Hi
I have a Zyxel P2002 (ATA) with this config.
Registration works but i cant call inn. Outgoing works fine.
Any clue?
Thore
- Original Message -
From: Paul Dracevich [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent:
Hi,
You don't need a licence. Look at:
http://chan-sccp.sourceforge.net
I use this with a Cisco 79607914 and added some of my own patches, but this
driver is not stable.
It supports only g711 alaw and ulaw, but not g729 (the Cisco-phone does it!)
I tried to contact the developer to get and provide
How about setGlobalVar()
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Where can I find a good how-to to do this job. A small starting
how-to that let me understand the principles of setting a PBX with
asterisk. The handbook does not like starting guide.
Try this:
http://automated.it/guidetoasterisk.htm
___
I guess if you add the g729 license (or open codec if you are outside the
us and don't want to support patents) and add the ability to the driver it
should work.
Did you see the new version of chan_sccp? The standard Easter version
doesn't compile with * stable, the cvs version should.
On
You can't see the sweat, but ...
I would like tp post my improvements to ASTCC somewhere, ... but where???
bye
Ronald
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I checked it out. You have indeed created a very functional BBS setup,
using open source software. I like it a lot. But you will need to
attract a critical mass of Asterisk users in order to succeed in making
it an effective Asterisk and VOIP community resource. It takes people to
make a BBS
Hi List
As I have a Cisco PIX 515, with NO QoS functionality,
and Im looking for a router that does outgoing QoS to put in front of my
PIX. Problem is that Im using my 768/8096Kbit ADSL for both data and VoIP,
and as soon as data is being sent to the internet the sound quality drops to
Hi,
I have read in the wiki-pages, that I doesn't need the g729 license, if
I use it
only in path-thru-mode.
Of couse I added AST_FORMAT_G729A to chan_sccp capability, but it
dosn't worked.
That's why I tried to ask the developer.
The Easter version works fine with the * stable, if you add some
Version 0.73 - 3. April 2005.
* Italian Language added - Thank you to Francesco Romano for translating
* IPSwitchBoard can minimize to tray
Download: http://ipswitchboard.thorben.dk
IPSwitchBoard is now available in English, Danish and Italian; would you
like to help translate IPSwitchBoard?
Is it possible to run Asterisk with another GKs using Neighbor mode?
If it is possible, we can run asterisk with several gnugks.
On Apr 2, 2005 10:41 PM, Alex Vishnev [EMAIL PROTECTED] wrote:
I don't think you can. The rules of h323 is so that you can register with a
single gk at a time.
No, I'm not ignorant of how this works. You'll notice I put it
appears bad when I posted my results. Yes, it's not a perfect way to
show problems -- but taken with a grain of salt it's not half bad.
Especially when sampled over a longer period of time, and if the
original poster can correlate
If I were to buy 20 did's how do I know within asterisk which number was
dialed? (like say I want a few of the did's to ring specific extensions
if they are dialed and others to go through the menu)
Is there any ${var} that has the number dialed in on? (that would be
optimum).
It varies
Argh. I can't figure out what I'm doing wrong. I can dial with my SIP
phones just fine, but I want to set up an analog phone plugged into my FXS
port... and, while it gets dialtone, no matter what digit I press, I get
stuff like:
VERBOSE[21963]: -- Starting simple switch on 'Zap/1-1'
While on my network I can register ok with xlite but outside my firewall my
Xlite says that
regestraion has failed but I am still able to make calls
through it. I have opened ports: 5060 udp/tcp and 1-2 udp/tcp is
there another port
Xlite needs for proper regestration? Is is this
You need to upgrade these phones to the latest firmware for it to work
well with asterisk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thore
Sent: Sunday, April 03, 2005 3:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nathan Reeves wrote:
| Anyone running Cisco Call Manager and using Asterisk for voice mail
| services? Things working well or is the concept a bit of a hassle
| to implement?
|
Hi,
I'm using asterisk with a SIP trunk as a voicemail system for CCM
Hi Courtney -
If I were to buy 20 did's how do I know within asterisk which number
was
dialed? (like say I want a few of the did's to ring specific extensions
if they are dialed and others to go through the menu)
Is there any ${var} that has the number dialed in on? (that would be
optimum).
Your
Hi Paul -
I am very new in asterisk community. I just compiled installed
asterisk on a fedora core 3 machine and I want for test purpose to do
a small PBX that use X-lite windows sip clients and no trunk for the
begining.
Where can I find a good how-to to do this job. A small starting
how-to
With recent discussions in regards to a forum, I have set-up a
multi-faceted Asterisk and Open Source Discussion Board. The link is
www.voipnewbie.com/forum It is open and ready for use.
Hey Great! Thanks! Just make sure to get linked from the asterisk
website (probably in the Digium
Scott,
First, you need to get the most recent os
for the pix, otherwise you will have a lot of problems with udp packets and
translations (including bad checksum on your udp packets). I am running both
pix515 and pix501 without a problem with sip and h323. you dont need to
open any
I can say that I use FPDF.org for my OSRAIDS project. Take a look at
how I create PDFs on the fly.
http://OSRAIDS.org
On Mar 31, 2005 1:15 PM, Max W Blackmer Jr [EMAIL PROTECTED] wrote:
I am just beginning work on Trabas now. nothing as of yet. I just liked
the features that it currently
Charles,
I don't think asterisk is a full GK. So if you are asking if asterisk will
send out LRQ to the neighbors then I don't believe it would. As far as
registering with multiple gk, I wanted to correct myself. An endpoint/gw can
register with one primary gk and a number of backup gk. If the
On Sat, Apr 02, 2005 at 01:20:37PM -0500, Josh Alberts wrote:
I'm having trouble getting asterisk to run at startup using Ubuntu.
I've checked, and the asterisk dameon is set to run at init 5. However,
I'm not seeing anything that says that asterisk has been started during
the boot process.
Thank Matthew:
I do that, i create the database with tables for support RT Asterisk, then i
create the context deafult in the database, but the macro that i use is
steel in the etension.conf and its works.
Database Extension:
IDCONTEX EXTENPRIORITYAPP APPDATA
1
Chris Blake wrote:
Greetings *`s,
I have set up a call which constantly loops a pre-recorded message
waiting for the user to press a digit on their phone. At this point the
call is sent elsewhere in the dialplan.
But if the called party doesn`t press any buttons and hangs up, the
message carries
Hi
Im having Jasomi peerpoint far end SBC im trying to integrate this with asterisk .
When i call any no it directly goes to his voice mail.
But when i start debug on asterisk it received 403 Forbidden Proxy OutBound Policy from Peerpoint and call is not working .
isanybody using asterisk with
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Apr 2, 2005, at 8:00 PM, Nathan Alberti wrote:
I'm currently in the process of getting it to work for a CCME install,
I have it all working except for one thing.. I think it was calling a
phone from the asterisk server the call transfer back to
Hi
I haven't read all of the messages in this lengthy thread, so I hope
I'm not repeating something from it.
Just a couple of questions:
1. What about mail-archive.com for archiving the list?
2. The archive need not be related to the list. It just needs to be
subsribed to it. Anybody want to
Hi guys,
Is it possible to make Dial to call two extensions at the same time?
I want when the user pressed extension it call to two SIP phones at the same
time... Who wakeup first get the call...
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Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
Our community is also growing
Jozeph Brasil a écrit :
Hi guys,
Is it possible to make Dial to call two extensions at the same time?
I want when the user pressed extension it call to two SIP phones at the same
time... Who wakeup first get the call...
Dial(SIP/extensionIAX2/otherextensionOH323/...)
--
Daniel
I attempted to use the incominglimit and outgoinglimit in iax.conf and it
doesnt seem to work anylonger, running CVS-HEAD 3/16/05
So I tried using the SetGroup but, in the dialplan I am already using Get
and Check Group. I tried it with different variables and it still doesn't
workany ideas?
IAX is not an option as Sipura devices do not support AIX.Yes, the sipura will handle the different packet sizes...
Is it possible to reprogram asteris to do this?
On Apr 3, 2005 1:55 AM, Steven Critchfield [EMAIL PROTECTED] wrote:On Sat, 2005-04-02 at 21:16 -0500, Matt wrote: I'm aware that
Hi!
As I have a Cisco PIX 515, with NO QoS functionality, and Im looking for
a router that does outgoing QoS to put in front of my PIX. Problem is
that Im using my 768/8096Kbit ADSL for both data and VoIP, and as soon
as data is being sent to the internet the sound quality drops to
An option, but what about multiple inbound calls? I'd be worried that they
trip over each other.
But - given the odds of this happening (variable is set and then read
instantly) - it may be the route to go.
Thanks - Joe
-Original Message-
From: [EMAIL PROTECTED]
On Sunday 03 April 2005 06:33, [EMAIL PROTECTED] wrote:
Hi List
As I have a Cisco PIX 515, with NO QoS functionality, and I'm looking for a
router that does outgoing QoS to put in front of my PIX. Problem is that
I'm using my 768/8096Kbit ADSL for both data and VoIP, and as soon as data
is
NVC List Manager wrote:
As usual there's nothing that will beat OpenBSD. Takes 15 minutes to build
following the instructions on the CD cover.
To someone who has never installed OpenBSD (or FreeBSD + pf for that
matter) the learning curve is going to be much much higher than 15
minutes,
On April 3, 2005 08:13 am, Tim Pushor wrote:
To someone who has never installed OpenBSD (or FreeBSD + pf for that
matter) the learning curve is going to be much much higher than 15
minutes, although one you learn PF you will never go back!
I've never seen the great advantage to pf over ip and
Have you tried putting in some NoOp lines to verify the values of
${screenresult}?
Also, wouldn't you get the desired result by removing the 'g' option
from your Dial()?
You might want to add an 'h' extension for further processing on the
dead channel.
Hi,
From what I can see in the documentation the title of the section in
sip.conf is the username that the user logs in as. Is there a way of
seperating the names so that you can login with a normal username, but
call them with SIP/extension. Like so:
[904]
authuser=john
secret=password
etc...
Hi!
On the snom (I've tested this on the 220 and 360), the phone will
immediately reject any new INVITE that arrives with 486 BUSY HERE if
there's already a call on the phone opening
That is very interesting - can you present a review of the Snom 360
hardware, even if it is a short one?
Rod Bacon wrote:
I'm glad I'm not the only one
Now... for a solution?
Well at least this rules out a misconfiguration on the telco's end
(unless both our telco's made the same mistake). Does /anyone/ at all
have any suggestions, or is there some debug information we can send to
the list to
Dan Morin wrote:
Sorry for the double post, I tried to paste and accidently sent the
email
I've been playing with Asterisk for a few weeks now, and I've gotten
everything to work well with softphones, so I'm ready to move on to
normal VoIP phones. I've been looking around and reading
Hello all,
I was hoping to be able to call a mobile and if it is un-reachable for
whatever reason (e.g. switched off) then I was expecting an unobtainable
response that would be detected in Asterisk. It seems that the operator
(Virgin in UK) imedately completes the call and plays an automated
The packet size is a function of the number of milliseconds of sound sent
in the RTP packet. I don't know how to force * to change this, but you
*can* unilaterally change the RTP packet size on the Sipura. By doing
this, RTP packets sent by the Sipura will be larger or smaller than the
default
A google search shows exactly one reference, so it appears to exist
somewhere. It's in somebodies CVS, any ideas?
--
Martijn van Oosterhout
Ecomtel Pty Ltd
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This is traditional accross the mobile / cell providers, and there is
no real way around it.
Background : The only way to ensure that a mobile is truly there is to
page the mobile, normally based on the Mobile Switching Centre (MSC)
coverage area, and thats after looking up on the subscirbers
We have a majority of IP300's, and a few IP500's. The IP300's are great
phones if you need to simply drop in a bunch of VoIP phones quickly and
cheaply. The IP300's simply lack certain features like speakerphone
that you may want. Aside from that, its a great phone.
-Courtney
Dan Morin
Hi,
I'm trying to go route some of Asterisk users already proposed for Asterisk
minimal system. I've started from Suse Rescue system image - I've put it
into HD partition. But since rescue is spawned from working system it has
empty /boot directories and is not directly bootable if put on HD.
Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue
(audio going from the phone over wireless is slightly choppy).. while
audio coming in (20ms) is ok... where do you change it on the sipura?On Apr 3, 2005 4:07 PM, Bruce Komito [EMAIL PROTECTED] wrote:The packet size is a
On Mar 31, 2005 11:26 AM, Chuck Bunn [EMAIL PROTECTED] wrote:
I am new to Asterisk and the first thing I have noticed about Asterisk
and Pingtels open PBX's is that they are using this dinosaur method of
running forums. It is a real pain getting every message in the forum and
essentially
Never mind... blah spoke before I thought :P
Found the setting
On Apr 3, 2005 5:23 PM, Matt [EMAIL PROTECTED] wrote:Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue
(audio going from the phone over wireless is slightly choppy).. while
audio coming in (20ms) is ok...
I have to admit this still doesn't make sence.. if sipura's default is
.03ms and asterisk is 20ms.. why is the sipura dumping out around 60
frames/sec while the sipura is dumping out around 30 frames/sec??
Shouldn't the frames / packets per second go UP as the packetization gets smaller?On Apr 3,
One would hope so, but one
of the fist posts I see is someone ranting on about how if you haven't
read x or y you don't deserve an answer.
It is that kind of a social misfit that should not be welcome anywhere,
but seems to have too loud a voice here.
JN
Ty Carter wrote:
Thank
you for your
Well, you COULD use your
delete key.
You DO have one, don't you?
And you complain of others posting stupidity
JN
C F wrote:
What can be done to this shmuck?
Everytime I post anything to the list I get one of these. I'm sure
I'll get one for posting this one as well.
--
Hello, Matt!
MR fine. If you have to do any sort of transcoding a soekris is not the
MR way to go but for a small installation it works great.
Well.. Cisco's 17xx series router is a device which you can take, plug,
configure and have office PBX. But price tag is $2K.
Why the same can't be done
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, April 03, 2005 3:33 AM
Subject: [Asterisk-Users] Router with QoS recommendations
As I have a Cisco PIX 515, with NO QoS functionality,
and I'm looking for a router that does outgoing QoS to put in front of my
PIX.
PixOS
iptables looks very powerful, thats for sure.
I prefer PF's approach to security first, convenience second, and I
*really* like the fact that PF has a real parser. As the requements get
more complex, having everything in one file, and very readable and
structured is a huge plus. Also, the
Hi,
Philip Hofstetter wrote:
Now may next step has been to enable dialing out with the softphones.
This does not work as expected.
I was able to fix this problems by downgrading from kernel 2.6.11 to
2.6.10. There must be a CAPI-Problem hidden somewhere.
Last saturday was so much fun for me,
Did you try issuing show translation recalc # where # is any given
number of seconds to recalculate for? For example, speex tends to show
weird numbers for me on my dual proc xeon 2.8ghz, until I do a show
translation recalc 1, then I get more sane numbers.
Just my thoughts.
-mishehu
On Apr 3, 2005 5:45 PM, John Novack [EMAIL PROTECTED] wrote:
Well, you COULD use your delete key.
Actually nope, I can't because I'm using gmails web client to read my email.
You DO have one, don't you?
Yep I do, how did you know?
And you complain of others posting stupidity
Please read
I'm trying to get IAXTEL inbound working in my log I'm seeing all this
noise (below).
I understand I'm in DEBUG mode but I'm not doing anything yet ... what do
all these messages mean???
Apr 2 02:47:44 DEBUG[28339]: Immediately destroying 2, having received
INVAL
Apr 2 02:47:44
Look at:
http://www.voip-info.org/wiki-Asterisk+cmd+setgroup
read example 2 revised.
On Apr 3, 2005 1:20 PM, Mark Halverson [EMAIL PROTECTED] wrote:
I attempted to use the incominglimit and outgoinglimit in iax.conf and it
doesn't seem to work anylonger, running CVS-HEAD 3/16/05
So I tried
I'd put the device and another machine on a separate physical network
where you can make whatever IP configurations you need in order to be
able to send data to the IAXy. Then you can load new configuration to
it there.
There might be a better way to do i, but I don't know for sure.
-mishehu
Has anyone been successful getting a Cisco VG248 gateway
to speak MGCP with Asterisk? If so can you share either
your mgcp.conf or at least tips on getting the two devices
working together.
Thanks
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Dial(SIP/904)calls whoever logged on as john.
You could define a variable in extensions.conf.
Nabeel
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-BEGIN PGP SIGNED MESSAGE-
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Hi folks
I've a strange problem, probably a mistake but I don't see it :(
Description:
My ephone-dn number on ccme, that is a simple connection plar for all
ISDN calls, is 601
The voicemailmain on asterisk is 5900.
CCME: 192.168.17.1
*: 192.168.17.10
I put the Who? in Mishehu wrote:
Did you try issuing show translation recalc # where # is any given
number of seconds to recalculate for? For example, speex tends to
show weird numbers for me on my dual proc xeon 2.8ghz, until I do a
show translation recalc 1, then I get more sane numbers.
I
hi's
i have been trying to configure my AS5300 to work with
my asterisk box. i have tried SIP, calls come,
answered and AS5300 sends BYE message after not more
than 5 secs. I have also tried MGCP, but i believe i
am not configuring that right. here is the output of
the sip debug. please help me
hi's
i have been trying to configure my AS5300 to work with
my asterisk box. i have tried SIP, calls come,
answered and AS5300 sends BYE message after not more
than 5 secs. I have also tried MGCP, but i believe i
am not configuring that right. here is the output of
the sip debug. please help me
John Novack wrote:
An even BETTER question is: When will what is already out and more or
less working have enough accurate documentation to make it acceptable to
a wider audience?
Once more people start contributing.
As one small example: the recent postings regarding wctdm. If all the
options
Hello all. I am trying to architect a large-scale solution and need to
know some of the capabilities of * using realtime configuration (I have
read some docuemntation on the WIKI, but have not yet played with Realtime).
As the supporting docco is a little light-on at the moment, I'm hoping
to
erm, how much u willing to sell ip500?, i would like to get 1 or 2 for my
developments testing purposes. BTW if u do sell me, I'm in Malaysia, is it
a problem for u to send it over? :D
thanz.
At 04:39 AM 4/4/2005, you wrote:
We have a majority of IP300's, and a few IP500's. The IP300's are
Courtney Couch wrote:
If I were to buy 20 did's how do I know within asterisk which number was
dialed? (like say I want a few of the did's to ring specific extensions
if they are dialed and others to go through the menu)
Is there any ${var} that has the number dialed in on? (that would be
I think this problem is exactly the one I am having.
The issue is:
http://www.pastebin.com/266724
042 Found no matching peer or user for '192.168.17.1:56730'
to which asterisk generates a SIP/2.0 404 Not Found (line 057)
yet you have it configured here:
[operator]
type=peer
canreinvite=no
hello
i dont know why unixodbc is not working. i am trying
to make odbc connection. yesterday my odbc connection
was working with mysql on my one mechine but now it is
not working. is there any problem in code.
/etc/odbc.ini
[test]
Description = My test dsn
Trace = Off
TraceFile = stderr
Driver
On Apr 3, 2005 8:56 PM, Ian Hailey [EMAIL PROTECTED] wrote:
Hello all,
I was hoping to be able to call a mobile and if it is un-reachable for
whatever reason (e.g. switched off) then I was expecting an unobtainable
response that would be detected in Asterisk. It seems that the operator
(Virgin in
I put the Who? in Mishehu wrote:
I'd put the device and another machine on a separate physical network
where you can make whatever IP configurations you need in order to be
able to send data to the IAXy. Then you can load new configuration to
it there.
There might be a better way to do i, but
Mark Halverson wrote:
exten = _1NXXNXX,1,SetGroup(${CALLERIDNUM})
Try using ${ACCOUNTCODE} and make sure the account code is unique to
each phone.
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This is quite interesting.
I tested calls to 2 mobiles that I knew were off, and not diverted to
voicemail. 1 with Telstra, the other with vodafone (I'm in Australia). Via
ISDN, both calls were shown as unanswered by asterisk. When the calls went
to voicemail, the call was deemed to be
Empty yer bloody mailbox...
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hello
can any one tell me what is the problem in my odbc
connection.
here is my sql.log connection with mysql is working
and with freetds is giving me error jawad is one
windows server having MS Sql server
#isql kdsn
src/tds/login.c: tds_connect: jawad:1433: Connection
refused
[ISQL]ERROR: Could
Rod Bacon wrote:
This is quite interesting.
I tested calls to 2 mobiles that I knew were off, and not diverted to
voicemail. 1 with Telstra, the other with vodafone (I'm in Australia).
Via ISDN, both calls were shown as unanswered by asterisk. When the
calls went to voicemail, the call was
If u want some help put your 53xx and sip config files.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jafar mohammed
Sent: Sunday, April 03, 2005 9:41 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AS5300+SIP+ASTERISK or AS5300+MGCP
Hi,
If this belongs on a different list, please let me know.
I oversee an Altigen IP-based PBX. We're wanting to make VoIP calls
through the Internet out to PSTN via a service like BroadVoice or
similar. I think Asterisk is the ticket of this.
I have successfully configured Asterisk to
-Original Message-
[mailto:[EMAIL PROTECTED] On Behalf Of
Dan Perik
Subject: [Asterisk-Users] Asterisk - Altigen
Has anyone successfully tied together an Altigen system to an
Asterisk system using VoIP (ie. not using hardware (FXO/FXS
cards, etc.))?
My experience with the
Hi,
I am looking at a project using asterisk for a particular purpose. We
already are using an Asterisk box for things like voicemail, call
recording, ip phones etc. and it connects to an old standard PBX through
ZAP.
What I am looking to do is have calls coming into asterisk via either
VOIP
Hi everyone
Presently all our calls are channel to one provider and we would like to change that based on LCR.
the following is what we have presently;
# Dial the requested number, if we got something from the caller.if ($dialto != ""){ $AGI-exec('SetAccount', $accountnum); if ($debug) {
Greetings!
This is my first post to the list...and I'm kinda' new to Asterisk, so
please be kindI did a fair amount of Googling but was not able to
find an answer.
I am using [EMAIL PROTECTED] 0.8
I was wondering if there is a way to select the outbound trunk based
on the extension that
Greetings!
This is my first post to the list...and I'm kinda' new to Asterisk, so
please be kindI did a fair amount of Googling but was not able to find
an answer.
I am using [EMAIL PROTECTED] 0.8
I was wondering if there is a way to select the outbound trunk based on the
extension that
to a load-balanced (not sure which mechanism I'll empoy here yet)
I was quoted a $21,000 layer-7 switch from F5 Networks to do SIP load
balancing.
outside)? In other words, can the registering server update a USRLOC
type database on the fly, so all other servers know where to route calls
I was wondering if there is a way to select the outbound
trunk based on the extension that making the call.
Set the context in the sip.conf file for that user to a context in
extensions.conf that only has entries for dialing out through specific
providers.
Nabeel
Hi,
I have a problem with ATA-186 configured for silence supression (AudioMode
bit 0 = 1). When enabled and listening music on hold no sound is heared (if
I talk I began to hear the music and again mutes when I stop talking).
If I configure for silence supression off everything goes fine. Any
On 03/23/05 04:15 Jesse Guardiani said the following:
This should be has some issues. I do not consider
the FreeBSD zaptel support to be production quality
in any way. I experienced reproducible system hangs
(mostly after an asterisk restart), interrupt issues
(audio skips and SSH pauses during
Hi everyone,
I just install Linux and asterisk on one of my pc. I
want to run some basic functionality tests.
Is it possible to use a v92 modem as a FXO or FXS
card. If yes how do we configure and install the card?
What are the commands?
Thanks in advance for your help
AC
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