On Mon, Feb 28, 2005 at 08:58:48PM +, Tony Mountifield wrote:
I've just set up a new box with FC1+updates and the latest Stable
Asterisk from CVS.
Asterisk is started with the default safe_asterisk script with a
console on TTY9.
The coloured text on this console is made up of weird
Hello,
I need to know the sdp block of the sip user agent calling.
Is there anyone who can help me?
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Salut Guy,
I have the same problem with a Cirpack (B3G carrier)
What I see is that you use sip info to detect DTMF.
The problem is that there is no normalisation on the content of the sip
info frame for dtmf detection.
First, asterisk try to detect the header application/dtmf-relay
and you have
I think that the chan_capi-0.3.5 only works with stable Asterisk, or else you
need a patch(http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2)
but I never got that to work.
Read more here http://www.voip-info.org/wiki-Asterisk+Linux+Fedora
Janne
-Ursprungligt
Hi all,
A new DIAX version 0.9.10f is available for download at one of the followind
locations:
http://www.laser.com/dante
http://www.cosmica.ro/dante
http://www.geocities.com/tdanro
What's new in 0.9.10f comparing with the old 0.9.10a:
- volume and mic levels saved in the config file;
- master
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On Mon, 28 Feb 2005, Anders F Eriksson wrote:
Today I received a TDM11B (1 FXO and 1 FXS) and got it installed just fine.
I bought the card mainly to get caller ID to work properly in Sweden, and
that works just fine.
However, if the called or
Hi Lyle, I live in Turkey.
Btw, hoping that busydetect or callprogress could detect hangups, I've
changed the PSTN tones according to ITU tones-0203.pdf (both in
indications.conf and zonedata.c re-made). But nothing changed, when I
debug the zap channel, I can see that * still detects the
Dear Users,
Thank you all for a great Forum.
In the past with my analog PBX I had 6 separate telephones configured auto
pickup and they where connected to the meeting room's microphone. When we
had a meeting, it was then possible for 6 members simultaneously to call in
with a normal landline
I looked over and over again and I still can't find it. Can you please
point it out to me? Maybe I'm just looking for the wrong term or something.
Thanks in advance
Rennes Neps
Eric Wieling wrote:
Rennes Neps wrote:
Hei!
Does anyone know how to configure this phone to autodial the number
after
Title: Message
Is there a
knownproblem with the latest CVS and the UK CLID patches or have new
commands been introduced to asterisk around musiconhold?
I have just updated
my machine following the instructions at h##p://www.lusyn.com/asterisk/patches.htmland
MusicOnHold stops workingwhen I
That issue is fixed in the CVS HEAD version of asterisk.
There are a couple of workarounds possible with 1.0.6. Check
the bugtracker for the bug where it was implemented for more
information. (sorry, don't remember the bug-number and don't
have time to look it up right now).
You might
Thanks to Dave Cotton, Adam Goryachev, Bob Goddard, Doug Lytle and Craig
Guy for their help on this, I think I am going to revert to Mandrake 9.2
and try the CAPI route again - my annual leave is running out!
Ray
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Hi,
When I use NoCDR application I obtain this warning in console log:
Mar 1 11:16:08 WARNING[3513]: cdr.c:114 ast_cdr_free: CDR on channel
'SIP/492-7371' not posted
Mar 1 11:16:08 WARNING[3513]: cdr.c:116 ast_cdr_free: CDR on channel
'SIP/492-7371' lacks end
Can someone explain to me what is
Oh btw, my version:
Asterisk CVS-HEAD-02/27/05-17:01:45 built by [EMAIL PROTECTED] on a i686 running
Linux
- Original Message -
From: Soner Tari [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 01, 2005
I just did a cvsup and noticed that app.c along with a few more files
has been changed since yesterday so I gave
it a shot and tried to compile again and now it worked (although along
with some warnings, but still..).
Nice!
/Patrik
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Hello asterisk users,
I am in a problem. I have an astersik setup and a broadvoice number. I
configured my asterisk and i can easily make call using that. But no
incoming call is in progress. The IVR said the line is busy. But actually
the phone(Broadvoice number is free). I am here using Xlite
I've just installed asterisk on a Debian Linux (apt-get it)
And i have placed two sip phones in sip.conf and i'm testing parking
with them
I have phone1-SIP/1000 and phone2-SIP/1007
The following happens if i park from calling party and everything is OK
1. Pickup Phone2 and call to Phone1
2.
Hello,
the Call Forwarding Always was not working in that firmware version (on busy
and on timeout should still work). It will be fixed in the next release.
Regards
Nils Ohlmeier
On Tuesday 01 March 2005 06:10, Rod Bacon wrote:
I am just playing with a SNOM 190. Overall, I'm very impressed
Kristian Kielhofner schrieb:
Bastian Schern wrote:
If you look back in the archives, you will see that many, many, many
people have gotten tripped up on the make linux26 issue. Sorry to
offend you. Remember that your original post never mentioned key
details that would help. Speaking of
Just a couple of guesses:
Have you configured the switch to supply a VLAN trunk to the phone?
Yes
Since the phone lets you configure actual tagging, that's what it
needs; if you've just enabled VLANs on the switch, and placed the port
the phone is on in a specific VLAN, the phone should not have
I'm trying to place a call from my Cisco 7960 and I'm receiving this error:
Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to
create/find channel
Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to
create/find channel
I can't place calls, but I can
Patrik,
you are more likely to get an answer on the asterisk-bsd list for these type
of questions. Because of the size of the Linux user base new features tend
to be added by Linux users which on occasion break the tree for us non Linux
users. These problems are normally picked up in a few
I'm using Asterisk, libpri and zaptel v1.0.6 on a SuSE Linux 9.1 with
Kernel 2.6.9. I had not installed this Kernel, it is the default kernel
on this dedicated server.
As per README.SUSE in /usr/src/linux, try doing the following in
/usr/src:
make cloneconfig
make modules_prepare
Then
Hey guys.
Im trying to setup Music on Hold. If I transfer a call (with dial) I like to
put the call on Music on hold..
Here's what I've tried so far:
On my I extensions.conf
exten =1,1,WaitMusicOnHold(30)
exten =1,2,Dial(SIP/mateo,18)
exten =1,3,VoiceMail(1001)
I have also added this line to
Would anyone know why Voicemail in * doesn't get the DTML keypresses
from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do
with dtmf_avt_payload: 101 setting in SIPDefault.cnf in the tftp server?
Thanks for any help!
Derek
--
Derek Conniffe
Rivertower Ltd
DDI: (Local Ireland)
Hello,
I didn't find any reference to this point. On my fresh install, I can't
record busy message to my voicemail. When connecting to personal voice
mail, I have no option in the advanced option menu : in fact just press
star to return to the main menu)...
Is there any configuration parameter
Just a couple of guesses:
Have you configured the switch to supply a VLAN trunk to the phone?
Yes
Since the phone lets you configure actual tagging, that's what it
needs; if you've just enabled VLANs on the switch, and placed the port
the phone is on in a specific VLAN, the phone should not have
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On Tue, 1 Mar 2005, Anders F Eriksson wrote:
That issue is fixed in the CVS HEAD version of asterisk.
There are a couple of workarounds possible with 1.0.6. Check
the bugtracker for the bug where it was implemented for more
information.
Tim De Lange schrieb:
I'm using Asterisk, libpri and zaptel v1.0.6 on a SuSE Linux 9.1 with
Kernel 2.6.9. I had not installed this Kernel, it is the default kernel
on this dedicated server.
As per README.SUSE in /usr/src/linux, try doing the following in
/usr/src:
make cloneconfig
make
Dear All
Im facing wearied problem with Sipura 3000 and asterisk .
Im trying to configure Asterisk with Sipura 3000 . I have configured asterisk with FSX port which is working fine.
I want to configure Asterisk FXO port for my outgoing and incoming calls.
Once Sipura received call from
Hi,
I am just new to Asterisk, and would like to experiment a little before
implementing and investing...
We have a Siemens SX353 DECT Base Station/desk telephone with USB
connection for PC, I know there are (beta) I4L drivers for this device
available.
Does anybody know whether this device
That issue is fixed in the CVS HEAD version of asterisk.
There are a couple of workarounds possible with 1.0.6. Check the
bugtracker for the bug where it was implemented for more
information. (sorry, don't remember the bug-number and don't have
time to look it up right now).
Hi,
methinks that in the good 3 months since i ordered an IAXy, things have
changed so much that now almost anybody out there with a VoIP hardweare
website offers complete phones for less money than the IAXy, with support
for both IAX2 and SIP in many cases, and fully configurable via its own
Hi all,
If anyone has any experience with a Cisco 7912 and chan_sccp
please email me directly. I am oh so close to making it work and have a few
questions.
Thanks,
-E
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Le mardi 1 Mars 2005 12:33, Hermann Wecke a écrit :
I'm trying to place a call from my Cisco 7960 and I'm receiving this error:
Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to
create/find channel
Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to
On Tue, 2005-03-01 at 10:53 +0100, Satchid wrote:
Dear Users,
Thank you all for a great Forum.
In the past with my analog PBX I had 6 separate telephones configured auto
pickup and they where connected to the meeting room's microphone. When we
had a meeting, it was then possible for 6
Hi,
What are your opinions on the iaxy? I have one coming.
From what I have seen, at least in the uk, iax2 hardphones are NOT
widespread; iaxtalk.com are the only store I can find which sell them?
C
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daiku
On Tue, 2005-03-01 at 07:52 +0100, Edgar de Leon wrote:
Hello, yesterday when i wasnt in the office the asterisk server stop
working, it was registering the sip terminals but cant make calls, because
im not in the office i told the people to reboot the server to make the
server works again but
Hey!
I have some problems finding the right versions of openh323 and the
pwlib. could anyone please help me finding them, does anyone have
openh323 1.12.* and pwlib 1.6.6?
Thanks alot!
Best regards,
Jan Eirik Sandnes
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I have some simple questions and i need your help guys.
I have ser server which working fine, between users.
I am trying to add some more features to the ser. Most important is the IVR.
I installed Asterisk and i am trying to register user in asterisk with no success.
Part of ser.cfg file where
Hello All,
Im new to the list and the whole voip server side. Im
trying to setup Asterisk to just do internal dialing, no access out to the pstn
is required/wanted at the moment.
Im running Fedora Core 3 with Cisco 7960s
phones (running SIP 6.3).
Ive set it up following these
What you are seeing is normal. You told the channel NoCDR so when the
channel gets destroyed, its informing you that it didn't save the CDR, as
you requested. Its kinda goofy way to tell you but...
-Matthew
- Original Message -
From: Scanna [EMAIL PROTECTED]
To:
Hi all,
I have a server with an Athlon 64 3200 and Fedora Core 2 x86_64.
I have compiled and installed Asterisk 1.0.6 without any problems. When
I try to make asterisk-addons-1.0.6 it say me:
[EMAIL PROTECTED] asterisk-addons-1.0.6]# make
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE
Hi, haven't found anything in google's, i wonder if there is a
comparative page of what to expect from running * on motherboards like
the EPIA and similar ones.
Since i have not used *ever* such kind of mini atx form factor boards,
I have no clue about their performance.
SIP-SIP communications,
Hi there, I just wondered if anyone has used multiple Fritz ISDN PCI v2.0
cards in their setup. This card is a single port BRI and is good value,
(apparently works well with *) putting 4 of these in a server is
significantly cheaper than buying a 4port BRI from Junghanns.
Has anybody had any
I downloaded the Janus-patch versions of pwlib and openh323 from the
download section of
http://www.inaccessnetworks.com/projects/asterisk-oh323
yesterday and compiled the oh323 module without trouble following the
ReadMe instructions.
Regards,
Cristian
On Tue, 1 Mar 2005 14:55:37 +0100,
On Tue, 2005-03-01 at 14:14 +, BCS Support wrote:
Hello people, I've been following the Astrerisk program for some years now
And then I'm going to ask some questions that suggest I have not been.
and I was wondering wether this is something that our company could supply
as a value added
On Tue, 2005-03-01 at 14:43 +, Brett, Gary wrote:
Hi there, I just wondered if anyone has used multiple Fritz ISDN PCI v2.0
cards in their setup. This card is a single port BRI and is good value,
(apparently works well with *) putting 4 of these in a server is
significantly cheaper than
I run mandrake 9.2, one FXO (x100p clone), 5 sip phones, MusicOnHold,
voicemail, etc. off my EPIA Classic/5000 with 512MB memory (I know 512
is totally OTT but had a spare SD stick lying around after upgrading my
main PC) and it works fine.
I would like to also run a Fritz ISDN card but am unable
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On Tue, 1 Mar 2005, Anders F Eriksson wrote:
I'm running CVS-HEAD-02/28/05-23:18:39 at the moment, and
it still happens.
I've seen the bugs in Mantis, but the answeronpolarity
doesn't seem to
make any difference ...
Could you post a
I am having a problem with Polycom auto-answer. I have the auto-answer
working between PhoneA and PhoneB, but when I try to use the intercom
between more then one phone I start having problems. PhoneA dials *3
which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only
one will pick
Derek Conniffe wrote:
Would anyone know why Voicemail in * doesn't get the DTML keypresses
from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to
do with dtmf_avt_payload: 101 setting in SIPDefault.cnf in the tftp
server?
Thanks for any help!
Derek
I have the same line in my
There are pointers to all the libraries from the documentation and on the
website. The most difficult part is making sure you have the make file set
up correctly with paths to the right directories for the libraries.
The docs do a good job of explaining how do get it complied and installed.
Steve Kann has developed a new jitterbuffer for IAX2, that hopefully
will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable
relase.
Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs
support in the form of funding in order to take the time to test this
Alex,
If you are forwarding calls in SER based on URI patterns rather than the
location database, you don't need to register Asterisk with SER. Instead
of the register line, you should have a peer for SER; something like this:
[ser]
type = friend
host = IP address or hostname of SER
context =
If you use the mISDN Fritz! driver with CAPI you should be able to use up to
4 Fritz! cards. I have it working with one card but have not tried four.
Craig
- Original Message -
From: Brett, Gary [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I set dtmfmode=inband for my 7960 in order for voicemail to work.
Craig
- Original Message -
From: Mark Johnson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 01, 2005 11:20 PM
Subject: Re:
John Cianfarani wrote:
Asterisk seems to start fine but the FOP op_server.pl doesnt seem to
want to start. Ive tried running it by hand as the asterisk user but
it doesnt spew any errors, and I cant find any log files that would
help me troubleshoot this issue.
Ive searched different archives
Thanks, this looks like what I need. Setting it up looks like a career
though but hey it's free so what can you do?
:)
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial
Olle E. Johansson wrote:
Steve Kann has developed a new jitterbuffer for IAX2, that hopefully
will be integrated into Asterisk v1.1 soon, to be part of the 1.2
stable relase.
Zoa and his bulgarian team is porting this buffer to SIP/RTP, but
needs support in the form of funding in order to take
We are in the process of ordering a Voice PRI to plug into Asterisk. Of
course we will be buying a card from Digium for this.
Question is this, there seem to be MANY options technically when ordering
this PRI (in the US) but since this is the first time ordering a voice
circuit I am clueless
Eric Rees wrote:
I am having a problem with Polycom auto-answer. I have the auto-answer
working between PhoneA and PhoneB, but when I try to use the intercom
between more then one phone I start having problems. PhoneA dials *3
which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but
If you are talking about the epia etc boards, they are mini ITX..
I am running an 800mhz one with 256mb ram as a test server, purely voip,
using a couple of SIP and IAX clients. No moh yet.
I had to modify the makefile in order for it to work, but once working its
fine so far.
C
-Original
In ser.cfg
--if (method == "INVITE") { if (uri =~ "sip:[EMAIL PROTECTED]"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xxx.xxx.xxx.xxx:5061"); t_relay(); break; } }
In sip.conf
Steve Underwood wrote:
re here: http://www.astertest.com/forum/viewtopic.php?t=13
Thank you for your contribution!
The hard work of building the thing was done for free, and now someone
brings out the begging bowl for the relatively minor activity or porting
into to another home. Frankly, that
Thanks I set the option for selinux to disabled in the
/etc/sysconfig/selinux config and that seems to have fixed the issue.
Thanks for your help
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker
Sent: Tuesday, March 01, 2005 11:04 AM
Steve-U,
This sip jitterbuffer stuff is still for free, no one has to contribute
anything, but any help financially, with code or testing is greatly
appreciated.
Everything is GPL and code is disclaimed to digium.
We spent the last 2 months researching + working on a sip jitter buffer,
first
Hi ,
I have setup SER and Asterisk on the samebox (behind a NAT) and try to
connect to SER from outside using Xlite.
But I got 405 error.
Here is my setup diagram.
NAT
SIP Client - Internet (PPPoE ---Router
That's kind of what I thought, but I am trying to put together a phone
to multi-phone paging system. I all ready have and overhead paging
systems, but the powers-at-be want a phone paging system.
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March
Title: Asterisk PBX Manager
Does anyone on this list have any experience Thirdlane.com's Asterisk PBX Manager?
And if so what do you think of it?
Regards,
Michael DiMartino
Director of MIS
The telx Group, Inc.
17 State St, 33rd Floor
New York, NY 10004
T: 212.480.3300 X2022
C:
Hi,
What are the SIP
phone models that proved to be working well with Asterisk? I appreciate your
recommendations.
Thanks
Walid
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The test is did for my presentation on astricon were partially done on a
via samuel 2
See: http://www.astertest.com/forum/viewtopic.php?t=10
Zoa.
C. Tomlinson wrote:
If you are talking about the epia etc boards, they are mini ITX..
I am running an 800mhz one with 256mb ram as a test server, purely
The problem is the Got event Ring/Answered(2) line.
Normally, a ring should not be detected and the DTMF-cid
times out and no incoming call is registered.
Make sure you load wctdm with the parameter
'opermode=SWEDEN', it might help.
You might also try to increase the 'RING_DEBOUNCE' define
The problem is the Got event Ring/Answered(2) line.
Normally, a ring should not be detected and the DTMF-cid
times out and no incoming call is registered.
Make sure you load wctdm with the parameter
'opermode=SWEDEN', it might help.
You might also try to increase the 'RING_DEBOUNCE' define
All,
I've put a new Integrics Tip. This one is on how to go about recording
voice prompts for your IVR. It's available at:
http://integrics.com/tips/recording_prompts/
--
Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699 479
http://integrics.com/
We have been seeing tons of additional SPAM coming through our Modus 4
server, mostly medical stuff.
Is anyone else seeing a big increase lately? I have not seen the list for a
bit seems I was unsubscribed somehow.
Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com
That's kind of what I thought, but I am trying to put together a phone
to multi-phone paging system. I all ready have and overhead paging
systems, but the powers-at-be want a phone paging system.
Fortunately for us Polycom people, somebody took the time to write a
Perl AGI script for just this
Andres,
Is your client based on libiax2? It is based on iaxclient?
-SteveK
[EMAIL PROTECTED] wrote:
Hi
We just developed an IAX web client. We are currently testing it,
and we hope to be ready to market around 15 of this month. It is based
on our own OCX, that has a lot of funtions to
Doh!
Wrong list, please ignore..
Sorry.. 30 lashes for me..
Todd
--
Start Your Own ISP!
http://www.YourOwnISP.com
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To UNSUBSCRIBE or
I have plead ignorance to Turkish standards, but will re-iterate that the
filter should not cause this.
Lyle
- Original Message -
From: Soner Tari [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 01, 2005
Take a look at
http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a
few small modifications it should work like a champ on the Polycom phones.
B. J.
-Original Message-
From: Eric Rees [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 01, 2005 10:38
To: Asterisk
Here's a better answer...
www.justfuckinggoogleit.com
Mohit.
On Tue, 01 Mar 2005 15:52:34 +0100, Dave Cotton
[EMAIL PROTECTED] wrote:
On Tue, 2005-03-01 at 14:14 +, BCS Support wrote:
Hello people, I've been following the Astrerisk program for some years now
And then I'm going to ask
Andres sounds as if this is Andres's own development. He mentioned IAX,
not IAX2. My guess is that he might have used one of the IAX GPL
Libraries and source trees, based on iaxClient and not libiax2. It is
possible that Andres is not aware of the GPL terms that he has to adhere
to, if he wants to
Hi.
It is propbably a really naive problem, but I really couldn't find
answer how to connect two Astrisks via SIP. I managed to do it via IAX
without any problem. But this is a test installation and I would like to
connect them via SIP.
So I have two computers:
pbx1 - 10.1.3.207
pbx2 - 10.1.3.204
Hi!
- accept URLs during a call and open that page in the default browser
when a call is answered;
Excellent!! :-)
Cheers, Philipp
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heya David ( Josephson ) :)
Only one brand of cheap GMRS radios that I've seen (Garmin) has the
duplex mode that allows use with repeaters and duplex base stations. I
think this is essential for successful integration with a phone system.
so i check 2-way radio on the garmin site
On 13:07, Tue 01 Mar 05, Damian Minkov wrote:
I've just installed asterisk on a Debian Linux (apt-get it)
Asterisk verion - Asterisk 1.0.5-BRIstuffed-0.2.0-RC7e
Hi,
Did you apt-get that * version ?
Where is the deb for the bristuffed version?
I can only find 1.0.5, both on packages.debian.org
Craig Guy wrote:
I set dtmfmode=inband for my 7960 in order for voicemail to work.
This will only work if you are using ulaw or alaw codecs.
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On PSTN-Line tab
Subscriber Information
User ID: 99
Password: 99
Dial Plans
Dial Plan 1: S0:99
PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: Yes
PSTN Ring Thru Line 1: Yes
PSTN Caller Default DP: 1
That should be it I think.
--
#Joseph
On Tue, 2005-03-01 at 04:34 -0800, dhananjay
I have installed asterisk - hurray!
I want to configure, i.e.,
800#---switch-asterisk-phone
need to create members and callers. Realize I need to configure the
dialplan in extensions.conf, however isn't there a checklist that
helps coordinate the many extensions of this
Hi,
Is anyone using mozPhone?
If so any feedback you can provide?
Thanks,
Glenn
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Asterisk should also tell the phone to turn off the light with a Notify
Message.
Flow should be:
Bootup:
Snom-* - SUBSCRIBE
*-Snom - 200 OK (Subscribe)
Call hits the hint:
*-Snom - NOTIFY(look at the XML in the Message)
Snom-* - 200 OK (Notify)
Stuff happens or call is over:
Hi All,
I've been playing about with the RECORD FILE agi function and am
finding two distinct problems with the resulting wav file when using a
non zero sample offset. Specifically, I call the function with a zero
offset and a given filename (the original recording), and then later
call it with
B. J. Bomar wrote:
Take a look at
http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a
few small modifications it should work like a champ on the Polycom phones.
B. J.
Polycom has a much better way to do auto-answer using SIP_INFO. My
sample configs have both a AutoAnswer
hi,
is there anyway to implement callback on busy and callback on no answer
on asterisk? has anybody done this before?
thanks,
Paradise Dove
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Asterisk-Users@lists.digium.com
I didn't test it on a live system. Just on their demo but it looks very good.
On Tue, 1 Mar 2005 11:46:40 -0500, Michael Di Martino [EMAIL PROTECTED] wrote:
Does anyone on this list have any experience Thirdlane.com's Asterisk PBX
Manager?
And if so what do you think of it?
Kristian Kielhofner wrote:
B. J. Bomar wrote:
Take a look at
http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a
few small modifications it should work like a champ on the Polycom
phones.
B. J.
Polycom has a much better way to do auto-answer using SIP_INFO. My
sample
Looking for a tech that can go onsite to Two Embarcadero
Center, Suite 670
San Francisco, Ca 94111 to work on a simple voip networking
issue. Anyone interested please call me at
9734333001 ext 226 asap.
Thanks
John Bittner
Simlab.net
___
I haven't seen any duplex handhelds that I can remember (ther must be
some).
The repeater would most likely be duplex. Many repeaters are built from
either two under-dash mobile radios or the old GE MasterII units
The question you really want to ask is if they can operate on different
Sipura 1000 or 2000?
--On Tuesday, March 01, 2005 10:15 PM +0900 Daiku [EMAIL PROTECTED]
wrote:
Hi,
methinks that in the good 3 months since i ordered an IAXy, things have
changed so much that now almost anybody out there with a VoIP hardweare
website offers complete phones for less money than
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