not possible to do. You could however utilize the local channel
to accomplish something like rollover. Check out forking in the wiki
On Tue, 21 Dec 2004 17:26:18 +1100, Eric Bishop [EMAIL PROTECTED] wrote:
Does any body know if it is possible to group SIP channels just like
it is
Just out of curiosity, is that a network cross-over cable or a T-1
cross-over cable? For the pinout for a T-1 cross-over cable see
http://www.voip-info.org/wiki-crossover+T1+cable
On Tue, 2004-12-21 at 01:49 -0800, Jonathan Augenstine wrote:
Has anyone successfully connected a Digium T100P to
I have that set up ...
zapata.conf
faxdetect=incoming
still no dice... It has to be something trivial that I am missing. I
will probably spend 15 hours on it an notice later that I spelled facts
instead of fax :)
Sean
On Tue, 2004-12-21 at 08:59 -0500, Jon Radon wrote:
It's not all that
Sean Cook wrote:
Does anyone know of any obscur reference for detecting an incoming fax.
I currently have AMP running and everything else is working great.
Installed the spandsp patches and software... using the default AMP
extensions.conf, I start sending a fax, I hear it pick up and transfer
to
On December 21, 2004 08:29 am, Jim Van Meggelen wrote:
5 Watts per FXS
They also told me each FXS port would support a REN (ringer equivalence
number) of 5.0, which means that you should be able to to ring five of
the old electromechanical telephones simultaneously off of each FXS port
Actually it is the default install, no changes yet... Maybe the dial
group getting answered before fax detection...
Sean
On Tue, 2004-12-21 at 07:34 -0700, Jason Becker wrote:
forum.
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is h323 per user
based working??? I have setup this:
[User1]type=userhost=xx.xx.xx.xx
context=international
incominglimit=30
But all calls from
xx.xx.xx.xx are not routed to context international, it is
working?
I am using
chan_h323
Thanks!!
Sebastian
Nocetti.
---
Checked by
That was the problem... dial groups drop the call into a different
context before fax can be detected.
I knew it was something simple that I was missing ;)
On Tue, 2004-12-21 at 09:41 -0500, Sean Cook wrote:
Actually it is the default install, no changes yet... Maybe the dial
group getting
Hello!
Am Dienstag, 21. Dezember 2004 13:59 schrieb Senad Jordanovic:
It seems that Queue() won't continue at a specific priority - like
n+101 - if there are no members in the queue.
Use...
Joinempty=yes
Perfect! Thanks.
Andi
--
- Andreas Roedl- Senior IT Manager
- NATIVE
Sean Cook wrote:
Actually it is the default install, no changes yet... Maybe the dial
group getting answered before fax detection...
[EMAIL PROTECTED] root]# grep FAX_RX
/etc/asterisk/extensions_additional.conf
FAX_RX = system
FAX_RX_EMAIL = [EMAIL PROTECTED]
These parameters are in the
When a new member is rejected from the queue (because there's a limit,
or there's no agents logged into the queue), is it possible to either
set an announcement, or to elevate the caller to a new priority (i.e.
n+100) or something?
Regards
Ben Merrills
-Original Message-
From: [EMAIL
HI
How would I get the MWI working on the Grandsreams?
Thanks
Doug (Yip another one!)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Doug Lytle
Sent: Monday, December 20, 2004 5:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
You are correct... typo on my part... extentions_additional.conf are
braught into the extensions.conf from the include... under globals
(didn't do a cut and paste for the context)
Thanks,
Sean
On Tue, 2004-12-21 at 07:52 -0700, Jason Becker wrote:
Sean Cook wrote:
Actually it is the default
Hello,
I am trying to exchange SMS between a fixed phone and an Asterisk. The
intention is to make the Asterisk become a SMS Center, because we do
not have public SMS Center in our country.
I have two phone lines, one for Asterisk and one for the SMS enabled
fixned phone. I also config the fixed
...For Asterisk gurus, that believe that you can take the exam without
attending the training, there will be exam oppurtunities setup in
combination with Astricon conferences. When we update dCAP for future
releases of Asterisk (1.1, 2.0), you will be able to upgrade your
certification at
in your sip.conf.
voicemail=your extension
you do not need to change grandstream configuration...
Diego Aguirre
- Original Message -
From: Doug Reid - Stormcorp [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; [EMAIL
what is the best way to have 2 asterisk servers communicate with each other?
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On Tue, 21 Dec 2004, Stefan de Konink wrote:
E. Versaevel wrote:
Hello, Im trying to implement a function available on the PSTN net
here, if you dial a number which is busy and you press 5, you will be
called back when the busy party hangs up.
Cron job, eighter parse every 10s both
Webmin and Web Admin interfaces are two different things. I have not
come across any Asterisk Modules in Webmin format, where they can be
managed (Installed/De-Installed) from a Webmin Window using webmin port
at http://ipaddress:1000
If anyone knew of such modules for Asterisk or re-create
Ben Merrills wrote:
When a new member is rejected from the queue (because there's a limit,
or there's no agents logged into the queue), is it possible to either
set an announcement, or to elevate the caller to a new priority (i.e.
n+100) or something?
I have been wanting this as well for some
I reported this on dev yesterday.. I thought I saw it fixed in dev but
not stable according to the cvs list..
Modified Files:
chan_sip.c
Log Message:
Minor ACk fix (bug #2687, again)
So the stable version is still borked.. but head should be cleared
up..heh, stable ain't that stable right
chek nationalprefix= and internationalprefix= settings in zapata.conf.
DUnno but maybe somebody included them from bristuff
On Mon, 20 Dec 2004 14:15:52 +0100, Gunnar Schaller [EMAIL PROTECTED] wrote:
Hello,
I have a little problem with signalling. An E100p is connected to an
Alcatel PBX,
Hello All,
Can asterisk play voice prompt and collect digits on the IP leg ( ie.
The incoming VoIP call)?.
Thanks Regards
-kts
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To
B G wrote:
I have two phone lines, one for Asterisk and one for the SMS enabled
fixned phone. I also config the fixed phone to have the SMS Center
number as the phone number for Asterisk.
This may be a dumb suggestion, but do you have CLI enabled on
your phone line ? I read somewhere that some
Hi,
Just a quick word on this since I was fortunate enough to attend.
There were about 18 people, almost all French (if you include the
marseillais as French, they may have objections :) Not that I was
counting, but there was one female human there.
Thanks Mark for your generosity and the good
I use SetCallerID() and it displays the number just fine on my
GrandStream phone.
Regards,
Scott Stingel
www.evtmedia.com
David Ishmael wrote:
I'm confused, should I be using SetCallerID(${CALLERIDNUM}) or
SetCIDNum(${CALLERIDNUM})? Also, I don't think it matters but I'm trying to
forward
William Betts [EMAIL PROTECTED] wrote:
what is the best way to have 2 asterisk servers communicate with each
other?
Probably using IAX2.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
_/ _/_/ _/ _/
Wow babelfish interpreted that French pretty bad =(
Since j'ai forgotten to do it, if you see this message, send to me your mall
so qu'on remains in contact. I opposite Marek, where were you with the table
are the preque old man? With soon!
Robert A. Huddleston, KF4BYY
Cavalier Telephone LLC.
At 04:17 PM 12/20/04, you wrote:
On December 20, 2004 04:02 pm, Greg - Cirelle Enterprises wrote:
Could I ask how you've connected the t1s? I'm going to be getting a
non-pri t1 ( 9 channels of voice, the rest off ). I assume I'll just
get an rj45(ish) plug to plug into the back of the card
Hi all,
I cant ring my phone with fxstest, but all else works (playing tones,
stats , regdump).
My ring voltage is -53.7680 volts as reported by stats
option to fxstest.
However, I can ring my phone with Asterisk !!!
With fxstest, I just hear a click if I am off hook,
and ask it to ring my phone.
IAX = Inter-Asterisk eXchange
-Original Message-
From: William Betts [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 21, 2004 9:10 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] asterisk server to asterisk server
what is the best way to have 2 asterisk
Hi,
I am new with asterisk. I am setting a Wildcard TE405P. E1s in Italy come in
on a pair of RG-59 coax cables with BNC connectors. So I need an
adapter/balun http://www.allcomtlc.com/al_g703n3.htm . I have It but I am
not sure that It works. I have configured my asterisk in this way:
This reminds me of the local computer store owner that put all his certs up
on the wall. One that really stood out was his pearl certificate. (pearl?
wtf I think he wanted to fake that to say PERL) Never underestimate the
power of stupid people with printers.
bkw
-Original Message-
hi all
Can some bady send me the sip firmware for a cp7902g phone ???
i buy cisco equipment by a third person and he don´t want to help me.
Thanks in advance
Regards
wert
Do you Yahoo!?
Yahoo! Mail - Easier than ever with enhanced search. Learn
Doug Reid - Stormcorp wrote:
HI
How would I get the MWI working on the Grandsreams?
Thanks
Doug (Yip another one!)
Doug,
Currently, my voicemail is on extension 5700, so under the GS web
interface, under Voice Mail User ID, I put 5700
Now, when pressing the message button, I get the
Is it possible to use the sms_app over zap without the .call file?
I tried the example with zap behind a hipath. Reciving a SMS works fine, but if
i send an SMS using the .call file i recive an SMS without a message.
Why didn't you use the lines?
Application: SMS
Data: default,,MOBILE
On December 21, 2004 10:51 am, Greg - Cirelle Enterprises wrote:
Is this an hdlc implementation?
My particular application is not, it's CAS T1 (channelized T1, voice only)
-A.
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Hess
Sent: Tuesday, December 21, 2004 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] upgraded source now ata's ring but stop
maybe a dumb question but what do we have here???
sip-friends.sql
#
# Table structure for table `sipfriends`
#
CREATE TABLE `sipfriends` (
`name` varchar(40) NOT NULL default '',
`secret` varchar(40) NOT NULL default '',
`context` varchar(40) NOT NULL default '',
`username` varchar(40)
Hi,
I'd like to build a personal PBX connecting 4 or 5 analogic phones with a
ADSL line and I'd like to know what is the right card I need
I visited digium site and I think TDM400 could be the right choice but I
cannot understand how it works...I think it has 4 slots where 4 modules
(FXS or FXO)
At this time of year, I think sending Christmas cards would be a nice way for
them to stay in touch.
On Tue, 21 Dec 2004, William Betts wrote:
what is the best way to have 2 asterisk servers communicate with each other?
___
Asterisk-Users mailing list
I don't think I did anything special for that to work other then
configure voice mail for that extension.
MWI just works (the display flashes and there's stutter tone).
I reviewed my settings to check if something has an impact on that.
In sip.conf I have [EMAIL PROTECTED]
In voicemail.conf
Did anybody already attempt to strip down an asterisk config
to an absolute minimum for a specific use?
Let's say I have a home installation and want to use capi and
iax exclusively, and load only the channels, apps, codecs,
file formats I really need.
Obviously, to dig through the whole stuff,
Fyi, just saw a fix go through for stable :)
John Hill wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Hess
Sent: Tuesday, December 21, 2004 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Howdy--
I'm playing with different IAX softphones. I've got DIAX and IAXPHONE on
a windows (XP) machine on my network, and I'm running asterisk on a
newly loaded Fedora Core 3 machine.
I set up a separate IAX account for each phone. I was EXPECTING them
to each register seperately with
Hi-
Just a couple of things to check:
I assume that the LED's blink RED when you first power up, meaning that
the board is basically alive.
Since you're E1, just checking that you've installed the 4 required
jumpers on the card for E1
It's possible you need a crossover cable instead of
--On Tuesday, December 21, 2004 9:37 AM -0700 Steve Murphy
[EMAIL PROTECTED] wrote:
Howdy--
I'm playing with different IAX softphones. I've got DIAX and IAXPHONE on
a windows (XP) machine on my network, and I'm running asterisk on a
newly loaded Fedora Core 3 machine.
I set up a separate IAX
I'm new at this, too, but I think I can field this one. FXS ports are for
connecting to your phones and FXO ports are for connecting to the PSTN. In
your situation you would get a TDM400P card with four FXS ports which will
allow you to configure 4 individual extensions. The DSL doesn't use a
Hi All;
Hi Matthew;
I used Mysql-realtime for sipfriends and extensions
and it is ok for me.Now I come up with 2 questions:
1) What I understand from Realtime is , there is no
need to reload Asterisk to read new setting. I am right? Can you plz explain me
in depth.
2)Is there any GUI
On Tue, 2004-12-21 at 17:05 +0100, Gutzke Klaus wrote:
Is it possible to use the sms_app over zap without the .call file?
in newer versions of asterisk there is smsq - a tool that sends sms from
the command line.
see the wiki at http://www.voip-info.org/wiki-Asterisk+cmd+Sms
I tried the
Hello all,
The new versions 0.7.1 (for Asterisk CVS HEAD) and 0.6.5 (for Asterisk
STABLE) fix a deadlock in outgoing H.323 calls and a bug that caused
chan_oh323 to update incorretly the DIALSTATUS variable.
Download from the usual location:
http://www.inaccessnetworks.com/projects/asterisk-oh323
On Sat, 18 Dec 2004, Anders F Eriksson wrote:
I've never tried softphones on Linux, but my guess is that since you run
kphone and asterisk on the same server you get a port conflict. If the
client uses port 5060 (default sip port) it would defenitely have
problem connecting to an asterisk on
Hi,
i want to know if it is possible to use
an analogic phone with a modem in asterisk ?
i don't want to use a digium card or any FOX/FXS module !
perhaps a modem card ??
--
Mathias Houngbo
http://mathias.houngbo.net/
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Giorgio.
To connect 5 analog phone you need five FXS ports.
As for the ADSL line, please clarify your intentions.
If you wish to use the ADSL line to connect to a VOIP provider, like vonage or
voicepulse, then you connect it to your machine with a network interface.
If you wish to use it for
Joel Moore wrote:
I'm new at this, too, but I think I can field this one. FXS ports are for
connecting to your phones and FXO ports are for connecting to the PSTN. In
your situation you would get a TDM400P card with four FXS ports which will
allow you to configure 4 individual extensions. The
Hello.
I'm having this same problem with a Budgestream phone: I've correctly
installed G729 licensed codec (I've made all the checks you said in this
thread), but Asterisk keeps giving the fatal message:
*CLI Dec 21 18:02:49 WARNING[2375]: chan_sip.c:2764 process_sdp: No
compatible codecs!
Dec
Hi all,
I had recompiled and installed more new 4.1.0 version of KPhone, and my big
trouble had gone - thanks!
Vasya.
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To
I've downloaded and compiled zaphfc and libpri.
To do that i've downloaded bri-stuff and commented out the asterisk related
stuff because i've installed it from a debian package.
Does this means that i have to rebuild the whole asterisk thing to support
zaphfc?
thanks
ciao
Hello.
I'm having this problem with a Budgestream phone: I've correctly
installed G729 licensed codec in my asterisk box, but when I set my
budgestream to use only g729 codec, asterisk throws this message:
*CLI Dec 21 18:02:49 WARNING[2375]: chan_sip.c:2764 process_sdp: No
compatible codecs!
Mathias Houngbo [EMAIL PROTECTED] wrote:
i want to know if it is possible to use
an analogic phone with a modem in asterisk ?
i don't want to use a digium card or any FOX/FXS module !
perhaps a modem card ??
You could consider a Sipura device. The SPA-2000 comes with two FXS
ports, for
Christopher L. Wade wrote:
Joel Moore wrote:
I'm new at this, too, but I think I can field this one. FXS ports are
for
connecting to your phones and FXO ports are for connecting to the
PSTN. In
your situation you would get a TDM400P card with four FXS ports which
will
allow you to configure 4
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