How to catch some incoming call
Date: Tue, 15 Jun 2004 19:37:28 +0100
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Hi all,
I'm configuring meetme in asterisk but no luck. I dont have a digium card,
so based on the wiki, i have to install the zaptel driver and enable the
ztdummy on it. I compiled, install the module and enable it on the
module.conf. When i tried to start the asterisk, asterisk wont start
Setting up a new system using Fedora Core 2..
Tried following the instruction below (from the mailing list archives)
that worked before..
cp configs/config-for-my-kernel .config
make oldconfig
make include/asm
make include/linux/version.h
make SUBDIRS=scripts
.. but now the FC2 kernel has been
I've been banging my head on this one for a few days and am quite stuck.
I've got a gatekeeper running and everything works there. Netmeeting works
calling other netmeeting clients. Netmeeting calling asterisk connects, but
netmeeting can't generate the signals to make the demo do anything other
Glynn Condez wrote:
Jun 16 15:36:56 WARNING[-1084989312]:
/usr/lib/asterisk/modules/app_meetme.so: undefined symbol: ast_moh_stop
Jun 16 15:36:56 WARNING[-1084989312]: Loading module app_meetme.so failed!
It is not blatantly obvious? Music On hold has to be loaded prior to
app_meetme.
Also, you
Update to version 0.6.2a. It compiles with today's Asterisk CVS HEAD.
http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.6.2a.tar.gz
Michael.
Michael M. Saunders wrote:
The other problem is that version. Doesn't seem to work well with e1'. I
rephrase it changes
I think I do agree with your assessment that *BSD are more stable that
linux, no disrespect meant to linux as I think it is wonderful in its own
right.
What version of FreeBSD ? BSD you are using ? I am looking to build an
athlon 64 server soon and am wondering if FreeBSD would be a better
I managed to get sms text messaging working and integrated with our system.
This was my last major task.
I would like to take this opportunity and thank everyone involved in asterisk. I've been able toautomate my business in every way Iwished.
When I've got a few quid I won't forget you!Want to
Hi there,
our old pbx contract will expire soon and we have now 2 pbx companies trying hard to
sell us a new VoIP one. I am usually a friend of open source software and read about
the Asterisk poject, but I can not assess if Asterisk fulfills all our requirements.
It would be great to
Hi,
Maybe, maybe when the IPs on dynamic servers change, * has different
information internally hence the transfer fails?
My feeling is that you have a firewall/NAT issue.
Sorry - I know positively it's not. There's NO NAT between any of the
asterisk boxes - none whatsoever. And the box
I have TDM400P , with 1 FXS and 1FXO
I'm tring to forward all incoming calls to a SIP phone
in the context where all calls from the fxo come i have :
exten = s,1,dial(SIP/phone1000,5)
the phone rings but when i answer the sip phone ( phone1000 ) is connected
but the phone from which i'm ringing
Fancy knocking up a howto for it?
From:
Gary Ruddock [mailto:[EMAIL PROTECTED]
Sent: 16 June 2004 9:34 AM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SMS
in the UK
I managed
to get sms text messaging working and integrated with our system.
This was
my last
Hi,
Now it is pretty obvious that my setup is ok since it work half the
times. Forbidding transfers is not a good idea - the whole point of this
setup is to be able to transfer since network wise the two systems with
dynamic addresses are much closer to each other (ping round-trip of
about
Hi,
I have a problem with call forwarding.
When I call forward I need to forward the call with the callerid on the
called phone, not the callers.
How do I do this in a smart way ?
-- Called 89
-- Got SIP response 302 Moved Temporarily back from x.x.x.248
-- Now
Hi
I thought this might be of general interest.
Recently I purchased an X100P from a Digium reseller in the UK. Very
pleased with the card; works perfectly. My friend (known for his deep
pockets and short arms) purchased an X100P card from Ebay. He's had no end
of problems with line noise,
Hi all,
I got to make the conference work by the help of Jeremy and Philipp, thanks
guys.
Right now im having a problem again, my server hang up after a user exit on
a conversation in a conference.
unloading the modules ztdummy the server works very well. what should be the
problem, im using
This is a digium list so I'm not going to promote other clone cards but I
will say there are cards out there that are much closer match to the digium
but probably have not gone through the regulatory european certification
procedure which the digium cards have been through.
- Original
I'm using both for the past one year.
Facing Obsolutely *ZERO PROBLEMS* on both.
As a matter of fact I always like to spend less money.
I don't mind whether you put a DSP chip or no chips.
- Original Message -
From: Matt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 16,
On Wed, 16 Jun 2004 11:20:25 +0100, Matt [EMAIL PROTECTED] wrote:
I've got a couple of high quality (4000x3000 17mb JPG or 40+mb RAW)
images
if anyone wants to compare have a look - for obvious reasons I though it
best not to post them to the list ;-)
I wouldn't mind a look at the eBay one,
Hi,
I have seen a couple of scripts that should be able to remotely reboot the 79xx
phones, but I haven't been able to make it work for my 7940.
Anyone able to guide me in the right direction?
I am running the SIP 7.1 firmware.
--
Med venlig hilsen / Best regards
Michael Løjtnant - Systems
I wouldn't mind a look at the eBay one, perhaps at a smaller res though
(1024x768 ?)
I've got one of them waiting to be fitted, and I'd be interested to see
if it's the same one.
--
-S
Me too. I'm using a clone X101P I bought from Goods2World a while back
and get occasional dropped calls, so
On Tuesday 15 June 2004 20:45, Aaron J. Angel wrote:
And for those of you who don't like HTML email with different fonts or
colors, etc., there's this thing called CSS .
And for the rest of us, there's /dev/null which is where html email belongs.
If you expect everyone else to use a client
I'm looking to program some sort of web-services function: user presses
a button and send some info to a web server or scripting program. The
web server or script returns some text and/or imagery for the screen.
Lather, rinse, repeat.
I saw in section 3.7.1 of the manual referenced below that
On Jun 15, 2004, at 9:45 PM, Billy Huddleston wrote:
Yes, I use it. Here's a sample extension of how to use it.
exten = 1234,1,Answer()
exten = 1234,2,MailboxExists(1234)
exten = 1234,3,Dial(SIP/1234,20) ; Try to ring for 20 seconds, no
answer
goto voicemail
exten = 1234,4,Voicemail(b1234) ; send
There was some discussion on this list recently about the voiptalk silver
service. I've just had an e-mail from them saying that the price has been
reduced to 2.99 per month. However, they still only provide an 0870 number
whereas pipecall provide a local call rate 0845 number in the fee.
Chris
I have been using ohphone once in a while (the last i tried was 1.4.1).
The first channel driver i tried was asterisk-oh323, which gave me very
poor results (asterisk core dumping, if i got a connect i had echo + 10
secs delay on a Lan connection, stuff like that).
With chan_h323 i get pretty
On Wed, 2004-06-16 at 11:06, [EMAIL PROTECTED] wrote:
Hi there,
our old pbx contract will expire soon and we have now 2 pbx companies trying hard to
sell us a new VoIP one. I am usually a friend of open source software and read about
the Asterisk poject, but I can not assess if Asterisk
Have recompiled a few times any ideas?
*CLI oh323 debug toggle
Verbose debug info for OpenH323 channel turned on.
*CLI Jun 17 23:28:55 ERROR[20499]: chan_oh323.c:2297 ast_oh323_new:
Internal channel initialization failed. Bad binary?
*CLI set verbose 4
*CLI Jun 17 23:29:24 ERROR[21523]:
Got one weird one and one prob easy one.
1. I have upgraded our BT101's to Program--1.0.5.0Bootloader--1.0.0.17
HTML--1.0.0.34VOC--1.0.0.6
after doing this i have some phones on different subnet's ie 255.255.255.248
or .192 or .252 and i am now unable to login to these phones from
I
was interested in it too, but it's of little use with anything other
than home * installs. I've got a client that I'm about to put in a *
server for, and he'll need up to 4 concurrent outbound calls. When I
asked, I was told that you could not have more than one concurrent call
on any one
I only have experience with the first problem. This happend to me using
netscape on a linux box but has never happend using IE on a windows machine.
- Original Message -
From: Simon [EMAIL PROTECTED]
To: Asterisk-Users [EMAIL PROTECTED]
Sent: Wednesday, June 16, 2004 9:14 AM
Subject:
Have you enabled non-default compiler flags in Asterisk's
top-level Makefile (e.g DEBUG_THREAD)?
Michael.
Michael M. Saunders wrote:
Have recompiled a few times any ideas?
*CLI oh323 debug toggle
Verbose debug info for OpenH323 channel turned on.
*CLI Jun 17 23:28:55 ERROR[20499]:
Hi!
after doing this i have some phones on different subnet's ie 255.255.255.248
or .192 or .252 and i am now unable to login to these phones from different
subnet's . I have one at home which is on a .248 ( Using an external IP for
the phone ) i can access this from my home network ( the
Aaron J. Angel wrote:
Perhaps I've missed something, but I can't find any more info
on the Busy() and Congestion() applications. I have a
standard extension macro set up that calls the Busy()
application when a station is unreachable. For whatever
reason, the IAX channel decides to hang-up
I was wondering if there was a way of setting up the dialplan in a way
that if you dial an extension that is NOT in the dialplan then it would
play a not-in-service gsm file and then play congestion tones. I would
rather like this better than just hearing a busy signal on my phones.. I
DID
Set up a general pattern match with the message and congestion.
Extension pattern matching looks for the most specific match in any one
context. So if a specific extension is not found, it will take the
general pattern.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On Wed, Jun 16, 2004 at 11:35:45AM -0300, Ray Burkholder wrote:
Set up a general pattern match with the message and congestion.
Extension pattern matching looks for the most specific match in any one
context. So if a specific extension is not found, it will take the
general pattern.
ive had enough of cisco unity and microsoft exchange and im looking for
alternatives to our voip system. right now, we have 3 cisco callmanagers, 1
cisco ip icd system, and 1 cisco unity voicemail system. all phones are
cisco 7940/7960's and some ata186/188's. voice gateways are cisco
I have several SNOM200 phones at various remote locations all behind some
kind of NAT. Unfortunately I see where most of the phones go from REACHABLE
to UNREACHABLE quite often. Is there anything that I can change to help with
this issue. Length of registration? qualify time? Any help would be
Hi,
one month ago, I announced, that I will look at the openss7
project in order to use it together with asterisk.
It took a while for me to check the capabilities
of and around the project. Since the openss7 project consists
of only one person, and when there was just silence for a long
time in
Ray Burkholder wrote:
I was wondering if there was a way of setting up the dialplan in a
way that if you dial an extension that is NOT in the dialplan then
it would play a not-in-service gsm file and then play congestion
tones. I would rather like this better than just hearing a busy
signal
Roger Schreiter wrote:
Hi,
one month ago, I announced, that I will look at the openss7 project
in order to use it together with asterisk.
It took a while for me to check the capabilities
of and around the project. Since the openss7 project consists of only
one person, and when there was
Kevin Walsh mailto:[EMAIL PROTECTED] scribbled on Tuesday, June 15,
2004 2:32 PM:
If you use Microsoft Outlook then you might find this utility
interesting:
http://home.in.tum.de/~jain/software/outlook-quotefix/
Just a note for those using Word as their editor (an option you can
I had a tough time with this too, but I see the logic now...
FWIW Here's what I've found to work. At least with SIP channels, Asterisk
doesn't seem to want to go to the 'i' extension on invalid when you make a
general pattern. And I see the logic in this because SIP channels are
virtual and you
Here is the mail I just received from VoicePulse
Hello Steve,
Thank you for contacting VoicePulse.
The issue with VoicePulse Connect! has been resolved. Please verify that
Connect! is working. Our engineers are working to add more servers in the
next few days to handle the increased call volume.
I was able to get vg200 (pots) no pris to work with Asterisk. I was unable
to get mgcp to work correctly.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chad Whitten
Sent: Wednesday, June 16, 2004 9:54 AM
To: [EMAIL PROTECTED]
Subject:
Senad Jordanovic schrieb:
...
Are you aware of bounties posted to get SS7 working with *?
If not look at http://bugs.diuim.com .
...
No, I'm not.
Unfortunately, the link you mention, does not work.
I tried http://bugs.digium.com and found a login form.
I logged in with anonymous and looked
I know the horse is dead, but I still want to get a couple of hits in.
I've been using the usenet/internet since the early 90s, and I've had my
share of newsreaders, mailclients, etc. I've been active on the web
with my own sites since 94/95.
Yet... Gasp!... I PREFER TOP-POSTING (and I will
Tell the SNOM phones to re-register more often, maybe every 15-30
seconds. This should keep the NAT open for them.
-Original Message-
From: Brian Rathman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 16, 2004 9:56 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] NAT and Qualify
Hi,
Im about a week old with * and was hoping someone can push me in the right
direction. I am trying to send pstn calls to my cisco gateway via SIP. Ive
read documentation at
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sip.conf which
says I need to create a Peer in the
Roger Schreiter wrote:
Senad Jordanovic schrieb:
...
Are you aware of bounties posted to get SS7 working with *? If not
look at http://bugs.diuim.com .
...
No, I'm not.
Unfortunately, the link you mention, does not work.
I tried http://bugs.digium.com and found a login form.
I logged
Hi,
I'm still hassling with the consultative/attended transfer stuff. Someone
please help me identify this
A lot has already been said about the ATA186. Some report it works fine,
others say it doesn't. Lets get clarity on this.
My scenario is reasonably simple (I think)
Phone A: SIP/video1
On Wed, 2004-06-16 at 13:03, Roger Schreiter wrote:
Senad Jordanovic schrieb:
...
Are you aware of bounties posted to get SS7 working with *?
If not look at http://bugs.diuim.com .
...
Roger.
Or here:
http://www.voip-info.org/wiki-Asterisk+bounty+SS7
respectfully, Joseph - (606)
Martin List-Petersen wrote:
I have been using ohphone once in a while (the last i tried was 1.4.1).
The first channel driver i tried was asterisk-oh323, which gave me very
poor results (asterisk core dumping, if i got a connect i had echo + 10
secs delay on a Lan connection, stuff like that).
As
http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty
-Original Message-
Subject: RE: [Asterisk-Users] Status-info 1: Signalling C7 / SS7
No, I'm not.
Unfortunately, the link you mention, does not work.
I tried http://bugs.digium.com and found a login form.
Hi!this is the situation so far.the welltech 3804 is in peer mode, 2nddial is set to 2, the bureau table is pointing to extension 9 in extension.conf.we still have three problems!1. if I call from outside the 3804 call ext 9 and I can hear the asterisk voice telling me to dial the extension
Yes but I try that and it doesnt even go to it, I am trying to have the
invalid handler be executed when the extension a user tries to dial from
the SIP phone is not in any of the contexts (non-existant)... I've tried
this and placed it in several contexts and it does not work.
Steve
Rob
On 12:53 AM 6/16/2004, Jeremy McNamara wrote:
Glynn Condez wrote:
Jun 16 15:36:56 WARNING[-1084989312]:
/usr/lib/asterisk/modules/app_meetme.so: undefined symbol: ast_moh_stop
Jun 16 15:36:56 WARNING[-1084989312]: Loading module app_meetme.so failed!
It is not blatantly obvious? Music On hold
I took a little foray into pricing out IP Phones for my home pbx
yesterday. $75-$750 seems to be quite a range, so I took a closer look.
Cisco, for example, has different models such as the 7940 and 7960 which
seem to only differ in the software. And buying a Cisco 7920 should
cost you $500
- Original Message -
From: Roger Schreiter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 16, 2004 7:03 PM
Subject: Re: [Asterisk-Users] Status-info 1: Signalling C7 / SS7
Senad Jordanovic schrieb:
...
Are you aware of bounties posted to get SS7 working with *?
If
Hi,
I am interesting is there any way to use Cisco DSP Modules with
Linux?
Best Regards,
Miroslav Nachev
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 897-31-95
E-Mail: [EMAIL PROTECTED]
[EMAIL PROTECTED]
As I understand it, if I understand you correctly, the register
parameter is for the client side. The nat=yes parameter is for the
server side, so it has nothing to do with your register statement. The
sip debug displays no nat because sip.broadvoice.com is not behind
the nat, it's in front of
In article [EMAIL PROTECTED],
Brian Buhrow [EMAIL PROTECTED] wrote:
Hello. I've seen this behavior. What happens is that the
Grandstreams forget to continue registering with Asterisk after a while. I
bet when you find this happening, that sip show peers doesn't show ext/ext
ip address
On Wed, 16 Jun 2004, Nicholas Bachmann wrote:
You might try reading http://www.caliburn.nl/topposting.html -- it
explains why people don't like top posting.
Or read this quote:
A: Because we read from top to bottom, left to right.
Q: Why should i start my reply below the quoted text?
- --
On Wed, 2004-06-16 at 09:32, Stephen Rosebush wrote:
I was wondering if there was a way of setting up the dialplan in a way
that if you dial an extension that is NOT in the dialplan then it would
play a not-in-service gsm file and then play congestion tones. I would
rather like this better
Just to get an idea of hardware cost involved here:
- I can buy a 4-port router with built-in firewall, web-server and
email-client for $20-$30 RETAIL. That would indicate a hardware cost
of $10 max.
- I can purchase a Sipura SPA-2000 for $100 -- actual hardware cost
should be $50-$75.
On Tue, 15 Jun 2004, Lars Boegild Thomsen wrote:
Since only one of the asterisk servers are on a known IP, the two
systems on dymanic IP registers at the one in Europe.
Just one question: is there any reason not to use a dyndns name for these
two dynamic boxes? I believe they are PPPoE xDSL, so
Jay Milk [EMAIL PROTECTED] wrote:
I've been using the usenet/internet since the early 90s, and I've had my
share of newsreaders, mailclients, etc. I've been active on the web
with my own sites since 94/95.
Yet... Gasp!... I PREFER TOP-POSTING (and I will continue to do so).
Perhaps
exten =
_,3,Playback(/usr/src/test/asterisk-sounds/sounds/jedi-extension-tri
ck)
exten =
_,4,Playback(/usr/src/test/asterisk-sounds/sounds/please-try-again)
I'd love to get a copy of that jedi-extension-trick sound
W. Kevin Hunt
CCIE #11841
hi,
im looking at deploying asterisk in a small corporate enviroment which will
have approx. 1200 IP Phones and an average of about 100 to 200 calls at any
given time. The calls will be sent out SIP to my Cisco Gateway. Im
running Asterisk on a Dell Dual P3 1.2ghz running Fedora. Is there a
On Wed, 2004-06-16 at 18:34, Michael Manousos wrote:
Martin List-Petersen wrote:
I have been using ohphone once in a while (the last i tried was 1.4.1).
The first channel driver i tried was asterisk-oh323, which gave me very
poor results (asterisk core dumping, if i got a connect i had
Steve,
What I had problems using 'i' also, what worked for me was the
following...
For example, I have a bunch of extensions I'm matching on in a
particular context. Keep in mind, the stdexten macro you see below is
not defined properly, and is for example only.
[extensions]
exten =
Has anyone tried to run * on Soekris Engineering net4801 board?
If so, what were the results in terms of performance?
Ta
SJ
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
Michael Løjtnant wrote:
Hi,
I have seen a couple of scripts that should be able to remotely reboot the 79xx
phones, but I haven't been able to make it work for my 7940.
Anyone able to guide me in the right direction?
I am running the SIP 7.1 firmware.
Telnet to the phone's ip address, enter
haha that worked like a charm. thanks.
Brian D'Arcy wrote:
Steve,
What I had problems using 'i' also, what worked for me was the
following...
For example, I have a bunch of extensions I'm matching on in a
particular context. Keep in mind, the stdexten macro you see below is
not defined properly,
W. Kevin Hunt wrote:
exten =
_,3,Playback(/usr/src/test/asterisk-sounds/sounds/jedi-extension-tri
ck)
exten =
_,4,Playback(/usr/src/test/asterisk-sounds/sounds/please-try-again)
I'd love to get a copy of that jedi-extension-trick sound
That sounds like a good thing to use after
Cvs checkout asterisk-sounds
=)
Brian D'Arcy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of W. Kevin
Hunt
Sent: Wednesday, June 16, 2004 12:02 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Invalid Extensions -- More like
traditional PBX
Has anyone done any work on making the voicemail interface
user-configurable (and/or does anyone want to)? It's something I've given
some thought to and I have some ideas for and wouldn't mind working on,
but I wouldn't want to step on anyone's toes and would like to have some
people to
Am I dreaming?
Yes.
Community based development is too unreliable.
Just to refer to ongoing projects... Look at the farfon
(www.farfon.com), It's an active project in the final stages of development.
It offers the benefits (modular, programming of your own features,
quality components, low
I've been getting the same type of answers for the past month.
Assaf Benharoosh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, June 16, 2004 12:08 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Voicepulse Down
At 1:18 PM -0500 on 6/16/04, Jay Milk wrote:
I took a little foray into pricing out IP Phones for my home pbx
yesterday. $75-$750 seems to be quite a range, so I took a closer look.
[snip]
Am I dreaming?
Well, yes, you're sort-of dreaming.
The trick is not designing the hardware or the software -
I have a similar issue with Sipura using compact headers, but not with
regular headers. I am working on reproducing with the latest CVS.
Maybe you are using compact SIP headers on your ATA186?
http://bugs.digium.com/bug_view_page.php?bug_id=0001843
Stephen
-Original Message-
From:
Why is this conversation still taking place? It's a matter of personal
preference and I think it should be dropped.. If you don't like someone's
post, then don't read it. Problem solved.
I won't be replying to this topic again.. lets just drop it.
-Jon
-Original Message-
Jay Milk
Hi -A
Just an information about E1s.
The E1 use TS (time slot) from 0 to 31. TS 0 is for the synchronization, TS 1 to 15 are for Speech (B channels), TS 16 (D channel) is for the signalling, TS 17 to 31 are for Speech (B channels).
I hope this help you,
Angel.
On Tuesday 15 June 2004 03:17,
I have an asterisk server up and running, using Firefly in IAX mode
works great, even with Firefly behind a NAT (as expected, since IAX
works really well with NAT).
Now I'm trying to get X-Lite and/or Firefly to work in SIP mode from
behind the NAT, and I can't seem to get there.
At this
Following the installation directions on the wiki, I got festival built
and installed. However, when I hit it from my dialplan, I get:
Feature Token_Method not defined
I found only one reference to this error message in the archives and
there was no solution...
Thanks!
-Michael
Hi,
I have a nufone
connection (IAX2), works fine.
In my iax.conf I do
not specify a time interval that * needs to renew registrations with nufone
server.
However I can see
following registration messages on my cli every 90 seconds
(approximately)
--Registered to
'198.22.67.70', who
Thank u very much Rich!
I did what u suggested me, but im still having problems with the Mediatrix, actually i
dont have the MIbs for version 2.4.10.68, i tested 1204 with a different SIP server
called 3050 from Mitel www.mkcnetworks.com and it worked ok. Could help me with the
mediatrix
On Jun 16, 2004, at 11:18 AM, Jay Milk wrote:
Cisco, for example, has different models such as the 7940 and 7960
which
seem to only differ in the software.
IIRC, the 7940 and 7960 run the same software, but differ slightly in
hardware. The 60 has 6 line appearance buttons, while the 40 has 2.
John Todd wrote:
failed companies run by engineers. Selling direct is a limited market;
there are only so many Asterisk home users that you can advertise to via
the mailing lists. :-)
I think there are enough of us resellers/consultants around here to make
a very viable business for a decent
Michael Bielicki wrote:
- CTI support (dialing from within Outlook using hardware VoIP
phones)
there is a project for that which sems to work although we havem't
tested it yet
I'm using asttapi https://sourceforge.net/projects/asttapi/ and it works
fine.
Hi,
I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04).
These two boxes talk to eachother via sip, not iax. Since the upgrade, I
get the error Failed to authenticate on INVITE trying to make calls to/from
either box. Removing the secret from each box's sip config seems
Well, yes, you're sort-of dreaming.
The trick is not designing the hardware or the software - anyone with
$100k (or much, much less) and the right engineers can get something
working to the point where it is ready to be produced.
You will hit the wall with:
- finding reliable
Sounds
like a firewall issue to me. How does your FW handle
state.
TL
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
SathyaSent: Wednesday, June 16, 2004 2:23 PMTo:
[EMAIL PROTECTED] Digium. ComSubject: [Asterisk-Users] IAX
Kevin P. Fleming wrote:
I have an asterisk server up and running, using Firefly in IAX mode
works great, even with Firefly behind a NAT (as expected, since IAX
works really well with NAT).
I have the same scenario, but after about 4 hours, the Firefly phones can
still make calls, but asterisk
I've got Zaphfc working running Asterisk v.
0.7.2
Then I have tried with Asterisk V. 1.0 and the
latest from CVS - with no succes.
Has anybody got zaphfc working with newer version
than 0.7.2 ?
NR
If you take a look at http://sipp.sourceforge.net/ there is a utility
which claim to check the SIP performance of the specific system. (btw
don't try this on a target number which has voicemail, then the test
becomes a bit subjective ;)
I see asterisk more and more as real cool pbx with
Well if you just take a look at the sand that is needed to make the chips
you even get better prices...
Sand -- silicon -- chips -- PCB -- phone -- a lot of talking
It's not the material of the phone, it's the payroll of the people who make
the -- happen.-)
Never mind my rude simplification,
Hi,
We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54).
We've been having some serious problems:
1) All the phones randomly reboot themselves. Typically when trying to
answer or initiate a call.
2) All the phones will disconnect from a calls with the PSTN after 2-3
Claudio,
Claudio.loletti wrote:
Hi!
this is the situation so far.
the welltech 3804 is in peer mode, 2nddial is set to 2, the bureau table
is pointing to extension 9 in extension.conf.
Could you inform which firmware version are you using?
we still have three problems!
1. if I call from
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