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Hello everyone,
I'm relatively new to the subject - so pleace don't punish me for
idiotic questions. ;-)
Can Asterisk act like a normal Sip phone and e.g. connect to another
sip-gateway? Background: There is a new german company at:
You can use,
;sip.conf
register = username:[EMAIL PROTECTED]/extension
to make asterisk as a SIP client.
to forward calls to another client use canreinvite=yes, (if the client
supports reinvite)
and in the extensions.conf
exten = s,1,Dial(SIP/username:[EMAIL PROTECTED])
Kannaiyan
-
Hi,
I downloaded this the other day and finally got it to stop crashing. It appears that
any response from asterisk
that implies an error (for example dialing a non-existant number, using the wrong
password, selecting a codec
that you've configured a local * not to use etc) resulted in a crash.
Anton wrote:
you can do it with a well setup cluster
OK, so what success have people had with which clustering technologies?
I'm more interested in resilience than performance.
Thanks a lot,
F
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[EMAIL PROTECTED]
Hi!
Also, does anyone feel a need to have the voicemail system speak the
date and time the voice mail message arrived for those that access
messages by phone instead of the usual email?
Did you look at voicemail.conf and the tz= settings? Simply create a
timezone that fits your needs.
Hi,
just installed Firefly. Looks great, sound is also great. I just got the
following problem.
I'm using Firefly with my asterisk*-box. When I enter a contact with the
number +00233612345 Firefly just erases the 00 when I restart it. Am I
missing something?
Thanks! Great software!!!
On Fri, Jan 30, 2004 at 08:20:48AM -, Kannaiyan Natesan wrote:
You can use,
;sip.conf
register = username:[EMAIL PROTECTED]/extension
to make asterisk as a SIP client.
[...]
Can Asterisk act like a normal Sip phone and e.g. connect to another
sip-gateway? Background: There
I think there is a loopback.
Did you debug that with sip debug in console and look at SIP Messages what
is doing ?
Kannaiyan
- Original Message -
From: Walter Doerr [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 9:42 AM
Subject: Re: [Asterisk-Users] Can Asterisk
On Fri, Jan 30, 2004 at 09:51:44AM -, Kannaiyan Natesan wrote:
I think there is a loopback.
Or, their SER forwards packets to itself. Hard to tell without knowing
their config.
Did you debug that with sip debug in console and look at SIP Messages what
is doing ?
Yes.
-Walter
--
hi,
just signed up and it works like a charm. :-)
They even support g711 :) and multiple channels :)
make sure you have in sip.conf:
register = :[EMAIL PROTECTED]/extension in your context
you will get the too many hops if you try to register
with their proxy (proxy.de.sipgate.net).
best
The phone works fine with oh323, its just the need to authenticate the
endpoint and match a non-fixed ip to a number that has sent me off in the
direction of gnugk. If I could do it all in * I would.
thanks,
brian
At 18:05 29/01/04, Roger wrote:
Hi,
I also had some problems using chan_oh323
Hi.
I am having the following warning when using music on hold.
It works from X-Lite to Grandstream. I get a lot of errors
and warnings.
1.Warning, flexibel rate not heavily tested!
2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread:
Request to schedule in the past?!?!
Hi
I am using asterisk 0.7.1
I am testing with 3 SIP phone.
Phone A call to Phon B , and Phone B is Ringing.
I want to pickup that call , So I press '*8' for pickup the call on Phone
C.
But I can not pickup the call.
I can see "NOTICE[6151]:chan_sip.c:5198 handle_requst: Nothing to
I have * working on my Sony Vaio PCG-FX120 laptop. I am trying to get * to
recognize my internal PCI Intel modem as an FXO port. I have modified
wcfxo.c in order to identify the PCI modem properly. Based on output from
dmesg, wcfxo didn't recognize the modem until I inserted the proper
Dear All,
Even its not much relevant to ask at this forum but If anyone can commnet..
I m trying to run CP4/LSI on Linux RHL 8.0 box, tried with LINUX_SR5.1.tgz, LiS-2.17.A.tgz
It gives errors like..
Dialogic Shared RAM Protocol ModuleVersion 2.0Linux 2.x.xKernel 2.4.xCopyright (C) 2001 Intel
On Fri, Jan 30, 2004 at 12:06:47PM +0100, Klaus-Peter Junghanns wrote:
hi,
just signed up and it works like a charm. :-)
They even support g711 :) and multiple channels :)
make sure you have in sip.conf:
register = :[EMAIL PROTECTED]/extension in your context
I believe that I
Hi,
I've got a fairly working Asterisk setup, with a few minor glitches, one of
which is very very irritating.
Sometimes, during a call, the remote end just drops off. We're using software
SIP phones (SJPhone) connecting to * then out through analogue lines with
X100P cards.
There is nothing in
Hi to all i have made a little modification to app_dial.c to play a mex when a
call is connect like the A(mex) but from caller side option my new is B(mex).
If someone think is good think a made patch for *. I use this mod to play a
DTMF wav of C tone :-)
Thank in advance
Dimitri
PS:Nick how
Steve Foy wrote:
Hi,
I've got a fairly working Asterisk setup, with a few minor glitches, one of
which is very very irritating.
Sometimes, during a call, the remote end just drops off. We're using software
SIP phones (SJPhone) connecting to * then out through analogue lines with
X100P cards.
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On Friday 30 January 2004 00:57, Eric Wieling wrote:
Is there any chance 0.7.2 will include a fix for PRI Cause Codes not
being translated into Asterisk Cause Codes and being passed back to
app_dial (as well as fixing the apparently never working
On Fri, Jan 30, 2004 at 12:22:18PM +0100, Walter Doerr wrote:
On Fri, Jan 30, 2004 at 12:06:47PM +0100, Klaus-Peter Junghanns wrote:
hi,
just signed up and it works like a charm. :-)
They even support g711 :) and multiple channels :)
make sure you have in sip.conf:
register =
How? Is written in CDR?
Regards,
Gus
- Original Message -
From: Tais M. Hansen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 9:20 AM
Subject: Re: [Asterisk-Users] Echo worsens in 0.7.1
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On Friday 30 January 2004
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On Friday 30 January 2004 13:31, CW_ASN - Gus wrote:
HANGUPCAUSE is working fine here (cvs).
How? Is written in CDR?
CDRs contain BUSY when busy and NO ANSWER on the rest.
extensions.conf:
[provider-out]
...
exten =
Hello
Is anyone using a commercial billing
software with * which product is that?
i am looking for using with pre-paid as
well as post paid.
Also where can i find info about voip
regulation/licenses to become a provider???
Thanks
Deepak
I have compiled the zaptel library and zaprtc on a system that gives the
following from uname -a:
Linux fxx76.mydomain.de 2.4.18-64GB-SMP #1 SMP Wed Mar 27 13:58:12 UTC
2002 i686 unknown
Makefile for zaptel had the following line uncommented:
#
KFLAGS+=-D__SMP__
When doing the make load for
Hi,
I have problems with Asterisk recognizing dtmf tones sent by Mediatrix 1104
FXS. I can not enter mailbox number (voicemail) or pin code (meet-me).
Asterisk shows 'username not entered' when dialing in voicemail.
Both asterisk and Mediatrix have set inband dtmf. Can anyone help me out ?
Shot in the dark here ...
Do you have:
canreinvite=no
Set in sip.conf for the SIP phones in question ?
Ciao,
-b
Quoting Steve Foy [EMAIL PROTECTED]:
Hi,
I've got a fairly working Asterisk setup, with a few minor glitches, one of
which is very very irritating.
Sometimes, during a
Hi all,
I have looked through the wiki for any information on how to make an
extension autodial another extension when it goes off hook.
Anyone done this or know how it's done.
regards
Dave
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Todd Wallace wrote:
I noticed in the BUGS that there is a memory leak with * using
asterisk-oh323. If we use SIP primarily as the main protocol, but OH323 on
occasion to test some international routes on our Nextone MSW...How bad is
the Memory leak that is described??
Todd Wallace
This was a
I'm confronted with an issue that I am sure many others are too with Asterisk and
scalability. I'd like to be able to build a cluster of Asterisk boxes to handle a
large volume of simultaneous calls but have the feeling that the hardware requirements
to handle large volumes of RTP streams
hi,
I use IAX softphone with asterisk and I notice that
a call between two IAX softphonesend after 1 min. Then I can't hear
anything but the call still in progress.
I have this log in asterisk IAX debug:
Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type:
IAX Subclass:
ACK Timestamp:
Hi!
just signed up and it works like a charm. :-)
They even support g711 :) and multiple channels :)
make sure you have in sip.conf:
register = :[EMAIL PROTECTED]/extension in your context
Their tech support just told me that it takes a while until a registered
user becomes available to
On Fri, Jan 30, 2004 at 01:18:29PM +0100, Olle E. Johansson wrote:
Enable 'sip debug' at the CLI and send some detailed log file.
It's very difficult to catch the logs when this happens, it doesn't happen
all the time, and I'm hardly ever on the phone so, it would be even less
likely to happen
Jeff Crews wrote:
[snip]
Any thought of having maximum number of messages be defined globally
in voicemail.conf or on a per user basis?
I think this is a good idea. But instead of the two extremes, maybe we
could come up with a class of service definition (idea shamelessly stolen
from
Bill,
On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote:
Shot in the dark here ...
Do you have:
canreinvite=no
Set in sip.conf for the SIP phones in question ?
No, I don't.
All I have in sip.conf is the general stuff like:
[general]
port = 5060 ; Port to
Chris Albertson wrote:
What are you talking about? You can already reload the dialplan
without affecting existing calls.
[snip]
But for my use this is not important as at most I'd be running
a small office with ~10 lines were people go home at night.
But with 25,000 active users ...when
GREAT!!! Just got my asterisk* calling firefly users. Setup was really easy:
just add an extention
exten = _8XXX,1,Answer
exten = _8XXX,2,DigitTimeout,5
exten = _8XXX,3,ResponseTimeout,10
exten =
_8XXX,4,Dial,IAX2/*YOUR_FIREFLY_NUMBER*:[EMAIL PROTECTED]
.com/${EXTEN}|60|T
now I
Isamar Maia wrote:
I know that it was commented here already but how many X100Ps
I can plug per PC?
How many PCI slots do you have? How many IRQ's can your BIOS allow you to
assign? There is not a hard and fast rule, as far as I can tell, but these
questions may give you an idea of how many
[This starts to look very off-topic for the -users list, which is why
I've proposed on several occasions a -biz list, but since there is no
-biz list, I'll continue this thread here.]
I currently do not have German as a language at my command, and there
are no English translations on the sipgate
Hi all,
I have looked through the wiki for any information on how to make an
extension autodial another extension when it goes off hook.
Anyone done this or know how it's done.
regards
Dave
Depends on the phone. If you have an FXS interface, look for
immediate= in your zapata.conf file.
If
Greg Boehnlein wrote:
On Thu, 29 Jan 2004, Michael Welter wrote:
I have 1.0.4.45 (beta) on my tftp server. Try it at 66.250.23.58.
Cheers,
Michael Welter
Is there a changelog available for the Beta release train? I'm looking to
see if they have fixed Early Dial yet.
When GS connected to
At 8:16 AM -0600 1/30/04, David Gomillion wrote:
[snip]
The when convenient is usually tied to restart. Reload reloads
configuration files. Restart now will, indeed, drop conversations, while
reload will not. Reload will not change everything, so for some changes,
you will have to issue a
Ok, but is not working as expected... we can't see clear ISUP causes. We
can't make different treatments or store other causes than busy (cause=17)
in cdr's .
Regards,
Gus
- Original Message -
From: Tais M. Hansen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004
G'day list,
I am getting a lot[1] of traffic on my Internet link, ICMP messages from
69.73.19.178 telling me UDP port 5036 is unreachable (this IP address
belongs to iaxtel.org).
I see from the wiki that IAXtel supports only IAX2 from December 2003.
Fine, however it looks like my * still
Hello all,
I would like to just verify where to purchase the
G729 license for Asterisk. Like I want to run G729 codec for all my calls
passing thru Asterisk (voicemail, parking, via ZAP, via SIP, etc). The
list says license is taken from Digium, does that apply also if I have Dialogic
greetings
I am interested in building asterisk on a BSD/ OSX platform.
is there a source i can compile?
for testing purposes would i be able to use a modem for outside line
connections?
any info would be helpful
thanks
--jeff
---
jeff donovan
basd network
Dear ALL
i need to develop a web frontend for my * app i need only manage data from
MySQL db, i will pay to develop it (not much :-) )
Thanks in advance
Dimitri
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On Thu, 29 Jan 2004 20:01:05 -0600, Tilghman Lesher wrote
On Thursday 29 January 2004 19:40, Asterisk VOIP wrote:
almost got it. I now get the following in the CLI,
ERROR[1226054960]: cdr_addon_mysql.c:203 mysql_log: Failed to insert
into database. db is setup correctly.
You probably
Hi!
the product offerings for me. I am looking for a phone number in .de
that will map by ENUM (and of course, by PSTN-to-SIP) to one of my *
servers.
Both nikotel.de and sipgate.de offer such a number. With Nikotel you'll
need to spend at least 7 per month for such a number (which is a
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On Friday 30 January 2004 15:59, CW_ASN - Gus wrote:
Ok, but is not working as expected... we can't see clear ISUP causes. We
can't make different treatments or store other causes than busy (cause=17)
in cdr's .
You could use my approach and
in the *short term* just add:
noload = chan_iax.so
to your modules.conf
Eventually we will move chan_iax2.c to chan_iax.c and chan_iax.c will
become chan_iax1.c and will likely not be a default part of the build
process.
Mark
On Sat, 31 Jan 2004, Vic Cross wrote:
G'day list,
I am getting
On Thu, Jan 29, 2004 at 04:10:42PM -0800, David Liu wrote:
Hi there,
I am trying to delay sending out DTMF from Voicetronix OpenLine4 to the CO
line. The reason being is that Voicetronix sends out the DTMF too fast even
before the line is fully established with the carrier. Usually when
John,
at the moment German and UK geographic numbers are available free of
charge with unlimited inbound traffic through SIP. ENUM mappings are
static in the way that enum zones cannot be configured by a user, but
point to a SIP UAS at which you can register an * box. Let me know, if
you want me
On Friday 30 January 2004 09:41, Asterisk VOIP wrote:
On Thu, 29 Jan 2004 20:01:05 -0600, Tilghman Lesher wrote
On Thursday 29 January 2004 19:40, Asterisk VOIP wrote:
almost got it. I now get the following in the CLI,
ERROR[1226054960]: cdr_addon_mysql.c:203 mysql_log: Failed to
I am having a little bit of a problem with BT rejecting my callerid values
as they are prefixed by hex b. This indicates that the caller id is user
provided and not verified.
Does anyone know how I can control where this appears in the cli?
The purpose of the separator is described below:
1 -
Hello All,
I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't
configured properly. Now heres the next question. 12 EM wink lines from telco. I
have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are
configured DPO. How do I signal
On Friday 30 January 2004 04:33, Craig Waddington wrote:
1.Warning, flexibel rate not heavily tested!
You're using variable rate mp3's. If you want to avoid the error,
recode your mp3s to a static rate.
2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request
to schedule in the
On Friday 30 January 2004 07:04, [EMAIL PROTECTED] wrote:
I have compiled the zaptel library and zaprtc on a system that
gives the following from uname -a:
Linux fxx76.mydomain.de 2.4.18-64GB-SMP #1 SMP Wed Mar 27
13:58:12 UTC 2002 i686 unknown
Makefile for zaptel had the following line
On Friday 30 January 2004 08:59, Vic Cross wrote:
G'day list,
I am getting a lot[1] of traffic on my Internet link, ICMP messages
from 69.73.19.178 telling me UDP port 5036 is unreachable (this IP
address belongs to iaxtel.org).
I see from the wiki that IAXtel supports only IAX2 from
Try adding it to the phones involved so it looks like this:
; Shirley
[100]
type=friend
username=xxx
secret=xxx
host=dynamic
dtmfmode=rfc2833
callerid=Shirley O'Neill 100
context=internal
[EMAIL PROTECTED]
qualify=yes
canreinvite=no
-b
Quoting
Hello All,
I've mostly solved my DID problem from a few days ago. Apparenly the
lines weren't configured properly. Now heres the next question. 12 EM
wink lines from telco. I have them all plugging into an Adtran 750 with
FXS cards. The Adtran ports are configured DPO. How do I signal
Oops, wrong email, please ignore.
Thanks,
Thilo
On Fri, 2004-01-30 at 17:16, Thilo Salmon wrote:
John,
at the moment German and UK geographic numbers are available free of
charge with unlimited inbound traffic through SIP. ENUM mappings are
static in the way that enum zones cannot be
Personally I would like AST_CAUSE to be the Asterisk cause code (which
should be the same for all technologies), TECH_CAUSE (IAX2_CAUSE,
SIP_CAUSE, PRI_CAUSE) would be interesting and useful to some people.
On Fri, 2004-01-30 at 09:57, Tais M. Hansen wrote:
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Are you sure this works for VoiceTronix Driver? It's not implemented in
app_dial, but in chan_zap.
On Fri, 2004-01-30 at 10:15, Walker Haddock wrote:
On Thu, Jan 29, 2004 at 04:10:42PM -0800, David Liu wrote:
Hi there,
I am trying to delay sending out DTMF from Voicetronix OpenLine4 to
I have been following and reading about the SIP
problem of transferring calls with Asterisk. I did not seethis
problem as havinga fix orhaving apatch for it. I can not
use the # in our system due to IVR systems we access.
Can someone let me know at what stage this is
at. This is a major
Low, Adam wrote:
I'm confronted with an issue that I am sure many others are too with Asterisk and scalability. I'd like to be able to build a cluster of Asterisk boxes to handle a large volume of simultaneous calls but have the feeling that the hardware requirements to handle large volumes of
What is your * app?
What should the frontend do?
Greetings,
Doichin Dokov
reseaux wrote:
Dear ALL
i need to develop a web frontend for my * app i need only manage data from
MySQL db, i will pay to develop it (not much :-) )
Thanks in advance
Dimitri
Thanks, I'll try that and see how it goes.
Cheers,
Steve
On Fri, Jan 30, 2004 at 11:46:05AM -0500, Bill Hamel wrote:
Try adding it to the phones involved so it looks like this:
; Shirley
[100]
type=friend
username=xxx
secret=xxx
host=dynamic
dtmfmode=rfc2833
I just fetched today's cvs (1/30/04 11:10:31). Compiles/installs on my
test machine (ASUS A7V, 900 MHZ). However, If I try to compile on my
production machine (Elite K7S5A, 2.4GHz, 512MB) while * is running the
zaptel and asterisk compiles seg fault. I am assuming that they will
compile
Does anyone know if Allison has recorded anything along the lines of:
You don't have permission to dial that number.
Thanks.
--Lance Arbuckle
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http://lists.digium.com/mailman/listinfo/asterisk-users
To
On Friday 30 January 2004 10:15, Walker Haddock wrote:
On Thu, Jan 29, 2004 at 04:10:42PM -0800, David Liu wrote:
Hi there,
I am trying to delay sending out DTMF from Voicetronix OpenLine4
to the CO line. The reason being is that Voicetronix sends out
the DTMF too fast even before the
Rattana,
I have had the same problem with IAX Phone. I think there is still
something slightly off in iaxClient_lib.c or one of the associated files. I
am trying to figure it out myself. Please send me any additional debugging
files as you generate them.
Thanks,
Steve
Lance Arbuckle wrote:
Does anyone know if Allison has recorded anything along the lines of:
You don't have permission to dial that number.
Or a more versitile way of saying it..
The number you dialed is not permitted.
This could then mean that *you* are not allowed to dial it or the
Lance Arbuckle wrote:
Does anyone know if Allison has recorded anything along the lines of:
You don't have permission to dial that number.
I think so... under tt-monkeys.gsm.
Thanks.
You're welcome.
PS. Sorry, I couldn't resist on this one.
See Bug Number 890 on bugs.digium.com.
--Eric
From: Tais M. Hansen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 9:20 AM
Subject: Re: [Asterisk-Users] Echo worsens in 0.7.1
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 30 January 2004 00:57, Eric
Michael,
I have the same thing -- 1 to 4 posts a day !
regards,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael Graves
Sent: Thursday, January 29, 2004 12:06 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] List traffic
All of a sudden my
Hello All:
Has anyone configured a meetme conference to use video?
I have successfully used video phones to talk through *, but I cannot seem
to get video when those phones dial into a meetme conference.
Is there something else that I need to be doing other than set the v flag
on my extension
On Fri, Jan 30, 2004 at 12:21:49PM -0500, Stephen R. Besch wrote:
I just fetched today's cvs (1/30/04 11:10:31). Compiles/installs on my
test machine (ASUS A7V, 900 MHZ). However, If I try to compile on my
production machine (Elite K7S5A, 2.4GHz, 512MB) while * is running the
zaptel and
I tried both featd and em in zapata.conf, to no avail. I restarted in between all
changes. Is it possible to signal the DPO ports on the 750 with fxo_ls or fxo_ks?
This is the last piece to my DID puzzle. Anyone else with experience on this oddball
config?
Thanks,
-sb
-Original
On Fri, Jan 30, 2004 at 10:58:15AM -0600, Eric Wieling wrote:
Are you sure this works for VoiceTronix Driver? It's not implemented in
app_dial, but in chan_zap.
I've only used it with a zap device. Sorry I didn't think this through.
On Fri, 2004-01-30 at 10:15, Walker Haddock wrote:
On
Regovich, Timothy wrote:
Hello All:
Has anyone configured a meetme conference to use video?
I have successfully used video phones to talk through *, but I cannot seem
to get video when those phones dial into a meetme conference.
What video phone did you use?
Thanks John,
I think it is not that simple. I am not using a phone but a Cisco ATA.
The scenario: -
User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100
(FXO))--Cisco ATA--Asterisk--Any extension
The Multitech MVP100 used to connect to my old analogue switch which was set
to
I have written my own. Java(JMF) based.
It is pretty rudimentary, but does handle audio (gsm, ulaw) and video (jpeg
and H263).
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Friday, January 30, 2004 1:30 PM
To: [EMAIL PROTECTED]
I would also be interested in this in regards to working with D-Link
videophones. They use the same setup as netmeeting h.263, but with another rfc
add on. I know current OpenH323 configs do not quite work with it, but I saw a
post that it is in cvs working using a patch to ffmpeg.
--
Jonathan
I've got a Newbridge CB hanging on the wall not being used right now and
I'd like to hear opinions on using it with Asterisk. If anyone has a manual
for it I'd like to get a copy of it. I tried the googling approach but
turned up nothing much except a Tech manual if I want to change out control
Thanks John,
I think it is not that simple. I am not using a phone but a Cisco ATA.
The scenario: -
User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100
(FXO))--Cisco ATA--Asterisk--Any extension
Any reason you can't use the H.323 load for the MVP200? I've not tried it
Did you say you were using Adtran FXS cards?
Bisker, Scott (7805) wrote:
Hello All,
I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 EM wink lines from telco. I have them all plugging into an Adtran 750
I was wondering if it was supported, and how.
It seems to me that video conferencing is a different beast than audio
conferencing because you cannot simply mix video like you can mix audio.
The conferencing server would have to
1) mix the video by creating one aggregate outbound paneled type
Yes, it is that simple, but of course there is a precursor
requirement that you need to 1) be able to configure your ATA, and
2) know how to configure your ATA.
Google is your friend. Please use Google before you reply back with
additional questions; it saves us all time and email bandwidth.
Yes. Adtran FXS cards.
Did you say you were using Adtran FXS cards?
Bisker, Scott (7805) wrote:
Hello All,
I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't
configured properly. Now heres the next question. 12 EM wink lines from telco. I
have
Hi Rattana,
Do you have jitterbuffer enabled?
Dan
On Fri, 2004-01-30 at 13:40, Rattana BIV wrote:
hi,
I use IAX softphone with asterisk and I notice that a call between two
IAX softphones end after 1 min. Then I can't hear anything but the
call still in progress.
I have this log in
Rob Fugina wrote:
[snip]
Is there a way to safely compile while * is running, so that I can
minimize down time of the server?
Seg faulting compiles usually indicate a memory problem on the
machine. Not lack of size, but bad memory, badly seated memory,
etc... There's no reason asterisk
I
purchased a license from Digium, If you ask they will can also give you a trial
license to test out.
Wes
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Jess
MagnayeSent: Friday, January 30, 2004 10:29 AMTo:
[EMAIL PROTECTED]Subject:
On Fri, 2004-01-30 at 14:26, David Gomillion wrote:
Rob Fugina wrote:
Seg faulting compiles usually indicate a memory problem on the
machine. Not lack of size, but bad memory, badly seated memory,
etc... There's no reason asterisk running, or the drivers being
loaded, should
cause a
On Fri, 2004-01-30 at 13:26, David Gomillion wrote:
Rob Fugina wrote:
[snip]
Is there a way to safely compile while * is running, so that I can
minimize down time of the server?
Seg faulting compiles usually indicate a memory problem on the
machine. Not lack of size, but bad memory,
Hi,
I have a wildcard x100p. I just installed asterisk by
following step:
# cd ../zaptel
# make clean ; make install
# cd ../libpri
# make clean ; make install
# cd ../asterisk
# make clean ; make install
# make samples
When I test Asterisk typing
# asterisk c
I find one error and one
Title: X-Lite, X100P, and Speex
I'm having a problem with using X-Lite to initiate a call via Asterisk out an X100P analog port, using the Speex codec. I've put in the registry fix for X-Lite and Speex so that works OK, and calling the echo test extension works. However, if I call out the
Dear all,
I have the following lines in my extentions.conf file;
;All US Calls
exten = _9001XX,1,Dial(IAX2/dornoch:[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
;Dial 9 for outgoing numbers
exten =_9.,1,Dial(Zap/g1/${EXTEN:1})
;include Brunswick
switch =
That's one of the things that's been on our (1control, I have nothing to
do with Digium) wishlist/to do list that just hasn't gotten done yet.
Currently, video in meetme is not supported. What we experience is the
audio will conference with the other audio streams but the video just
freezes.
On Fri, 2004-01-30 at 14:00, Shad Mortazavi wrote:
Dear all,
I have the following lines in my extentions.conf file;
;All US Calls
exten =
_9001XX,1,Dial(IAX2/dornoch:[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
;Dial 9 for outgoing numbers
exten
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