Hi
::: i have an MGCP voip phone (IPH-90), but i couldn't get it work with
asterisk.
::: i'm using asterisk 0.7.2 on openbsd 3.4
the config file and the debug infos are here:
http://nostromo.jol.hu/asterisk/
so not to flood the mailing list.
regards
wiking
I would like to call into my asterisk box and depends on extension, will call to different ATA (which has different SIP addresses). ATA will ring FXSFXO port converter and will allow me to dial a PSTN line that the FXSFXO porter converter is hooked up to. The problem is DTMF detection for the relay
Here is a copy of one that works perfectly.
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Gardiner
Sent: Friday, 2 April 2004 15:35
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] CISCO 7940 and directory/services problem
Thanks for
> I also tried pointing the Phone to the ones as listed on the
> voip-info site.
> This kinda worked, but most pages don't and the phone just
> tries and tries to get the page.
> If listing a directories page, it comes back with an error
> "CMXML Error"
>
> Any suggestions would help?
Telnet to th
Thanks for the responses on this issue, however, it appears after more
testing, that my phone is working FINE with typical MENU type pages,
however, when going to a DIRECTORY type XML page, it has the error as below.
Has any one else here with firmware 6.3 on a 7940 been able to get the
Directori
At 9:35 AM -0500 3/31/04, [EMAIL PROTECTED] wrote:
Hi Yawl,
I took delivery this morning of a used BetaBrite LED
display sign which I promptly set about playing with.
Having found a windows app that grabs XML headline
files from places like Slashdot and CNN as well as
stocks etc I had an idea.
Wha
At 7:54 AM -0600 3/31/04, Rich Adamson wrote:
From: Rich Adamson <[EMAIL PROTECTED]>
To: Asterisk-a-users-list <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Sip phone with push display?
Anyone know of a business class sip hard phone that includes a quality
display capable of supporting "push" data
(BHi
(B
(Bcould you please tell me your enviroment about this problem?
(B
(Band how do you fix it ?
(B
(Bplease disclose, when are you okay.
(B
(BRegards
(B
(Bmack_jpn
(B
(BOn Fri, 2 Apr 2004 12:40:52 +0900
$B4dED(B $B?-2p(B <[EMAIL PROTECTED]> wrote:
(B
(B> Hi.
(B>
(B> At l
Hi.
(B
(BAt last, I can compile asterisk.
(B
(BI had compiled low version of ncurses, glibc readline termcap and so on..
(BFinary, I coud compiled asterisk.
(B
(BThanks a lot!!
(B
(B
(B>
(B> Thanks for reply.
(B>
(B> Of course, I had already read ML like follows.
(B> This case error
I have quite successfully set up the Services button to work on the 7940
running SIP.
I have a metric-imperial converter, a foreign exchange rate calculator, a
calendar etc available to users.
The XML is really fussy though.
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Very cool and very fast! I've been playing with the AST GuiClient, but
this is actually more of what we need for the receptionist to be able to
see who's on their phone and who's free. Then she can just drag and
drop the call to transfer it. Thank you for submitting this!
-Original Message-
Hi all,
I've just started to put together a PBX system for my home office. I
have 2 x X100P cards and am waiting to get a hold of a TDM400P card.
What I am trying to setup is a solution as follows:
Line 1 is the home line, I want to give my DECT cordless phone system
it's own extension and thi
I downloaded iaxComm and get up my iax.conf file and the
extensions.conf. Here is the out but from CLI in iax debug. What did I
forget to do???
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 1ms SCall: 10489 DCall: 0 [192.168.50.66:4569]
U
how about for the stable version? im using 0.7.2.. is
there any known bugs in this release? should i upgrade to CVS?
At 01:55 AM 4/2/2004, you wrote:
On Thu, 2004-04-01 at 10:37, Sergi
Gabunia wrote:
> Hi,
>
> I have same problem with zap channels. I have E100P installed on my
asterisk
> box a
This rocks
Any way to display IAX2 channels?
I.E. I work from home and so don't have a normal phone, just an iax2
softphone.
What would I put in the cfg file?
I have tried:
IAX2/matt2
[EMAIL PROTECTED]
Is this possible?
Kind regards,
Matt Riddell
Title: ISDN BRI-U card suggestion for use in USA
Hello,
I'm looking for an ISDN BRI-U interface for use in the US. I'm primarily interested in using the BRI as a trunking interface into the PSTN with Asterisk. Naturally, cheaper is better.
I currently use a Nortel Norstar system with BRI
Il 01:02, venerdì 02 aprile 2004, John Vogel ha scritto:
> I am still experiencing the problem where you pick up an incoming analog
> call ringing on SIP Phone A with SIP Phone B using *8 but Phone A continues
> to ring. This happens with Grandstreams and Snoms on the 0.7.1 code base.
>
> My theory
Alex Volkov wrote:
That would be great. Steve, please also consider using sourceforge.net to
host the project.
Alex.
OK, I considered it.
No.
Steve, first thanks for the great work (especially the bugfixes).
As development on SoftFAX/spandsp is especially fast and from the
source it appears
Hi there,
I am attempting to set up a simple BRI and SIP based platform using *
with the quadbri card (it's not sharing an IRQ). I enclose my zaptel and
zapata.conf files. For the inital test I'm simply trying to connect to
the * demo menu.
The drivers compile (with a few warning that I believ
Okay, I've looked around for a FAQ on this, and it's kinda driving me up
the wall. I recently installed a new Asterisk system, and built mpg123 v.
0.59s from sources, and the audio comes out wy overmodulated.
Any thoughts?
___
Asterisk-Users mailin
Hello,
One of the Sipura 2k's I'm using has a dialtone that occasionally fades to
static when the user picks up the line. Are there any settings that I can
check that would affect this?
Regards,
Christopher
___
Asterisk-Users mailing list
[EMAIL PROT
you probably need to add a correct host entry in your /etc/hosts file for
your machine it goes
ip namealias
192.168.1.1 asterisk.goober.org asterisk
so
192.168.1.1 asteriskasterisk.googber.org
is n
John,
Yes, asterisk can do that, and in fact it's very simple. The problem at the moment is
your level of knowledge of asterisk, but this can be resolved...
There are a number of things you need:
1 Access to the PSTN - this can be done via a single X100P card (plugs into a standard
phone line
Hi,
From: Matt Lawson <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Where is the archive?
Date: Thu, 01 Apr 2004 14:20:04 -0500
http://www.mail-archive.com/[EMAIL PROTECTED]/maillist.html
only seems to go back a few days. Is there another archive somewhere that
goes back farther?
http://lists.
I received a recommendation to check out Asterisk, as a platform to host
a simple DTMF response system, something like:
Setup up VoIP endpoint on Linux/FreeBSD system
Answer incoming VoIP phone calls
User enters 100#, perl script plays back "foo"
User enters 101#, perl script plays back
actually figured this one out on my own, had incorrectly labeled
"secret" as "password" in iax.conf.
On Apr 1, 2004, at 6:14 PM, Steven Kokinos wrote:
Hello,
I've just compiled and configured my asterisk box, and so far things
have gone pretty smoothly. Now I am working on being able to make a
Hello,
I've just compiled and configured my asterisk box, and so far things
have gone pretty smoothly. Now I am working on being able to make a
call via voicepulse, and have run into an issue.
When I connect via my sip phone (using xlite) and dial a pstn number I
can see that it is being direc
I am still experiencing the problem where you pick up an incoming analog
call ringing on SIP Phone A with SIP Phone B using *8 but Phone A continues
to ring. This happens with Grandstreams and Snoms on the 0.7.1 code base.
My theory is that Asterisk is not telling Phone A to stop ringing when the
mmm... I just wondered, since it's very likely that most people ended up deleting it
*because* of the subject line. .. so it probably wont help ... well it might...
..When you dial the IAX(2) box you have 2 choices, stop using voicemail on that
machine or makeyour dial timeout less than the ri
Scott.
Please let us know if you are getting an answer off-list or if you
figure it out as I may experience the same issue soon, and I would
appreciate the information
Thank you in advance.
Thomas
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
htt
This I one of the things we have been looking for!!! I just installed it in
about 5mins and works great!!!. Excellent work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nicolas
Gudino
Sent: Thursday, April 01, 2004 2:52 PM
To: [EMAIL PROTECTED]
Subject: [
Thanks for the tip,
Ok I'll have a trie at this first thing tommorow morning,
Thanks,
Marc Dirix
On Fri, Apr 02, 2004 at 12:24:43AM +0200, Diego Ercolani wrote:
> Il 23:56, gioved? 01 aprile 2004, Marc Dirix ha scritto:
> > Hi,
> >
> >
> >
> > When I try to dialout over chan_capi everything
A while back, I asked about using Asterisk in a medical environment where the task
is to write a program that connects to a phone and sends a message like:
Hello Mrs. Jones. How are you doing today? Press 1 if you're
OK. Press 2 if you need help. Or start talking, and your
message w
It seems that the DG "gets lost" and keeps attempting to send RTP packs
to asterisk and it get an icmp deny.
The phones on that port will not work.
Other phones do.
So is this asterisk failing to hang up on the DG, or is DG not seeing a
"call over" message?
It is happening more frequently, but
On Thu, 2004-03-11 at 07:51, Zot O'Connor wrote:
> Is there an easy way to make the voicemail system say the extension
> number after the directory find (via name)?
>
> People want to know the extension once they have found the person to
> speed up the process.
>
> Thanks!
Actually I solved the
On Thu, 2004-04-01 at 04:09, Andy Powell wrote:
> Please don't tell me you deliberately used "LARGE BREASTS" as part of the subject
> for this...
>
I got got tired of asking questions that did not get answers while
watching people berate dead subjects or each other. The questions have
been tho
Il 23:56, giovedì 01 aprile 2004, Marc Dirix ha scritto:
> Hi,
>
>
>
> When I try to dialout over chan_capi everything works fine
> when I settle for
> msn=* in my capi.conf and use the primary msn of my ISDN-line.
> But trying to configure a different MSN the chan_capi doesn't dial
> and comes wit
Curious if anyone else has run into this.
I am testing this with a Sipura-2000 with firmware rev 1.0.33. On FC1 with
Asterisk 1.0 Stable branch 4/1/2004 CVS (no April fools jokes please :) )
The Sipura has the ability so when you dial *67 it turns ON CID block and *68
turns it back off. (This is
Dear Nicolas
i have try you demo from your site and i think is very great!! So i decide to
test on my simple pbx but ops.. some problem :-) i have create a subdir
called "astweb" in my html directory and put inside your "html" dir and the
perl script under dir "astweb/panel" this to hav
Hi,
When I try to dialout over chan_capi everything works fine
when I settle for
msn=* in my capi.conf and use the primary msn of my ISDN-line.
But trying to configure a different MSN the chan_capi doesn't dial
and comes with:
No one is available to answer at this time
What can be the prob?
Angus Berry wrote:
I haven't found this in any docs or faqs yet, so I'm wondering if I can
achieve what I would like to do.
On an Asterisk PBX with multiple PSTN lines, I'd like to call in from
one PSTN line, probably via cellphone and access the PBX as if I were
local to it. From here I'd like to
Yes you can, It's called DISA. Realize that using DISA has it's potential
security concerns.
>From the asterisk console type "show application DISA" for more information.
DISA = Direct Inward System Access
Ciao,
-bh
Quoting Angus Berry <[EMAIL PROTECTED]>:
> I haven't found this in any docs or
Hello-
Has anyone had experience connecting to a Marconi switch (in the UK) using
E1-PRI connections (TE410P)? In a new installation, my customer is getting
yellow alarms on every channel about every 30 seconds. These alarms clear
themselves immediately and then re-occur in another 30 seconds,
I have a regular conference call that I usually listen to and record, but
there are times when I can't be near an asterisk connected phone to call. In
those cases I'd like to place the call from the CLI and have it record
automatically.
The problem I'm having is that once the call is connected
*CLI> show application DISA
Ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus Berry
Sent: Thursday, April 01, 2004 1:08 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk call forwarding / remote dial-in/out?
I haven't found this in any
> On an Asterisk PBX with multiple PSTN lines, I'd like to call in from
> one PSTN line, probably via cellphone and access the PBX as if I were
> local to it. From here I'd like to get a dial tone and call back out. I
> know this isn't exactly call forwarding per se, but I'm wondering if
> this can
I'm trying to use iaxComm and I get the following error.
Apr 1 16:18:04 NOTICE[1142106560]: chan_iax2.c:3393 register_verify: No
registration for peer 'asterisk' (from x.x.x.x)
I'm VERY GREEN with this software so any help on list or off list would
be great
I haven't found this in any docs or faqs yet, so I'm wondering if I can
achieve what I would like to do.
On an Asterisk PBX with multiple PSTN lines, I'd like to call in from
one PSTN line, probably via cellphone and access the PBX as if I were
local to it. From here I'd like to get a dial tone an
Nice & elegant! Looks great.
jj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nicolas
Gudino
Sent: Thursday, April 01, 2004 1:52 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
http://sip.house.com.ar/operator
Its a
http://sip.house.com.ar/operator
Its a server/client combo that displays the status of your Asterisk PBX
in a web browser in real time.
You can also perform some actions. Hang-up channels and Transfers via
drag and drop.
The difference with other similar tools is that it displays status in
real
Hi,
I'm new to this list and also new to Asterisk.
I've set up asterisk in a way that my localclients connect to it thru IAXComm,
this works ok. In my asterisk server I have a Fritz!Card from AVM with the AVM Capi
driver.
People are able to call to my ISDN card from outside and they get connec
> The screen on the 7960 is a rather low resolution one, and therefore
> does not display much data. Pressing the "directory" button (and selecting
Resolutions and color depth on the phones are as follows:
7905/7912 192x53 Grayscale, Depth=1
7920 128x59 Grayscale, Depth=1
7940/
Thanks Scott - that worked.
James
Scott Laird wrote:
On Apr 1, 2004, at 10:48 AM, James Treleaven wrote:
Pardon my naivety - but how do I add a new tiki page? I want to add
one for 'wiki-Asterisk+cdr+pgsql'.
The usual way is to create a link to the new page from an existing page,
and then f
I have the same problem, it appears to be a problem with the echo
canceller. I have elected to install a DSP based T1 echo canceller, on
advice from TC.
Will report on how I make out.
Brent
On Thu, 1 Apr 2004, Justin Carlson wrote:
> how do you adjust ?
>
> -Original Message-
> From:
Hello,
I had heard that USB FXO/FXS are no good since linux has some problems
with USB and TDM? Can someone please throw some light on it? Does Linux
have any issue with USB (timing wise)?
Feel free to jump in if you have any feedback. Mark S. may answer this
better.
James
___
Hi, all
I have two computers. One has Asterisk(with X100P) on
the Redhat linux 9.0 and other is Windows XP. I want
to use Windows Messanger 6.1 to make or accept call
through. But I can config it. Is there anyone has this
configuration experience and can help me?
The IP address for Asterisk(with
On Thursday 01 April 2004 13:20, Matt Lawson wrote:
> I've been trying to search the archives for older messages, but the
> archive at:
>
> http://www.mail-archive.com/[EMAIL PROTECTED]/maillis
>t.html
>
> only seems to go back a few days. Is there another archive
> somewhere that goes back farthe
Hello,
From: pesb <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] H323 - SIP Interoperability
Date: Thu, 1 Apr 2004 12:37:17 -0300
So, I would like to call SIP/4 phone by dialing 014. Something like this:
exten => 01X,1,Dial(SIP/X) ; This is not working
How can I do that?
Try this:
exten => _01X,
I've been trying to search the archives for older messages, but the
archive at:
http://www.mail-archive.com/[EMAIL PROTECTED]/maillist.html
only seems to go back a few days. Is there another archive somewhere
that goes back farther?
___
Asterisk-Use
On Apr 1, 2004, at 10:48 AM, James Treleaven wrote:
Pardon my naivety - but how do I add a new tiki page? I want to add
one for 'wiki-Asterisk+cdr+pgsql'.
The usual way is to create a link to the new page from an existing
page, and then follow the link. I haven't tried it with this wiki,
thoug
Hi,
Pardon my naivety - but how do I add a new tiki page? I want to add one
for 'wiki-Asterisk+cdr+pgsql'.
thanks,
James
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
Just to confuse the issue a bit more, you _do_ use auth= and secret= in
the iax.conf, just not the sip.conf. Very subtle...
-Jeff
On Thu, 2004-04-01 at 10:38, Gregory Junker wrote:
> Oh for the love of...
>
> Well, that's what late nights do...
>
> Hey guess what...it registers now. ;)
>
> Th
There seems to be a problem with the rc14 zaphfc drivers. The latest
version RC16 seems to resolve the problem.
Rgds
Tim Robinson
zouhair echchelh wrote:
Hi,
Can someone tell what is this messages :
Apr 1 18:42:23 WARNING[147466]: chan_zap.c:6009 zt_pri_error: PRI:
received TEI check request f
how do you adjust ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Vogel
Sent: Thursday, April 01, 2004 12:25 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Echo's and dropped calls
Did you play with txgain and rxgain? Reduces echo but also v
That would be great. Steve, please also consider using sourceforge.net to
host the project.
Alex.
- Original Message -
From: "Jeb Campbell" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, April 01, 2004 12:01 PM
Subject: [Asterisk-Users] SoftFAX/spandsp cvs access
> Steve, f
Did you play with txgain and rxgain? Reduces echo but also volume -
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Justin Carlson
Sent: Thursday, April 01, 2004 10:21 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Echo's and dropped calls
on bo
on both the box with the zap interface and the remote office. it helped
some but the problem remains
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Vogel
Sent: Thursday, April 01, 2004 12:10 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Echo's
Do you have
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
In your zapata.conf file? Wiki is good for this -
John V.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Justin Carlson
Sent: Thursday, April 01, 2004 9:45 AM
To: [EMAIL PROTECTED]
our cvs is 02/25/04
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Juan J.
Sierralta P.
Sent: Thursday, April 01, 2004 11:56 AM
To: Asterisk Users
Subject: Re: [Asterisk-Users] Zap Channels Hang
On Thu, 2004-04-01 at 10:37, Sergi Gabunia wrote:
> Hi,
>
> I
On Thu, 2004-04-01 at 10:37, Sergi Gabunia wrote:
> Hi,
>
> I have same problem with zap channels. I have E100P installed on my asterisk
> box and I worked with CVS-02/22/04-16:30:20 and everything worked well (with
> Zap channels). I update asterisk to new cvs 2 days ago and incoming zap
> calls
Hi all,
I have a problem with echo and silence in the middle of calls. the echo
problem is that in the first 5 to 10 seconds of a call there is echo on the
sip side but not on the PSTN side, also the echo will randomly come back in
the call sometimes, I'd say 3 out of 10 calls. the other
Hello Again,
Does anyone here have experience with the Draytek Vigor G2600 or G2900
routers? They are said to provide traffic shaping, and look like a
decent replacement for my wonky Linksys BEFSR81.
Thanks,
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specia
chan_sip.c6524 reload_config= unable to get ip address from asterisk,
sip disabled
The ip address is working fine, Internet works great. Can anyone
help...Thanks
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/
Steve, first thanks for the great work (especially the bugfixes).
As development on SoftFAX/spandsp is especially fast and from the
source it appears that you are using version control, it would be very
nice for us users and testers to have read access to a repository.
My cvs/subversion is inte
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Doug Meredith wrote:
| "Jason A. Pattie" <[EMAIL PROTECTED]> wrote:
|
|
|>Is there any possibility to remove the "turnaround" leg or whatever its
|>called at the X100P? I'm just thinking of a scenario where none of the
|>outgoing signal is ever introdu
Hi,
Can someone tell what is this messages :
Apr 1 18:42:23 WARNING[147466]: chan_zap.c:6009 zt_pri_error: PRI:
received TEI check request for TEI = 127
Sending TEI check resp ri=27219 tei=95
Apr 1 18:42:24 WARNING[147466]: chan_zap.c:6009 zt_pri_error: PRI:
received TEI check request for TEI = 1
Hi,
I have just built my home Asterisk box into a better PC that became
available (still only a P2 350 but it only has to manage 1 analog line)..
Anyway I have built it on Fedora Core 1.. I have an X100P and a TDM400P
(1 module installed)..
These cards were working fine in my older PC that was
Hi,
(B
(BCan someone tell what is this messages :
(B
(BApr 1 18:42:23 WARNING[147466]: chan_zap.c:6009 zt_pri_error: PRI:
(Breceived TEI check request for TEI = 127
(BSending TEI check resp ri=27219 tei=95
(BApr 1 18:42:24 WARNING[147466]: chan_zap.c:6009 zt_pri_error: PRI:
(Breceived TEI c
Hi,
I would like to log all outbound calls made by my extentions ... is
asterisk able to account to a radius server ?
What accounting options do I have in general ?
Tnx !
--
Best regards,
Alessiomailto:[EMAIL PROTECTED]
_
Thanks for reply.
(B
(BOf course, I had already read ML like follows.
(BThis case errors are almost same, perhaps.
(BBut, tell the truth, I can't understand What I have to do?
(BI had recompile some version automake source, and tried to recompile asterisk.
(BThe result does'nt go well.
(B
(
On Thu, 01 Apr 2004 05:51:15 +, Miguel Cavazos wrote
> Yes it works with normal digium hardware
>
> Miguel
> On Thu, 2004-04-01 at 08:51, Otto Krumm wrote:
> > I was wondering if anyone has setup an * connected to E1
> > in Mexico?, what card would you recomend and do you have some
Oh for the love of...
Well, that's what late nights do...
Hey guess what...it registers now. ;)
Thanks!
Greg
> > secret=xxx ; generated per instructions in the
> > Wiki
> >
>
> This auth-style is news to me. Where did you find it in the wiki?
> I only know the usage of
Hi,
> [8010]
> type=friend
> host=dynamic
> dtmfmode=inband
> username=gjunker
> auth=md5
> secret=xxx ; generated per instructions in the
> Wiki
>
This auth-style is news to me. Where did you find it in the wiki?
I only know the usage of
md5secret=
Configuration
TDM400 port <=> * <=> sip
phone
When I make a connection between these phones the
TDM400 can hear the SIP phone just fine and the SIP can hear the TDM400 port
just fine most of the time. But sporadically you can not hear the TDM400
on the SIP phone it sounds like the micro
I simply cannot get X-Lite (Windows) or SJ (Linux) softphones (the only
ones I have tried) to register with Asterisk on the LAN (no NAT, no
routers). I have looked at every conceivable archived message regarding
401 Unauthorized, SJPhone, etc., and have looked at every relevant
article in the Wiki
On Thu, 1 Apr 2004 07:22:52 -0800, John Vogel wrote:
>
>If price is an issue but you have the slots for 3 - 4 cards you could try
>DigitNetworks. Their X100P compatible cards are only $28.50 (US).
>
>My quick evaluation of the alternatives is:
>
>1. Adtran 750 channel bank or something similar. T
I am getting ready to do my first build on this product. It's just for use as an overglorified answering machine right now and I will most likely play with some of the SIP functionality.
My big question though, is how much disk space do messages take up on the system? Are there any published metr
put notransfer=yes into asterisk e don't work
I see the SER log e receive this message
t_reply: ACKs are not replied.
On Wed, 2004-03-31 at 15:48, Welesley Sibelson Dias wrote:
> Hi All.
> I'am using Asterisk with SER. I can make call between two internal VoIP
> gateways or from na internal to
I've using CVS-03/30/04-14:38:02
Not sure where else to get the version number.
-Original Message-
From: John Vogel [mailto:[EMAIL PROTECTED]
Sent: 01 April 2004 16:45
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Hangup not detected on X100P
What version of *? I'm using 0.7.1
What version of *? I'm using 0.7.1 and it still has occasional problems
detecting call hangup.
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, March 31, 2004 8:54 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-U
Actually, the short answer any more is yes, you can use a modem.
I know it is better for several reasons to use an actual Digium X100P.
The main reason being that supporting them is a very good thing. They
are the reason Asterisk exists. However, I see lots of messages in
various forums wanting
I am using the Monitor command as follows
exten => Monitor(wav|monitest|m)
Now this creates the in and out files in /var/spool/asterisk/monitor as
I would expect. However I expected 'm' to mix the files together and
create a merged file. This isn't happening. Am I doing the wrong thing
or expecti
Hi there,
I would like to communicate H323 IP phones with SIP phones. My H323
phones are registered to a gnugk GK, and the SIP phones are registered to a
asterisk SIP proxy.
I could not create a dialplan that works. Inside my extensions.conf file I
created the following two entrances:
If price is an issue but you have the slots for 3 - 4 cards you could try
DigitNetworks. Their X100P compatible cards are only $28.50 (US).
My quick evaluation of the alternatives is:
1. Adtran 750 channel bank or something similar. This can handle up to 24
FXO lines and converts them to T1 to
(B Hello !!
(B
(B>Yes, works fine here. I'm using a mixture of Cisco 7960's with G711 and
(B>a Snom 200 with G729, and both can use the MeetMe function just fine at
(B>the same time. I just tested now to validate it. I'm running
(B>CVS-03/20/04-11:54:56 stable.
(B
(B Thank you for teachin
Hi,
Thanks for the help. You were correct. There was some data missing in the
extension.conf file
I was able to call one SIP phone from the other. I was even able to call an
H323 IP phone registered to the gnugk GK (It has Asterisk registered to him
as a GW).
But, I have another problem rigth no
Greetings,
I have seen a few postings in the past regarding the interop of Asterisk and
the Cisco 7920 WiFi phone. To date, I have not seen a definitive method to
getting the phone working. Assuming someone has this actually working, can
that person step up and answer these questions.
1) What C
Hello,
From: "Hall, Eric M." <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Newbie
Date: Wed, 31 Mar 2004 22:18:56 -0500
Could I do things like call other ext on the system? Check Voice mail? I
would like to test this before I put money in cards I may not need. What
Software Phone app is pe
Hi,
I have same problem with zap channels. I have E100P installed on my asterisk
box and I worked with CVS-02/22/04-16:30:20 and everything worked well (with
Zap channels). I update asterisk to new cvs 2 days ago and incoming zap
calls starts hanging.
I have mgcp extensions defined in my extension
I ask because of seeing proprietary hardware such as Dialogic's
trumpting about howthey can detect European & US DTMF. In the past I've
had cheap voicemail cards that couldn't pick up DTMF from UK callers
across the pond.
On Thu, 2004-04-01 at 09:10, Steve Underwood wrote:
> Angus Berry wrote:
>
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