Hi
I have also have the Sipura rebooting itself.
I changed the codec from G723.1 to G729 and this seems to have
helped fix the problem.
I have the latest firmware...2.0.10(e) I think..??
Hope this helpsstrange stuff though.
regards
Clive
On 14 Oct 2004 at 14:48, Mike Benoit wrote:
I
On Thu, 14 Oct 2004, David McNett wrote:
On 14-Oct-2004, Kevin Walsh wrote:
Red Hat have embedded their trademark all over their Enterprise
editions so that they can restrict sales in that way. Red Hat still
have an obligation to release the various GPLed components as usual but
don't
On Thu, 14 Oct 2004, Joe Greco wrote:
RedHat further encumbers RHEL with a EULA which extends the GPL and
further restricts your rights to use the product.
That, then, sounds like it might be a violation of the GPL. The GPL
is, sadly, a maze of twisty little untested legal strategies,
How often was it rebooting before, do you know?
Mine seem to be rebooting almost exactly 1hour apart, which is the
registration expire time. I've just recently changed it to 6hrs, so I'll
see if that makes a difference.
On Fri, 2004-10-15 at 08:26 +0200, [EMAIL PROTECTED] wrote:
Hi
I have
Hi,
I know there has been some
discussions regarding how you get a softphone to work across the Cisco PIX
firewalls but I did not find any answer for the following
scenario.
Softphone X-Lite(Cisco
VPN client) - Connects to PIX --- Asterisk - SIP client
registered
The
Hi,
After reading up
on the Asterisk, I have a question:
1. Is there a
software phone running on PC as a client that is compatible with
Asterisk?
My reason for
asking is that I wonder if I can run voip end-to-end with Asterisk in between.
Diagram:
NetMeeting
-- IP -- Asterisk --
Hi
Mine used to reboot on every call
Clive
On 15 Oct 2004 at 0:15, Mike Benoit wrote:
How often was it rebooting before, do you know?
Mine seem to be rebooting almost exactly 1hour apart, which is the
registration expire time. I've just recently changed it to 6hrs, so I'll
see if that
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 3:17 PM
Subject: Re: [Asterisk-Users] Am I stupid or is my card DOA.?
|
| ** Niles Wrote
|
| | I had this exact same problem
Hi all,
I am trying to compile Asterisk beginning with zaptel.
Now I get 2 compile errors (see below).
Can anyone give me a hint?
Thanks
Franz
-
sip:/usr/src/zaptel # make install
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE
check out:
http://www.voip-info.org/wiki-Asterisk+phones
At 12:23 AM 10/15/2004 -0700, you wrote:
Hi,
After reading up on the Asterisk, I have a
question:
1. Is there a software phone running on PC as a client that
is compatible with Asterisk?
My reason for asking is that I wonder if I can
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark
Sent: 15 October 2004 02:41
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Looking for recommendations for a low-cost FXO
toIP gateway.
At first I thought the X100P was what I was looking for,
Hi,
I don't know if I missed something on the recent posts
regarding running * on the linksys boxes (couldn't
make any sense of the gifs that were posted??)?
Getting back to the original question, does anyone
know where the firmware or source for a linksys box
running * can be obtained?
Aaron
Hi there,
Cheers for your suggestions, would be great to see the output of some
other reports.
Logins and logouts are available within the engine, just need to
represent them in some way now. What do you suggest would be a good
format? Typical duration of login? Only problem might be where
Hi All,
I wanted to know, if it's possible to use any
available interface of Asterisk for
authentication/authroization to plug some external
billing application with it.
The CSV file option is quite good for postpaid
billing, but is there any way to do authentication of
a PBX extension before
Hi,
SIP - Asterisk - H323 Gateway
What is the configuration of asterisk to make a call with this plan .
My soft phone is registered to asterisk and i can establish a H323
channel from my asterisk to the Gateway.
It's just about Dial function in extension.conf or other !!
Thx
Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk?
It works in IAX mode, but in SIP mode I am unable to hear anything (no
dialtone, no voice). I am able to setup a conversation with another SIP
phone though (Xlite, Grandstream) and the other side can hear the
FireFly
I have noticed that the data/time on the caller-id is incorrect on my Motorola 5ghz soho phone system.
I go thru and reset the time on the handsets b/c there doesn't seem to be an option on the base for time. Then when I recieve a new call the caller-id is passed thru and the time gets screwed
On Fri, 15 Oct 2004 11:12:30 +0900, Benjamin on Asterisk Mailing Lists
wrote:
On Thu, 14 Oct 2004 16:50:39 -0400, steve szmidt [EMAIL PROTECTED] wrote:
Please don't use PPTP as a security solution, because it really isn't. It's so
flawed you can even connect to it without having ANY encryption.
I am having a problem with my new SNOM190 and my asterisk
box.
Incoming calls to the SNOM work perfectly, but when i
dial-out I get a "Not Found: number dialed" on the SNOM display
everytime I try, nothing shows up on the console of the asterisk box so its not
even touching it.
I have the
I stand corrected, I forgot about CAMA trunks because we almost always just
use PRI.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: Thursday, October 14, 2004 10:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
I've an Asterisk connected to an Ericsson MD-110 PBX using MFC/R2. It's
working fine but the caller id from the PBX to Asterisk is not set on
the calls. From the debug i can see that the ANI is received but not set
on the callerid field:
Oct 7 19:38:37 WARNING[18451]: Offered on channel 0
Hi all,
i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco
ATA-186 3.1.1 atamgcp
We are used to make an special ;) blind transfer like
(Flash)Number(Hangup before anyone answers or ring).
Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp
If one
i am sending this message to my asterisk
first message:
INFO sip:172.16.0.32 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.21
From: sip:[EMAIL PROTECTED]
To: sip:172.16.0.32
Call-ID: [EMAIL PROTECTED]
CSeq: 21 INFO
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/dtmf-relay
ContentLength: 26
At first I thought the X100P was what I was looking for, but now it
looks to me like the X100P does not have an IP interface, so it would
require all audio to run through the CPU. I'm familiar with ATA186's,
which I think are comparable to the IAXy box, and I'd just like to find
something like
Figured it out...the SPA does a NTP call to the server running ntpd. As long as the timezone offset is up to date the caller-id will display the appropriate time and update all the handsets with the correct time.
-Jeff
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Tom Ivar Helbekkmo wrote:
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED]
writes:
And how many routers and firewalls out there do support OpenVPN? Do
Cisco routers support it?
Neither I, nor anyone else here, seems to be saying that OpenVPN is a
replacement for IPsec. There's
It has all gone very quiet - I still need this...I spent a fair bit of
time looking at it but never got it to work. Needs someone with a bit
more of an understanding of Asterisk's architecture really. Also, it
should really go in app_dial so as to make it applicable across all
channel types.
On Fri, 2004-10-15 at 01:24 -0400, Donny Kavanagh wrote:
Do the dialogic drivers from digium require those lame redhat 7.2/7.3
only drivers that intel released?
Seeings as Digium just wrote a channel driver to connect the hardware
driver to asterisk, I would guestimate that that would be
Subject: Re: [Asterisk-Users] how can I test canreinvite effectivness?
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
Try IPTRAF or TCPDUMP.
Denis.
Em Qui 14 Out 2004
On Fri, 2004-10-15 at 10:15 +0200, Franz Edler wrote:
make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/usr/src/linux-2.6.5-7.108'
Makefile:438: .config: No such file or directory
Here's your hint. Did you make xconfig or menuconfig or something on
I have a question concerning re-INVITEs and how/why Asterisks sends
them. On SIP to SIP calls with asterisk set up with canreinvite=yes,
after a call is setup, media ip address:ports are renegotiated, 2 way
rtp is established, and then one of the parties hangs up by sending a
BYE, Asterisk goes
Subject: Re: [Asterisk-Users] how can I test canreinvite effectivness?
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
Try IPTRAF or TCPDUMP.
Denis.
Em Qui 14 Out 2004
I use my asterisk to SIP H323 Gateway.
Softphone SIP - Asterisk - H323 Gateway - cellular phone
I hear very well in my spftphone when i speak in my cellular
But when i speak in my softphone the sound is very very very bad
and i have this message in CLI console of asterisk :
codec_gsm.c:164
I can tell you that you are not alone. It's an issue I believe with Firefly,
and not in your configurations.
Message: 8
Date: Fri, 15 Oct 2004 13:06:17 +0200
From: Willem de Groot [EMAIL PROTECTED]
Subject: [Asterisk-Users] FireFly w/ SIP
To: Asterisk Users Mailing List - Non-Commercial
Hi all
When calls are set up using a macro, the extension in the status events coming from
action: status shows s. Does anybody know what to do to make the extension show
the correct value ?
My dialplan is like this:
[local]
exten = 8056,1,Macro(standardcall,SIP/t8)
exten =
Awesome!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart
Sent: Thursday, October 14, 2004 8:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FireFly SIP Registration Interval
We'll add that to next
I use FireFly w/ SIP all day long and it works great, except for the SIP
registration interval which I was just told will be fixed in next weeks
version.
Are you using GSM or g711u?
[remote-laptop]
context=remoteusers
type=friend
username=remote-laptop
secret=hiddenfromlist
qualify=yes
FireFly is awesome, it's not giving quality issues like X-Lite is. FireFly's
only problem was it wasn't registering with the server often enough, making
that NAT box forget the connection and not allow incoming streams.
Adam Hart said they would add it as an adjustable feature to the next
On Fri, 2004-10-15 at 10:15, Franz Edler wrote:
Hi all,
I am trying to compile Asterisk beginning with zaptel.
Now I get 2 compile errors (see below).
Can anyone give me a hint?
It would be nice if you did your homework before sending a msg to 8000+
people on this list. voip-info has all
Speaking from personal experience using Cisco Callmanager and Cisco
VPNs (not PIX, but Cisco VPNs hosted on routers with AIM cards), I can
say that this is possible- but it's not easy.
Essentially, the problem is not the VPN, it's NAT. In the cisco IP
Softphone client, there's a rather
That's pretty good..
I have a similar situation, where I need to match all the area codes in
a particular state like:
exten = _[904|321|407|252]XXX,1,Dial..
But it doesn't work. I can get it to work with something along the lines of:
exten - _[904|321|407|352]X.,1,Dial
But I was
I always get a 401 Unauthorized result before the registration succeedes on
these SIP phones. Is that normal? A REGISTER packet is sent, then a 100
Trying, then a 401 Unauthorized, then another REGISTER and another Trying,
then OK.
Is it normal to always get that 401? Why would registration be
Throughout the discussion about this problem, I've learned more or less
what the causes are. But.
is rfc3389 support planned?
thanks
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
[EMAIL PROTECTED] wrote:
Is it normal to always get that 401? Why would registration be
unauthorized then suddenly work? Or is this some algorithm that SIP
uses to try different auth schemes?
Im see this too. I think the RFC says the UA shoudl try first
without password, then with password.
Matthew Boehm wrote:
I always get a 401 Unauthorized result before the registration succeedes on
these SIP phones. Is that normal? A REGISTER packet is sent, then a 100
Trying, then a 401 Unauthorized, then another REGISTER and another Trying,
then OK.
I believe this is normal; most of the phones
Yeah that is totally normal.
To help prevent replay attacks the SIP device (Asterisk in this case)
includes a authentication header in the Authentication Required
response. This includes (among many other things) a random string that
the initiator of the request (your phone) must include when
I am trying to connect a Cisco 3640 terminating a PRI to * with SIP.
When I call into the PRI, the Cisco answers the call and sends it on to *,
however there is no audio. The clue is, the following message out of *:
Oct 15 07:50:58 NOTICE[1094289728]: chan_sip.c:2679 process_sdp: Content
is
Is there a way from the manager interface to obtain a listing of all the
channels (callers) in a queue? I know as they join/part events are
fired, but I'd like to obtain a listing of them when I connect to the
manager interface.
Any ideas how this can be done?
Cheers,
Griffin Internet
T: 0870
I don't know anything about config of a Cisco 3640, do have a Cisco 5350
and have never seen it send SIP messages with multipart payloads. So
can't really help you on that front.
However I can tell you what that means.
The INVITE request coming from the Cisco has
Jason T. Nelson [EMAIL PROTECTED] wrote:
In our last exciting episode, Kevin Walsh ([EMAIL PROTECTED]) said:
If GNU/Linux was licensed under a BSD-style license then Red Hat could
easily close the source - just as Apple did when they stole BSD code
to create their OS/X effort. I don't
Jason T. Nelson [EMAIL PROTECTED] wrote:
In our last exciting episode, Kevin Walsh ([EMAIL PROTECTED]) said:
If GNU/Linux was licensed under a BSD-style license then Red Hat could
easily close the source - just as Apple did when they stole BSD code
to create their OS/X effort. I don't
See if you have the below configure under your dial peers or voice
service voip.
If you do, then issue this command no signaling forward unconditional
signaling forward unconditional
Kurt
___
Asterisk-Users mailing list
[EMAIL
How about:
exten = s,1,GotoIf($[${CALLERIDNUM} : ^(904|321|407|252)[0-9]{7}$] ?
2:3)
exten = s,2,Goto(somewhere,s,1)
exten = s,3,DoWhateverElse
On Oct 15, 2004, at 7:21 AM, Ben Wern wrote:
That's pretty good..
I have a similar situation, where I need to match all the area codes
in a particular
You might have silence suppression turned on in the soft phone... turn
it off.
If that's not the culprit, use a different codec... maybe the soft
phone just doesn't speak GSM right.
On Oct 15, 2004, at 6:16 AM, CHAUVELIN Samuel wrote:
I use my asterisk to SIP H323 Gateway.
Softphone SIP -
Does the option A(filename) not work for you?
On Oct 15, 2004, at 5:38 AM, Robinson Tim-W10277 wrote:
It has all gone very quiet - I still need this...I spent a fair bit of
time looking at it but never got it to work. Needs someone with a bit
more of an understanding of Asterisk's architecture
Joe Greco [EMAIL PROTECTED] wrote:
The GPL protects the freedom of the source code and couldn't care less
about the freedom of those who would seek to close the code.
So, in other words, it's all right not to offer freedom to all.
No, in other words freedom must be protected against
On Fri, Oct 15, 2004 at 04:11:33PM +0100, Kevin Walsh wrote:
Perhaps steal was a bit harsh then. Maybe I should have said Apple,
Microsoft and others close the source with no compensation nor
recognition given to the original authors, as allowed by the stupid BSD
license. It's the authors'
On Thu, 14 Oct 2004, Joe Greco wrote:
RedHat further encumbers RHEL with a EULA which extends the GPL and
further restricts your rights to use the product.
That, then, sounds like it might be a violation of the GPL. The GPL
is, sadly, a maze of twisty little untested legal
Joe Greco [EMAIL PROTECTED] wrote:
By the way, assuming you've contributed code to Linux, did you get your
check from RedHat for RHEL? Thought not.
I was invited to take part in their IPO, under the friends of Red Hat
scheme, which made me over £120,000 profit on my investment. Does that
I'm using the ulaw codecs, and checking again, I just realized we have
one SPA-3000 in the mix behaving exactly like the SPA-2000's.
Changing the registration expire time to 6hrs didn't seem to make any
noticeable difference unfortunately.
On Fri, 2004-10-15 at 09:42 +0200, [EMAIL PROTECTED]
Hi all,
Anyone know what the differences are between the Prepaid and the
Prepaid-modified apps is? The provided docs don't say much.
David Filion
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
No - this plays the message AFTER the # is pressed, not before
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chad Scott
Sent: 15 October 2004 16:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zap
Hello,
Background: Old to UNIX Linus, New to list. A techie Dad that supports
local k-8 school that my kids go to.
More background: Recently the school wanted to put phones in all the
classrooms for teacher communications to/from the office. Another Dad in the
telecom business spec'ed out a
In our last exciting episode, Kevin Walsh ([EMAIL PROTECTED]) said:
Perhaps steal was a bit harsh then. Maybe I should have said Apple,
Microsoft and others close the source with no compensation nor
recognition given to the original authors, as allowed by the stupid BSD
license. It's the
Joe Greco [EMAIL PROTECTED] wrote:
The GPL protects the freedom of the source code and couldn't care less
about the freedom of those who would seek to close the code.
So, in other words, it's all right not to offer freedom to all.
No, in other words freedom must be protected
You have more options than you know. You could go with a channel bank
if you want to keep support for the analog phones in the classrooms
now(my school had them) or you could goto the next step with the sip
phones. I have looked around and found a couple vendors to be fairly
inexpensive.
See comments inline...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stewart M. Ives
Sent: Friday, October 15, 2004 12:05 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] New Project - IP Phone Sources
Question: If I just want
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stewart M.
Ives
Sent: 15 October 2004 17:05
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] New Project - IP Phone Sources
Hello,
Background: Old to UNIX Linus, New to list. A techie Dad that supports
Question: If I just want to provide IP Telephony within the school and have
no
outside connections to the local phone system I suspect I can install
Asterisk
on a RH Linux server and plug in a bunch of IP Telephones on the network,
config it all and it will work. The only cost to the school would
Joe Greco [EMAIL PROTECTED] wrote:
By the way, assuming you've contributed code to Linux, did you get your
check from RedHat for RHEL? Thought not.
I was invited to take part in their IPO, under the friends of Red Hat
scheme, which made me over £120,000 profit on my investment. Does that
I am trying to call a my friend who has GS HandyTone-486 behind a
firewall but it goes to his voicemail straightway. Surprisingly, he can
call me fine. I also see that his device is properly registered. Can
anyone help me resolve this problem.
In my sip.conf I do have canreinvite=no and nat=yes.
Message: 6
Date: Fri, 15 Oct 2004 08:29:01 -0700
From: Chad Scott [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Zap Channel wait for #
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII;
In article [EMAIL PROTECTED],
Kevin Walsh [EMAIL PROTECTED] wrote:
That's not up to them to decide. Under the GPL, if you distribute
modified code then you must publish your enhancements for the benefit
of all. The team responsible for the core code can decide whether the
contributed code is
Salutations,
In hopes of accelerating the adoption of Asterisk and changing the
landscape of the small business marketplace, we are contributing our
administration interface to a new project that aims to bundle
best-of-breed applications to produce a canned (but fully functional)
turnkey small
Ben
Check out Action: QueueStatus - it'll list the stats for each queue as well
as listing each queue member verbosely.
Cheers
Paul
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To
Hi Stewart,
Nice project! Something I'd certainly love to be doing myself. Anyway,
the following replies I've made to your questions are based on my
experience and past research. There may be better/cheaper alternatives.
In any case, I hope it helps:
On Fri, 2004-10-15 at 12:05 -0400, Stewart M.
Ah cheers,
It seems to have changed to add the event ' QueueEntry' from when I last
looked at the src.
Cheers for your help
Ben
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick
Sent: 15 October 2004 18:04
To: Asterisk Users Mailing List -
On Fri, 2004-10-15 at 18:56, Jason Becker wrote:
Salutations,
In hopes of accelerating the adoption of Asterisk and changing the
landscape of the small business marketplace, we are contributing our
administration interface to a new project that aims to bundle
best-of-breed applications
Tony Mountifield [EMAIL PROTECTED] wrote:
Kevin Walsh [EMAIL PROTECTED] wrote:
That's not up to them to decide. Under the GPL, if you distribute
modified code then you must publish your enhancements for the benefit
of all. The team responsible for the core code can decide whether the
Never mind, I found out what the problem was. On investigating the
response 415, I discovered that codecs could not be negotiated properly.
I changed the codecs on server and HandyTone, works great now.
-- sudhir
___
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[EMAIL
Patrick wrote:
Thank you very much for your contribution. Small remark: no website is
complete withoutscreenshots! :)
Regards,
Patrick
Hi Patrick,
Right you are!
We'll work on getting some up. In the mean time, have a look at:
http://www.voxbox.ca/products.php?display=4
The interface is
In our last exciting episode, Kevin Walsh ([EMAIL PROTECTED]) said:
Perhaps steal was a bit harsh then. Maybe I should have said Apple,
Microsoft and others close the source with no compensation nor
recognition given to the original authors, as allowed by the stupid BSD
license. It's
Hello all. I have a problem which I do not find a solution to. I need to
have a Plain jane analog phone when you pick it up With you dialing any
numbers (Dial Pad is broken) it dials automatically for you. This is going
to be for a door phone. Or in another case it's for a phone in an elevator.
BRILLIANT MOVE!
Kudos to you for your decision to contribute your efforts back into the
community!
Regards,
Jim Van Meggelen
Core Telecom Group
[EMAIL PROTECTED]
416-429-1304
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jason Becker
Sent:
Joe Greco [EMAIL PROTECTED] wrote:
Have you ever written code for something like a medical monitor? For
numerous reasons, you don't want that code available to the public. You
don't need some not-smart-enough hospital techie trying to make changes
to it, figuring out how to override the
Salutations,
In hopes of accelerating the adoption of Asterisk and changing the
landscape of the small business marketplace, we are contributing our
administration interface to a new project that aims to bundle
best-of-breed applications to produce a canned (but fully
I have a grandstream BT-486 in the lab running 1.0.5.11 firmware.
For the past three days I've had no trouble dialing out without hitting #.
I had the setting for using # as dial key to no in the config.
Today the BT wouldn't pass outgoing calls. I turned on # as dial key and it
works now if I
Here is the relevant dialplan:
exten = _3XXX,1,Dial(SIP/${EXTEN},15,tr)
exten = _3XXX,2,Voicemail([EMAIL PROTECTED])
exten = _3XXX,102,GotoIf($[${DIALSTATUS}==CHANUNAVAIL]?i,1:103)
exten = _3XXX,103,Voicemail([EMAIL PROTECTED])
exten = i,1,Playback(invalid)
exten = i,2,Hangup()
What
Perhaps steal was a bit harsh then. Maybe I should have said Apple,
Microsoft and others close the source with no compensation nor
recognition given to the original authors, as allowed by the stupid BSD
license. It's the authors' fault really. They live and learn.
Perhaps
On Fri, 2004-10-15 at 13:27, Ariel's Hotmail wrote:
Hello all. I have a problem which I do not find a solution to. I need to
have a Plain jane analog phone when you pick it up With you dialing any
numbers (Dial Pad is broken) it dials automatically for you. This is going
to be for a door
[default]
;
; Eric Wieling
;
exten = 2120,1,SetVar(DND=)
exten = 2120,2,SetVar(CFU_DEST=)
exten = 2120,3,SetVar(CFU_TIMEOUT=)
exten = 2120,4,SetVar(CFU_MESSAGE=)
exten = 2120,5,SetVar(CFU_FLAGS=)
exten = 2120,6,SetVar(CFU_LIMIT=)
exten = 2120,7,SetVar(DIAL_DEST=Zap/2)
exten =
I have been messing with the T100P card with and without data
for over a week now, and still to no avail.
Just got off the phone with our T1 provider to make sure our settings
were correct for the T1 in zaptel.
zaptel.conf:
span=0,1,0,esf,b8zs
nethdlc=1-20
fxsks=21-28
loadzone = us
exten = _3XXX,102,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?i,1:103)
Matthew Boehm wrote:
Here is the relevant dialplan:
exten = _3XXX,1,Dial(SIP/${EXTEN},15,tr)
exten = _3XXX,2,Voicemail([EMAIL PROTECTED])
exten = _3XXX,102,GotoIf($[${DIALSTATUS}==CHANUNAVAIL]?i,1:103)
exten =
Hi,
I'm going to setup asterisk as a voip gateway for remote internet
users. I'm going to use cisco 2600 for it with E1 interface cards. And
I have a few questions.
1. My provider will provide me with a couple of real phone numbers
(MSN it's called iirc), is there a way to assign these numbers to
On Fri, 2004-10-15 at 14:25 -0400, Cirelle Enterprises wrote:
I have been messing with the T100P card with and without data
for over a week now, and still to no avail.
Just got off the phone with our T1 provider to make sure our settings
were correct for the T1 in zaptel.
zaptel.conf:
On Fri, 15 Oct 2004 13:26:12 -0500, Eric Wieling [EMAIL PROTECTED] wrote:
[default]
;
; Eric Wieling
Eric,
Great stuff! I wish more people would post their configs. A lot can be
learned from examples. Maybe find a home on the wiki for this!
-Chuji
signalling=pri_cpe
callerid=asreceived
I see that I get the callerID CNAM in the cdr records, but the
same information does not show up on the display on my Cisco 7960
phone only the ANI. I do get Callerid from voip to voip calls .
Just not on the zap to voip calls.
My question is does
Brian West wrote:
Anyway we could talk you into releasing the source? I would love to see
wider codec support. And the ability to launch the URL sent with the IAX
call.
Brian,
The codec stuff I did, and the source is all available at
iaxclient.sf.net. Afaik, all the existing IAX
I have set up 2 * servers and connected them via IAX2, the connection
works, so far so good.
To optimize on the phone bill however I would like to have calls
that are local for the remote * server placed through the remote server.
How is this accomplished? I first tried the manual approach
On Fri, 2004-10-15 at 16:11 +0100, Kevin Walsh wrote:
Jason T. Nelson [EMAIL PROTECTED] wrote:
In our last exciting episode, Kevin Walsh ([EMAIL PROTECTED]) said:
If GNU/Linux was licensed under a BSD-style license then Red Hat could
easily close the source - just as Apple did when they
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