Read The Manual Before Asking!!
Indeed:
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How can I have asterisk ignore incoming rings so it doesn't answer a
specific line. I tried setting up an empty context section but that didn't
work.
Make a long delay the first line of the phone's context. This can even
be turned on and off using a few more lines.
On Wed, 27 Apr 2005, Adam Goryachev wrote:
Just wondering, but does the AMD multi CPU architecture improve the
interrupt handling? My understanding of that architecture is that each
CPU can deal with it's own PCI bus/interrupts/etc independently of
each other, and also with their own
Hello,
I want to call a peer over the Asterisk Manager with this php-script:
htmlbodyPRE?$socket = fsockopen("192.168.204.44","5038", $errno, $errstr, $timeout);fputs($socket, "Action: Login\r\n");fputs($socket, "UserName: test\r\n");fputs($socket, "Secret: test\r\n\r\n");//fputs($socket,
Hi List,
I have seen this:
http://www.convergence.com.pk/iax2/trunked.html
According to this table, using trunking, you can have 16 channels with
171.7 kbps bandwith using g.729 + IAX2 trunking? Sounds too good to be
true...
Any comments on this?
Cheers,
Jean-Michel.
--
Ykoz Un Max - La VoIP
On 4/26/05, Matthew Boehm [EMAIL PROTECTED] wrote:
Trying to make a call via our PRI: (CVS everything,
CVS-NHEAD-04/23/05-16:08:12)
-- Executing Dial(IAX2/[EMAIL PROTECTED],
Zap/R2/2815699900|30) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called R2/2815699900
Irakli Natsvlishvili wrote:
100k question - does asterisk correctly handle following situations:
There are plenty of good documents on Asterisk, SIP and NAT on the
voip-info.org wiki. Please look them up. There are also information
within the configs/sip.conf.sample file within Asterisk.
1.
On 4/26/05, snacktime [EMAIL PROTECTED] wrote:
On 4/26/05, Sean Kennedy [EMAIL PROTECTED] wrote:
Hi folks,
I'm curious; What does everyone do for failover? I have two servers,
same os/compilation. I designate one the master, the other the slave,
and I rsync the config files once an
Hello,
Question: I use x-lite as softphone and I want to call with my softphone a peer who is the result of my ldap search.
Has someone an idea how I can fixe this problem??
THX
Gesendet von Yahoo! Mail - Jetzt mit 250MB kostenlosem Speicher___
You can record queue conversations, check out the configs in queue.conf
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Daniel Salama
|Sent: Martes, 26 de Abril de 2005 05:53 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
Not a bad idea at all!
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|William Suffill
|Sent: Martes, 26 de Abril de 2005 07:02 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] YAC and IPs
|
|Why not
Go have a look at http://www.asteriskguru.com/tool2.php and calculate it
for yourself.
This is without the signalling frames for call setup / teardown.
(bandwidth used by those is very small).
Greetz,
/Z
Incoming Bandwidth
Outgoing Bandwidth
Calls: 16 Calls: 16
RTP: 2.34 Kbps
Im trying to write some tutorial for these ever recurring SIP + NAT
questions.
Its far from ready, and its without layout, but the draft can be found
at: http://www.asteriskguru.com/natut.php it has all most of the
situations explained, and explains all the options you need to look at
in the
One thing that could be done is to have a disk array for voicemail and
all with dual controllers. Then plug that into each of two servers.
Bind the IP components to a IP that is transportable between machines.
When one fails ifconfig the failover machine to use that IP (could be a
virtual
On Wed, 2005-04-27 at 00:52 -0700, trixter http://www.0xdecafbad.com
wrote:
One thing that could be done is to have a disk array for voicemail and
all with dual controllers. Then plug that into each of two servers.
Bind the IP components to a IP that is transportable between machines.
When
Could you explain me some more how i could use dual controllers ? Is
this done with special harddisks ? What hardware do i need to do this ?
/Z.
trixter http://www.0xdecafbad.com wrote:
One thing that could be done is to have a disk array for voicemail and
all with dual controllers. Then plug
On Wed, 2005-04-27 at 11:17 +0300, Zoa wrote:
Could you explain me some more how i could use dual controllers ? Is
this done with special harddisks ? What hardware do i need to do this ?
We used a winchester drive array, which is not cheap, and way overkill
for asterisk. EMC makes similar
MOH is working in that a defined extension works just fine:exten
= 6000,1,Answerexten =
6000,2,MusicOnHold()musiconhold.conf is as per the
default:[classes]default =
quietmp3:/var/lib/asterisk/mohmp3,-zand zapata.conf and sip.conf
havemusiconhold=default and musicclass=default
respectively.However
I'm new to asterisk and still learning it.
I wanted to ease my efforts a bit and use AMP (Asterisk Management
Portal), and see what changed in the config files when I use it.
However, I realized that I can only add SIP, IAX2 and ZAP extensions - I
didn't see an option to configure an ISDN
Andrew Elchuk [EMAIL PROTECTED] wrote:
I did some searching and haven't found a solution to my problem. But
right now we are performing a transition from an old system to a new
system using asterisk and only a few people are on the new system and
testing it out. Anyways, I was wondering
I have 2 Billion cards and I can't get the hfc driver to work. I get
this error:
ZT_CHANCONFIG failed on channel 9: No such device or address (6)
What am I doing wrong ?
ztcfg -vvv gives me this:
88--8-
Zaptel Configuration
S.NASROLLAHI
hi
i am a new member
i want to learn what is TOS and LOG command in the access list and
what are they doing?
what is their advantage ?
when i should use them?
thank u
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Hello;
Does anybody know how can I get the Cisco 7.4 SIP firmware?
Many Thanks
Betul
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Bam wrote:
MOH is working in that a defined extension works just fine:
exten = 6000,1,Answer
exten = 6000,2,MusicOnHold()
I had some similar problems with asterisk v1.0.6, 1.0.7 solved this. (it
had something to do with SIP and MOH)
Cheers.
___
On Wed, Apr 27, 2005 at 11:08:46AM +0200, Thomas Andrews wrote:
I have 2 Billion cards and I can't get the hfc driver to work. I get
this error:
ZT_CHANCONFIG failed on channel 9: No such device or address (6)
What am I doing wrong ?
This is my /etc/asterisk/zaptel.conf:
[channels]
Hi all.
I am new in the list and i believe i have read enough to run an asterisk
pbx good, but i have a problem.
My instalation is enterely SIP based and i am trying now with
grandstream budge tone 102 because with x-lite softphone i cannot get
transfer, supervised or not, be fine.
Few
Hi,
The legal way is to buy a smartnet (support contract) for the soft.
That way you can download it from Cisco's
web site.
Try to contact your reseller.
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
Mark Johnson wrote:
If you don't have any facts to share, please don't bother. I am
desperate and don't have alot of time left and am begging for the list's
advice. I left probably the largest post this month with EXACTLY what I
have tried, the results, debug information, etc... I have
/path/src/asterisk/doc/README.variables isn't what you are looking for?
Jason Walker wrote:
I second this. Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama
Sent: Tuesday, April 26, 2005 3:20 PM
To: Asterisk Users Mailing List -
I'm trying to learn Asterisk.
So far I'm using kphone and an ISDN line (with Eicon DIVA 2.01 PCI card).
I have created that extension following The Asterisk Handbook (page 36):
[ext-local-custom]
exten = _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten = _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten =
I can't vouch for the image quality personally, but I have yet to hear of
any complaints regarding quality from the end users.
Craig
- Original Message -
From: Matthew Boehm [EMAIL PROTECTED]
To: Asterisk Users asterisk-users@lists.digium.com
Sent: Wednesday, April 27, 2005 11:16 AM
Title: Message
I had a look at it and...yes it seems to be the same
card and it costs much less then I payed for it :(
- j -
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
RazzaSent: 26 April 2005 17:16To: 'Asterisk Users
Mailing List - Non-Commercial Discussion'Subject:
try putting
exten = _0.,4,Hangup
like
[ext-local-custom]
exten = _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten = _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten = _0.,3,Congestion
exten = _0.,4,Hangup
regards,
Umair bari
Tomasz Chmielewski wrote:
I'm trying to learn Asterisk.
So far I'm using kphone and an
NBX does it due to the proprietary protocol.
- Original Message -
From: Jeremy Koski [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 26, 2005 11:56 PM
Subject: Re: [Asterisk-Users] Cisco 7960 earpiece
Hi,
I try to generate a dial tone (tone you hear when you pick up the hook).
The tone should be stopped as soon the user dials a single digit.
Unfortunately Playtones(dial) don't stop until another extension is
completely dialed.
DISA doesn't work either with our Siemens Phones.
The scenario
is there anyway of changing the default filename of the monitor file if
using the record option in agents.conf. The ChangeMonitor command seems
to work only for a channel if it's using the Monitor command.
Julian.
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Umair Bari wrote:
try putting
exten = _0.,4,Hangup
like
[ext-local-custom]
exten = _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten = _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten = _0.,3,Congestion
exten = _0.,4,Hangup
no, still does not hang up :(
I have to pick up the phone and hang up manually (or kill
In response to a previous question about disabling music on hold, I was
advised to do:
noload = res_musiconhold.so
Unfortunately, this keeps Asterisk (1.0.5) from running:
[chan_iax2.so]Apr 27 04:11:34 WARNING[19654]: loader.c:258
ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so:
On Wednesday 27 April 2005 12:12, Henry Jensen wrote:
Hi,
User wants to call the number 12345
1. User picks up the hook
2. User dials 0 - hears dial tone
3. User dials 1 - dial tone stops
4. User dials 2345 - phone 12345 is ringing
We're using chan_capi and had this same problem... The
Yes, I read them. But, then my question is: how can I make the file
name include the agent that will get the call once it's distributed?
Thanks,
Daniel
On Apr 27, 2005, at 3:40 AM, Anton Krall wrote:
You can record queue conversations, check out the configs in queue.conf
|-Original
Ed Greenberg wrote:
In response to a previous question about disabling music on hold, I was
advised to do:
noload = res_musiconhold.so
Unfortunately, this keeps Asterisk (1.0.5) from running:
[chan_iax2.so]Apr 27 04:11:34 WARNING[19654]: loader.c:258
ast_load_resource:
On April 27, 2005 12:42 am, Matt Klein wrote:
Most likely, they can give you Echo Can for free.
Bell Canada will not put echo cans on their PRIs unless you specifically ask
(and pay) for the service.
Indeed, the line techs were surprised to know that echo could even exist on
PRI; thse are
On April 27, 2005 12:03 am, Mark Johnson wrote:
day now. What I find strange is this... If I speak at a normal tone,
it sounds OK. I still get static noise when the other person speaks.
If I talk louder, I start to get what sounds like a partial echo. If I
yell, I get a definite echo.
It
Hi all. I ma having a problem with pstn/tdm lines.
After the system has been in use for a while, it seems that I can only
use one line in the system. I have three PSTN attached to Digium
TDM04B/400P and they are grouped into g0. I tried using callprogress and
busydetect in the zapata.conf but
On 4/26/05, Jeremy Koski [EMAIL PROTECTED] wrote:
Normally, when you speak into the receiver of a phone, you can hear
yourself in the earpiece at a very low volume. I have a Cisco 7960 phone
that I'm using with asterisk and I don't get that echo back on the
earpiece speaker. I only have
Does anyone know what the [WARNING: . Changethread: Can't change device
'**Unknown**'] line means below..
I just set verbosity to level 5, and noticed that error everytime a
voicemail is left.. Everything seems to work ok, and I have no idea how long
that error has been there, but I'm just
Title: Busy Tone
Hello,
I've connected an Panasonic KX-TD 1232 PBX to an Asterisk PBX through an ISDN-line. I use an AVM Fritz! ISDN PCI card on the Asterisk PBX and connect it to the S0 bus of the Panasonic. When I make a call from a softphone to a phone that is connected to the Panasonic,
Works, thanks.
--On Wednesday, April 27, 2005 6:36 AM -0500 Eric Wieling aka ManxPower
[EMAIL PROTECTED] wrote:
Ed Greenberg wrote:
In response to a previous question about disabling music on hold, I was
advised to do:
noload = res_musiconhold.so
Unfortunately, this keeps Asterisk (1.0.5) from
On 4/27/05, Andre Normandin [EMAIL PROTECTED] wrote:
Does anyone know what the [WARNING: . Changethread: Can't change device
'**Unknown**'] line means below..
I just set verbosity to level 5, and noticed that error everytime a
voicemail is left.. Everything seems to work ok, and I have
I'm attempting the following set up.
During Hours - Receptionist Takes the call (no problem works great)
After hours I would like to add a item to the receptionist to transfer the
call to my cell, any direction would be a great help.
I have 4 PSTN incoming lines as backup and Voicepulse.
On April 26, 2005 10:57 am, Eric Wieling aka ManxPower wrote:
We have terrible problems sending faxes via the TDM cards. Not even
using SpanDSP. Just TE110P for the telco side and TDM400P for the fax
machine.
Yes there is a timing issue that crept in somewhere in the last 12-15
Hi Cyril,
Good work. process_baud is a fairly big routine, and your backtrace
doesn't give the actual line number at which things fall over. However,
studying the code I see that I do not protect against the possibility of
a divide by zero during the initial coarse carrier estimation of any of
We've just had problems with a range of Intel boards (Bukner and
Avalon) that have the PCI-Express technology -
http://www.digium.com/index.php?menu=compatibility lists the Intel
SE7525GP2 but we've had problems with the Intel SE7221BK1-E.
Digium say Firmware release 10 fixes this issue. All new
Message: 5
Date: Wed, 27 Apr 2005 12:04:30 +0100
From: Johan Akerstrom [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Good FXO for UK use.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type:
Hi!
I looking for an example how to use the dialparties.agi from Asterisk
Management Portal 1.10.007a.
I tried to understand it by reading the extensions.conf of AMP, but
without success.
Is anybody out there, who can give me a more easy example or an explanation.
Thanks,
Christian
Just as a reminder for those using Outlook, a large percentage of us
that receive html postings to the list simply delete them. If you
want to see responses from a larger group, stop the html stuff.
From: Wiley Siler [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Shouldn't it be: ?
bchannel = 9,10
dchannel = 11
bchannel = 12-13
dchannel = 14
Julian J. M.
On 4/27/05, Thomas Andrews [EMAIL PROTECTED] wrote:
On Wed, Apr 27, 2005 at 11:08:46AM +0200, Thomas Andrews wrote:
I have 2 Billion cards and I can't get the hfc driver to work. I get
this
On April 27, 2005 09:04 am, Rich Adamson wrote:
I would sort of disagree with the spiking thingie (now). If you modify
the zttest app to provide timing output in terms of seconds and
microseconds, you don't see the spiking impacting those measurements.
Rather, you see 8,192 bytes arriving in
I would really appreciate any insight here. I have seen a number of
posts in the past regarding implementation of a voicemail detection
scheme using silence detection as well as the machine detect, but
without MACRO_RESULT, there doesn't appear to be any way to actually
implement this.
Thanks
Rich Adamson wrote:
Just as a reminder for those using Outlook, a large percentage of us
that receive html postings to the list simply delete them. If you
want to see responses from a larger group, stop the html stuff.
Mozilla Mail, at least, lets you do View / Message Body As / Plain Text.
It
hi folks,
I looked through the list archives and the wiki, but couldn't find an
answer to this. Apologies if I just missed something obvious.
I want to only have call waiting for certain calls (i.e., those that are
dialed directly to a user rather than going through a queue). It seems
that
I've heard a few times that the firmware for Cisco
Phones to use them with SIP is going to increase $150. Is this
true?
-
Dan Levine
CYTEXONE | Your Technology
Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED]
http://www.cytexone.com
Good points. I stand corrected.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Wade
Sent: Tuesday, April 26, 2005 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Extensions / Contexts
Wiley
Yep. I have this working now.
Thank you!
Wiley
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
HalesSent: Tuesday, April 26, 2005 4:31 PMTo: 'Asterisk
Users Mailing List - Non-Commercial Discussion'Subject: RE:
[Asterisk-Users] Polycom IP4000 Conference Phone
The
Causes me to wonder a couple
of things.
Why does ANYONE use Outhouse or Outhouse Express? There are many much
more friendly Windows E-mail clients, from Mozilla on down
Don't know nor care about Linux E-mail. For me Linux is a means to an
end , not a religion.
Even more of a question, why
Not that I know of I am a Cisco partner and the
Category 1 contract is still at least half that or less.
- Original Message -
From:
Dan Levine
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, April 27, 2005 8:49
AM
Subject:
Thanks a lot, I thought this is possible because I don't need to link
companies, also, I can solve the problem of the IVR depending on the
channel of the PSTN that originates the call.
thanks again
Sebas
Wiley Siler wrote:
The short answer is No.
The method you describe is intrinsicly
Hello,
I'm having some problems upgrading my system from 1.0.1 to 1.0.7. After
the upgrade the Playback and Background dial commands don't produce any
audio or extremely distorted. I've tried custom recordings and the
prepackaged ones with the same result. Calls work fine using ulaw
through
Sounds like a possible disconnect issue, When the lines are not available
try doing a 'zap show channel X' with X being the channel number 1,2,or 3
and see if asterisk thinks the line is onhook or offhook. Just a thought.
Henry
- Original Message -
From: Colin E. McDonald [EMAIL
Just as a reminder for those using Outlook, a large percentage of us
that receive html postings to the list simply delete them. If you
want to see responses from a larger group, stop the html stuff.
Mozilla Mail, at least, lets you do View / Message Body As / Plain Text.
It doesn't
I've been trying to upgrade a grandstream BT-101 phone, whatever I do,
it doesn't seem to want to upgrade. I've pointed it at the grandstream
TFTP IP listed on the wiki, I've also tried pointing it at my own TFTP
server (after putting all the firmware images downloaded from
grandstream website
Look at the agents.conf. There is an option there to record calls. Maybe
this will point you in the right direction.
- Original Message -
From: Daniel Salama [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday,
Hi,
how can I determine the status (busy, offline, ringing, duration of
current call) of an SIP phone?
Elmar
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Causes me to wonder a couple of things.
Why does ANYONE use Outhouse or Outhouse Express? There are many much more
friendly Windows E-mail clients, from Mozilla on down
Tight integration with Exchange 2003.
Find me an alternative client that is as stable and that has such tight
Perfect, that's exactly what I need.
I will try that, thanks a lot.
Sebas
Matt Riddell wrote:
Sebastian Silva wrote:
Hi everybody,
I am writing here because I can't find the solution to my problem (my
asterisk configuration). I hope somebody can give me a hand with it:
I need to provide a PBX
Christian
As I understand it
After a user dials an extension number, Asterisk calls dialparties.agi
dialparties.agi checks the asterisk database (show database [from
cli]) for data matching items like Call Wating (CW) Call Forward (CF)
etc.
If one is present (in a defiend order) then rather
I've heard a few times that the firmware for Cisco Phones to use them with
SIP is going to
increase $150. Is this true?
If you follow the cisco license agreements, yes.
The cisco list price of the 7960 (as an example) includes their
non-sip software for something like $650. (Street
Use this - http://gs-firmware.gratissip.dk/
Works fine, instructions are clear here.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Adam Goryachev
Sent: Wednesday, April 27, 2005 10:19 AM
To: Asterisk Users Mailing List -
I've been scratching my head trying to think of a way to do this, but
without success yet.
I'm using the Manager API. If I have two channels linked to each other
(i.e. direct connection), or even if they are independent channels,
I can transfer them both to the same extension by using Action:
AMP does exactly this, why not look at their dial plan for instructions.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Daniel Dziubanski
Sent: Wednesday, April 27, 2005 8:27 AM
To: 'Asterisk Users Mailing List -
Hey guys,
I'm fairly new to Asterisk. Our objective is to have a VoIP PBX
connected to our PSTN lines. So, right now, I have a box running OpenNA
Linux, with a 2.4.29 kernel. Asterisk 1.07 and the latest Zaptel drivers
also.
I have 2 Gnet SIP phones connected on the same switch as the
Elmar Haneke wrote:
Hi,
how can I determine the status (busy, offline, ringing, duration of
current call) of an SIP phone?
Remember that the SIP phone is a kingdom of it's own. Right now,
Asterisk does not really now anything about what is happening out there
in the SIP woods. We know about our
Cyril VELTER wrote:
I just installed it and will keep you informed if a new crash occur, but even
with pre15, crash where not very frequent and usually come in series (~ one
serie of 3/4 crashes every two weeks, so we might have to wait some time...).
I'm pretty happy with the receiving side
Tony Mountifield wrote:
Andrew Elchuk [EMAIL PROTECTED] wrote:
I did some searching and haven't found a solution to my problem. But
right now we are performing a transition from an old system to a new
system using asterisk and only a few people are on the new system and
testing it out.
Hi Steve,
Good work. process_baud is a fairly big routine, and your backtrace
doesn't give the actual line number at which things fall over. However,
studying the code I see that I do not protect against the possibility of
a divide by zero during the initial coarse carrier
Cross posting on purpose to transition the thread to -dev
The issue in this thread is the frame transfer rate for the TDM analog
card almost always exceeds the 1.000 seconds expected by the design.
The frame transfer rate seldem impacts voice (the missed frames aren't
noticed), but seriously
As ny 10 year old step-daugher says I don't get it..
Can't you just do a redirect if you specify the channels, * doesn't care
if they are bridged together or not. You may end up with zombie
channels if the other leg does not drop, but you could do a soft hangup
and take care of that..
Or am
On Wednesday 27 April 2005 10:40 am, Tony Mountifield wrote:
I've been scratching my head trying to think of a way to do this, but
without success yet.
I'm using the Manager API. If I have two channels linked to each other
(i.e. direct connection), or even if they are independent channels,
I
Not that I know of I am a Cisco partner and the Category 1 contract
is still at least half that or less.
He was talking about the SIP-license... Not the SmartNET. If you have a
SmartNET, you CAN download the SIP load but to use it, you need the license.
I think that's the point; to use
ChanIsAvail
Show application Chanisaval
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Elmar
Haneke
Sent: Wednesday, April 27, 2005 10:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Determinating Phone status
Hi,
how can I
On Tue, 26 Apr 2005, Dana Olson wrote:
You mean like the problem I described earlier on this list?
http://lists.digium.com/pipermail/asterisk-users/2005-April/103153.html
I am not sure why I didn't think of disabling call waiting, but that
seemed to work with a Grandstream BudgeTone
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Wednesday, April 27, 2005 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Determinating SIP Phone status
Elmar Haneke wrote:
Hi,
Andrew Elchuk wrote:
Tony Mountifield wrote:
Andrew Elchuk [EMAIL PROTECTED] wrote:
I did some searching and haven't found a solution to my problem.
But right now we are performing a transition from an old system to a
new system using asterisk and only a few people are on the new
system and
Maybe this would best be explained in a diagram:
1). person A --- music on hold and person B --- music on hold
2). *some manager API action*
3). person A --- person B
This is what I think he's asking about, how do you take two parties on
different conversations and put them
Also try the snom soft phone: http://www.snom.com/snom360softphone.html. Sorry,
Windows only:-(
But at least its free!
Enjoy, CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Guillermo Salas M
Sent: Wednesday, April 27, 2005 12:01 AM
To:
From: [EMAIL PROTECTED] on behalf of Sean Kennedy
[EMAIL PROTECTED]
Posted At: 26 April 2005 21:25
Conversation: [Asterisk-Users] Remote Phones - No Audio In Either
Direction
Posted To: Asterisk-Users
Subject: Re: [Asterisk-Users] Remote Phones - No Audio In Either
Direction
Paul Tyreman wrote:
Hi,
I successfully installed zaptel,libpri,asterisk and qozap in a Suse 9.2.
I removed the old modules loaded as default by Suse.
Now I'm triying to load qozap.ko but I receive this error:
insmod: error inserting 'qozap.ko': -1 Unknown symbol in module
and in dmesg:
qozap: unsupported module,
I see that G723 and G729 require a license to be used, or can be used
(in the case of G723) in pass-through mode only.
My question is.. if my voip terminator supports G723 and G729 only, do
I still need a license? Or is that considered pass-through? If so,
do I need to do anything special to
In article [EMAIL PROTECTED],
mattf [EMAIL PROTECTED] wrote:
Maybe this would best be explained in a diagram:
1). person A --- music on hold and person B --- music on hold
2). *some manager API action*
3). person A --- person B
This is what I think he's asking about,
Guy Boehm wrote:
fputs($socket, Channel: 6159bfb47b9\r\n\r\n);
Response: Error
Message: Invalid channel
the Channel: var needs to be in the form of type/dev/numbertocall
like Channel: IAX2/user:[EMAIL PROTECTED]/14085551212
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