with:
app_capiFax.c:34:34: asterisk/channel_pvt.h: No such file or directory
I haven't such file on my system!
Peraphs patches are for older CVS versions?
Look in the Makefile for a reference to app_capiFax and remove it.
Jason
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On Mar 31, 2005 1:00 PM, Robson Ribeiro [EMAIL PROTECTED] wrote:
Hi,
If I want a user to, while waiting for a transfer after responding to an IVR,
to listen to music instead of a ring sound, what is the change should i do in
extensions.conf? Is it on the IVR menu or on the optional
On Mar 31, 2005 12:31 PM, Marc SCHAEFER [EMAIL PROTECTED] wrote:
Hi,
is there a configuration in iax.conf to specify that if a call goes to
that peer, a second call should not be allowed.
Specifically, I do this:
Dial(IAX2/iaxcomm) # in extensions.conf for a specific extension
in
On Mar 31, 2005 1:05 PM, Muhammad Haris [EMAIL PROTECTED] wrote:
I've setup * with TDM400P w/1 FXS, 1 FXO modules.
I've one analog phone connected to TDM400P FXS module, 1 PSTN line to
one of the FXO module(ZAP) , and IP phone connected to asterisk on
LAN.
The calls between SIPs and zap
On Tue, 22 Mar 2005 10:45:42 -0800, Trevor Peirce [EMAIL PROTECTED] wrote:
Mike Dent wrote:
Hi,
the topic says it all really.
Does the Sipura 3000 detect and report UK clid correctly?
Yes it does
Jason
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do you really have
[specialized]
[specialized]
it is twice try removing one entry
Jason
On Wed, 23 Mar 2005 02:37:42 +, Scheda [EMAIL PROTECTED] wrote:
Hey, I'm currently using the GotoIf application to set it so if
certain caller ID's call my number, it will transfer it to my cell
= _X.,2,Playback(invalid)
exten = _X.,3,Playback(goodbye)
exten = _X.,4,Hangup()
And all should be good
Jason
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playing dial tone, this should do what you want
Jason
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of learning how to do
it. I can learn later... HEHE
Thanks again,
Jason
Red Hat Linux release 9 (Shrike)
Kernel \r on an \m
2 Each Digium T100X
(Snippets of my config files, some phone/server specific info changed for
post IE sip username/secret, CICCode and default IP)
-Extensions.conf-
[general
Do you remember what you actually changed to make it work cause that is the
same switch that I am dealing with myself if I am not mistaken.
Thank you,
Jason Miller
From: Dave Weis [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
in Advance,
Jason Miller
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. route regular TCP/IP traffic for any purpose to the established subnet.
Thank you in advance for your help.
Jason McAffee
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Anyone have experiece with polycom phones?
I am experiencing a really weird problem. In an
office where I have the following extensions:
100
101
102
103
104
110
111
120
130
140
141
150
200
On the Polycom phones, when I want to dial from
extension 100 to any extension 120 or above, or dial
OK I have a TDM400 with 3 incoming lines, with call
hunting. When u call any of the numbers it is supposed to goto our
greeting message. Now when you call the first number that is exactly what
happens. Problem is when you call either of the other 2 it just rings
forever and I see no
. for module in /lib/modules/`uname -r`/misc/*; do rm -i
/lib/modules/`uname -r`/extra/$(basename $module); done
followed up by a depmod.
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
.
The dial command will call more than one device
eg:
exten = 1234,1,Dial(SIP/1234SIP/2345)
extension 1234 will now ring sip device 1234 and 2345
Jason
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has it's own interrupt
I suspect not.
Jason
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(OH323/${EXTEN:2})
the :2 deletes the first 2 digits and removes the leading 40
Jason
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://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/85000
although I don't know if a bug was ever filed. I had a cursory look at
the time we were bitten by this but couldn't find one. Pulling a newer
CVS Stable and rebuilding resolved the issue.
Regards,
--
Jason Becker
Director CEO
Coalescent
ideas what I'm doing wrong?
Thanks in advance for any help.
Jason
This message along with any attachments is intended only for the use of the
individual or entity to which it was addressed. It may contain information
that is confidential and prohibited from disclosure. If you are not the
intended
and featdmf signaling in Zapata.conf to get ANI
on a t-1 span.
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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. For huge enterprise databases I use PostgreSQL.
Regards,
--
Jason Stewart | Tel: 616-532-2300
Systems Administrator/ | Fax: 616-532-3461
Programmer | Email: [EMAIL PROTECTED]
Right to Life of Michigan | Web: http://www.rtl.org
On Tue, 15 Mar 2005 14:13:40 -0500, Kanuri, Seshu (Company IT)
[EMAIL PROTECTED] wrote:
atxfer = *2 ; Attended transfer
Remove attended transfer capability and then you will be able o enter *2XXX
Jason
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a dialplan, you give access to other contexts with include
statements. With a properly segmented dialplan, you can accomplish your
goal very simply.
-Btw-welcome to the community and get to know
www.voip-info.org/wiki-asterisk It is your friend.
Jason Kawakami
www.optellabs.com
Salt Lake City
On Sun, 13 Mar 2005 21:49:52 +0200, Dimitris Kounalakis
[EMAIL PROTECTED] wrote:
Hello *Martijn,
Thank you for your response.
*That was my opinion too, it looses the context due to a bug, and can anyone
confirm it also?
But I have no output from the command Show channels, and it happens so
immediate=yes
;channel=1-3
channel=1
;group=2
;context=incoming_institute
;signalling=fxs_ks
;usecallerid=yes
;echotraining=yes
;immediate=yes
;channel=4
Thanks,
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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Chuck wrote:
does anyone know of a 2.4 or 5 ghz cordless phone system that has an ip
base station?
Uniden has the UIP1868:
http://www.uniden.com/productsupport2.cfm?product=UIP1868
But there's no documentation to speak of.
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc
On Fri, 11 Mar 2005 17:45:55 +0200, Calin Serbanescu [EMAIL PROTECTED] wrote:
hello list,
last night i upgraded my asterisk box from 1.0.5 to 1.0.6 and my music
on hold did not work anymore.
Download version 1.0.7 from Cvs this has the fixes in it
Modify the dialplan.xml on your tftp server to this
DIALTEMPLATE
TEMPLATE MATCH=* Timeout=4 User=Phone/
/DIALTEMPLATE
Jason
On Thu, 10 Mar 2005 10:31:34 -, Marshall, Ed
[EMAIL PROTECTED] wrote:
Hi There
I am currently having an issue with a Cisco 7960. The phone
try with a CVS head before 03/03/05 as the channel structure was
changed then, or get an updated version of asterisk-oh323 if there is
one availiable
Jason
On Thu, 10 Mar 2005 06:25:04 +0330, mohammad [EMAIL PROTECTED] wrote:
Hi;
I use the following asterisk, openh323, pwlib:
asterisk
On Thu, 10 Mar 2005 16:22:39 +0100, Deti Fliegl [EMAIL PROTECTED] wrote:
Hi,
how can I setup asterisk to use the number presentation bits on the isdn
side to suppress the number presentation? We need to transmit the
subscriber number for billing purposes via ISDN whether or not the user
On Tue, 08 Mar 2005 17:34:31 -0500, Jerry Geis [EMAIL PROTECTED] wrote:
SEPDEFAULT.cnf is not required nor recommended. The 7912 only uses the
gkMAC file
and the software version CP7912XXX file
The gk file must be lower case..
This phone 192.168.255.250 is requesting SEPXXX
It is
that the LCR programming determines is valid for that
provider.
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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Could you do something with the h (Calling party Hangup)
eg
exten = h,1,DoSomething
On Sun, 6 Mar 2005 15:00:23 -0500, George Burt [EMAIL PROTECTED] wrote:
I am trying to run a macro at the beginning of call and after the call is
terminated.
exten =
clears a call you
originated then the exchange you are connected to will hold up the
call for 20-30 seconds before disconnecting (this allows the called
party to place the phone on hook go to another phone and pickup the
call and resume the conversation. You will have no control over this.
Jason
play the same role and therefore
can't talk to each other.
-this is very true, however, the current version of the Axxess software
(9.0) supports SIP trunking natively on the IPRC. I just got my Axxess
upgraded and am salivating to get * connected to it.
Jason Kawakami
www.optellabs.com
Salt Lake
to the LEC so they are
damn well gonna charge you for reserving them. Usually, they are $0.15 per
number in a block and the block size varies.
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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cards in it. I tried to google for advice but
I didn't find anything that pertained to this.
-Thanks
Jason
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...
Site:lists.digium.com
That tells Google, to search only the pages from this email list.
Regards,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Hawthorne
Sent: Monday, March 07, 2005 11:50 AM
To: asterisk-users
Patch your chan_capi with this and you will be able to compile CVS
HEAD http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
Jason
On Thu, 03 Mar 2005 18:13:19 +0100, Massimo [EMAIL PROTECTED] wrote:
Hi,
I successfully installed asterisk 1.0 with Capi 0.35. In my pbx system I
The HEAD version was changed last night to be incompatible with the
patch I provided, My C skills are not good enough to fix this so you
need to checkout from cvs yesterdays code
cvs checkout -D 03/03/05 asterisk
Jason
On Fri, 04 Mar 2005 13:51:51 +0100, Massimo [EMAIL PROTECTED] wrote:
Hi
Try using the url
http://ip-of-machine/phpconfig/phpconfig.php
On Thu, 3 Mar 2005 12:45:03 +0300 (EAT), Julius Kidubuka
[EMAIL PROTECTED] wrote:
Hi,
I have just tried to get phpconfig to work but to no avail. In my browser
I type; http://ip-of-machine/phpconfig/ and this returns the
Contexts can be used to partition Asterisk, but the administration is
not multitenanted
On Thu, 3 Mar 2005 10:47:03 +0100, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi all,
Has any one tested or know if Asterisk support multitenant PBX, ie the
Asterisk
support either multiinstances on the
Try adding an exten = h,1,DoSomething
in the context
Jason
On Thu, 03 Mar 2005 14:40:10 +0100, administrator tootai
[EMAIL PROTECTED] wrote:
Hi everybody,
I'm running an IVR menu with different languages setted up by user when
they enter this menu. What I want is when they hangup
and google but cant find much related
to this problem.
Please read the following:
http://fedora.redhat.com/docs/selinux-faq-fc3/
and if you still have issues post to the amportal mailing list and/or
Help forum.
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
403.244.8089
written about
this?
-If you order an NI2 PRI you will be fine. That will give you name and
number, DID etc. if they want more than that, ESF,B8ZS, NI2, all b-channels
2-way DID.
Have fun!
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
it is possible.
-try that, it should work.
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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Im sorry if this has been asked before as I couldnt seem to find it.
I have an avaya partner ACS r3 system that I want to be able to hook
asterisk into with a x100p card, into and use asterisk to tie into a voip
provider then be able to dial (or connect) to an extension like an
intercom function
Im sorry if this has been asked before as I couldnt seem to find it.
I have an avaya partner ACS r3 system that I want to be able to hook
asterisk into with a x100p card, into and use asterisk to tie into a voip
provider then be able to dial (or connect) to an extension like an
intercom function
to the voip provider
Should work I think.
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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Hello,
Has anyone had any luck getting a D/41D ISA dialogic card working with *?
Thanks
Jason
This message along with any attachments is intended only for the use of the
individual or entity to which it was addressed. It may contain information
that is confidential and prohibited from
up and maintain. I am the only IT person in our organization that knows
how to work with Linux.
I am basically looking for some guidance here. Has anyone dealt with a
similar situation and how did you resolve it.
Any help would be appreciated.
-- Jason Fayre
Colorado Center for the Blind
be appreciated.
-you could use the existing analogs with a channel bank or some FXO gateways
but those options in the end will cost more money.
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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?
http://www.voicetronix.com.au/hda.htm
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
403.244.8089
www.coalescentsystems.ca
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. If
you didn't receive it ping me off-list and I'll resend.
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
403.244.8089
www.coalescentsystems.ca
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Hello,
It does appear to be an issue with the colon, as I ran
this test:
exten = _9X.,2,SetVar(REC_FILE_NAME=test)
exten = _9X.,3,Monitor(wav|${REC_FILE_NAME}|m)
and it worked fine. Indeed a colon is a valid
filename under Linux. So is this a bug?
Jason
--- Jim Van Meggelen [EMAIL PROTECTED
Hello,
I have been attempting to get the Monitor function to
accept a loal variable substitution in order to use
the same filename later in the same context. Monitor
does not appear to like it, as it attempts to use
wav|filename as the recording type, as opposed to just
wav.
Here is what I get
appreciated!
Thanks in advance,
Jason
--
Jason A. Crome
Senior Software Engineer, DEVNET, Inc.
E-Mail: [EMAIL PROTECTED]
http://www.devnetinc.com
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Are there any settings that I need to change on the phone to match this?
Thank you!
--
Jason A. Crome
Senior Software Engineer, DEVNET, Inc.
E-Mail: [EMAIL PROTECTED]
http://www.devnetinc.com
-Original Message-
From: [EMAIL PROTECTED
Hello,
There is no colon the filename below.
Jason
--- Jim Van Meggelen [EMAIL PROTECTED] wrote:
You are using illegal characters in your file name.
See this line in your output?
ast_writefile: No such format
'wav|rec_to_448704386865_at_16022005-16'
It can't get past it because
I'm having some problems getting meetme to work now that I have upgraded to
.5 I am able to conference calls but every time I try to manage the
conference through meetme it just says No users in this conference Any
ideas why it doesn't see the conference call?
Thanks for any help!
Jason
, then you
need to fix it right away before you get on umpteen million blackhole
lists.
Consult the docs and/or community for the MTA that you're using to fix
that.
Jason
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with some samples? Please? If I can get
one for 1NXXN. and 01144. I should be able to figure the rest out.
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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Firmware NO, a good skinny patch for running these phones.. (i have
two that work great)
http://www.blackratchet.org/chanskinnyplus/
Jason
On Sun, 06 Feb 2005 13:24:10 -0500, Mark Phillips [EMAIL PROTECTED] wrote:
Anyone know where I can lay my hands on some Skinny firmware for some
Cisco
Hello,
Newbie needs some help J
I read on the list of features for Asterisk that it can work
as a Conference bridge. Does anyone currently use this? How well
does it work compared to like an ATT conference bridge?
Thanks
Jason
This message along with any attachments
Revisiting this:I was able to get this to work as well, but voicemail doesnt work as intended. A user in companya-internal cannot get his voicemail when in the office. It gives login incorrect. However if the user dials in from outside through a zap channel the vm login works.Any
In order to put a shared pbx in an office building for
multiple businesses, I will have to make sure that the caller ID information
going out is correct.
i.e. company as main phone number is 5551212
company b is 5572121
company c is 5596767
Now I know how to distribute incoming
Pedro
Exactly my point. I have each company in a different
context. How do I SetCallerID to a number based on the context they are in?
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?
Is that accomplised wiht a single line? I guess this is done by the
voicemail picking up and the caller having to go through a menu to
get to the right VM. Something like Press 1 to get dad. Press 2 to
get Mom. Press 3 to get kid 1.
-yep
Good Luck!
Jason Kawakami
I have a 7960 desk phone and I'm running x-lite on my laptop. They are
both behind a NAT box so they would appear to * as being from the same
IP. I'm trying to make them ring at the same time but appear to
everyone as one extension. Is it possible to have them both register
to * as the same
-Original Message-
snip
Does anyone know what this might be and/or an easy way to have the ZAP
channel come off-hook, delay for 1/2 second or so, and then dial?
-look at the w option to the dial command on the wiki
Exten=???,1,Dial(Zap/G?/w${EXTEN})
Jason Kawakami
Has anyone tried Sipura products such as the 3000 in Japan?
Jason
Steven Critchfield wrote:
On Mon, 2005-01-31 at 12:53 +0900, Jason Frisch wrote:
Sorry for my ignorance, but what is J1? I actually hope to use Softbanks
fiber-based IPtel
service, but I believe they require VoIP TA so I guess
I'm using * 1.0.3 on Gentoo 2004.3, zaprtc from bri-stuff for timing.
When I put a caller on hold, the volume of the hold music in the
callers ear is extremely loud. I'm using the default entry from the
musiconhold.conf:
default = quietmp3:/var/lib/asterisk/mohmp3
Volumes with a called or
-Original Message-
snip
Based on the fact that the call center switch is connected to the PSTN with
2 E1s, and
only external incoming/outgoing calls are subject of recording, I thought
the Asterisk
could be a solution, in the following way:
- I set up an Asterisk-based switch with a
example but the second one is duck soup.
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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on how to
set it up, or which drivers to use.
Any help would be greatly appreciated!
Thanks
Jason
This message along with any attachments is intended only for the use of the
individual or entity to which it was addressed. It may contain information
that is confidential and prohibited from disclosure
So I have a problem. A customer of mine wants a PBX, owns an
office building. I want to sell him on asterisk. He has 4 tenants. I am
using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a
PRI card. Dont know if it makes any difference.
Anyway, I want to route
to have
4-5 outside lines.
Thanks!
Jason
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Sorry for my ignorance, but what is J1? I actually hope to use Softbanks
fiber-based IPtel
service, but I believe they require VoIP TA so I guess the end result is
just a standard
analog line.
Jason
Cory Andrews wrote:
Jason - I believe the Sangoma T1/E1/J1 boards may work in Japan, I
I asked Softbank and it seems that using SIP etc directly is not an option.
Something to do with theVoIP-TA being used for communications between
the providers call-agent.
Jason
Steven Critchfield wrote:
On Mon, 2005-01-31 at 12:53 +0900, Jason Frisch wrote:
Sorry for my ignorance, but what
Eicon Diva Server BRI = ISDN I think..
JAson
Leo Ann Boon wrote:
Jason Frisch wrote:
Hi all,
I am trying to setup Asterisk here in Japan in
my office. However I am having a hard time
finding hardware that is supported. I tried
Voicetronix but they said that they are too busy
to create a driver
://curl.netmirror.org/libs.html
So you likely need:
http://mirrors.kernel.org/fedora/core/3/i386/os/Fedora/RPMS/libidn-0.5.6-1.i386.rpm
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
403.244.8089
www.coalescentsystems.ca
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Hello,
I just checked out the latest CVS and compiled and now
get the following error:
[res_config_mysql.so] = (MySQL RealTime
Configuration Driver)
Jan 26 13:03:51 WARNING[27081]: config_old.c:27
ast_load: ast_load is deprecated, use ast_config_load
instead!
== Parsing
back through the NAT to the phone connected to the laptop.
That's what I'm seeing. My SIP phone cannot register to my asterisk
box through siproxd. I'm not sure if it's the phone or siproxd but
it's not asterisk -- asterisk doesn't care.
Thanks.
On Mon, 2005-01-24 at 13:54 -0500, Jason Lixfeld
On Jan 25, 2005, at 2:02 AM, Adam Goryachev wrote:
On Mon, 2005-01-24 at 23:51 -0700, Kim Lux wrote:
I'm trying to get similar working with a Grandstream. I'm getting a
lot
of echo. My laptop is crashing when the call terminates.
What are you using for the NAT setup on your laptop ?
-users
or Help forum:
http://sourceforge.net/forum/?group_id=121515
SUSE does some things differently - the main difference is the apache2
(httpd) configuration.
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc.
403.244.8089
www.coalescentsystems.ca
the computer. I
never get to the runonce. I've let it sit for 8 or 10 hours and nothing
happens. I'm using a 400PII, has the intel 440BX chipset, 256 meg of ram.
Is this simply a hardware issue?
Thanks for any help!
Jason
This message along with any attachments is intended only for the use
Ok, I have a 7960 that's plugged into my laptop. my home network is
wireless so I don't have a switch anywhere to plug the phone into
directly. I'm running siproxd on my OS X laptop and I can make
outbound calls from the 7960 fine (I guess I don't have the phone
configured to register
+Installation
[EMAIL PROTECTED] 0.3 uses CentOS 3.3 with a recent Asterisk:
Asterisk CVS-v1-0-01/22/05-02:50:58 built by [EMAIL PROTECTED] on a
i686 running Linux
It also bundles AMP ;-)
Project page:
http://asteriskathome.sourceforge.net/
Regards,
--
Jason Becker
Director CEO
Coalescent Systems Inc
I've got a simple MeetMe conference configured using Asterisk 1.0.3 on
Gentoo. I'm using zaprtc for timing from the bri-stuff package.
extensions.conf
exten = 37455,1,NoOp(Drill Squad Conference)
exten = 37455,2,Monitor(wav,drillsquad-37455,mb)
exten = 37455,3,MeetMe(37455,pMs)
Now, when I
on the installation.
After that, the system is theirs.
Always test emergency services access for premises equipment based solutions
unless you have signed documentation from the client that they do not want
911 access out of their system!
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
Luck!
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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I've recently switched my * server from FreeBSD to Gentoo using the
same configs from FreeBSD on my Linux machine, except the new Linux
machine is running 1.0.3 where the old machine was running 1.0.2.
Whenever I try to dial into one of my DIDs, I get this in the debugs
and the call gets
Anyone know what these messages mean? I see then scrolling about one
every 10 seconds while running asterisk -vcdg
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register_verify: No registration for peer '1'
(from 27.21.26.2)
I then issue this Dial cmd:
IAX2/1SIP/${JASON}SIP/${OFFICE}SIP/${LAPTOP}|20|tT
But get this when I attempt to receive a call:
Jan 18 21:04:44 WARNING[22491]: chan_iax2.c:2320
create_addr: No such host: 1
Jan 18 21:04:44 NOTICE
On Mon, 10 Jan 2005 19:38:23 +, John Middleton
[EMAIL PROTECTED] wrote:
Not an enterprise level system, but anyone used the www.intertex.se IX66?
Yes they work great no nat traversal issues,
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killall -9 mpg123 , this is a known issue after reloads ..
On Wed, 12 Jan 2005 13:25:50 +0100 (CET), Remco Barende
[EMAIL PROTECTED] wrote:
Hi list!
I keep getting abandoned / stale mpg123 processes. I cannot even kill them
off using killall I really have to do kill -9 to get them away.
III_dequantize_sample: mpg123: Can't rewind stream by
10 bits!
Segmentation fault
I am testing with another mp3 file now, but I would
never expect a 'Segmentation fault'. Has anyone seen
this? If not, should I report as a bug and how best
to do this?
Thank you,
Jason
if it is gateway related or Axxess related but I
ended up putting a PRI into the Axxess and connecting to * via that PRI,
then doing all of my IP stuff (via SIP) in *.
I have access to an axxess for testing so I will play with it a bit and see
if I can figure out a better way.
Jason Kawakami
Hello,
Ever since I started using Asterisk I always get this
error:
Jan 10 15:39:26 NOTICE[4501]: res_musiconhold.c:463
monmp3thread: Request to schedule in the past?!?!
I have a dedicated system system that really runs only
Asterisk:
- Pentium III 500Mhz
- 128MB of RAM
- 10GB of Disk Space
-
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