Bad form to reply to your own post, but even worse form when you can't
read the screen.
Try StopMixMonitor :)
Julian
Julian Lyndon-Smith wrote:
Is there an equivalent stopmonitor command if you are using MixMonitor ?
StopMonitor does not seem to have an effect on MixMonitor
Julian.
On Jun 21, 2006, at 6:31 PM, Rob Thomas wrote:
Yeah, I know. I was just grumpy. I shouldn't have whinged. I think if
you'll look at the time I sent it, it was close to 2AM after spending
all afternoon, night, and morning, fighting with dialplan 8)
So, definately, my whinge was unneccesary
Kristian Kielhofner wrote:
Mike Fedyk wrote:
I happen to have asterisk running as a router, so I use it doing QoS
with tc (traffic control) and wondershaper set to prioritize based on
port ranges. I sent a patch to the debian bug tracking system a
while back with a few improvements -- I
Title: SIP Channel hangup problem with re-INVITE enabled - ugrent
Hi List
I have UAs registered with Asterisk and make outbound calls via ITSP1, everything is fine without re-INVITE. When people call 178, the actual number 112233445566 at ITSP1 network will be called.
When UA or called
Matt wrote:
Hi,
I'm still in the process of debugging this, but I have a gotoif
statement that looks like this:
exten = 26,1,GotoIfTime(7:00-18:00|mon-fri|*|*?ext-queues,210,1)
exten = 26,n,Goto(ext-local,${VM_PREFIX}127,1)
I had similar problems, you have your time statement incorrect.
Hi,
I'm straggling with setting up a call via manager interface. Basic
functionality works fine but I try to use this addons:
Application: Playback
Data: beep
when a call is answered by A side, 'beep' is played correctly but no
further action is taken - I got hangup !!!
Why it's not
Hi. you use a Plantronics CS-50 USB headset.I use with Eyebeam and the quality is very good.
Best Regards
Josué
2006/6/22, Martin Joseph [EMAIL PROTECTED]:
On Jun 21, 2006, at 10:35 AM, Michael Graves wrote: With few exceptions a USB phone is just and audio device to the host
PC. Most will work
Hi all,
Is there any such as tools for multi call generation
to test, how much call can be done via Asterisk?
_
Best Regards,
---
Abdul Lateef
Nepal
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam
You must create an extension in dialplan and use this extension in your
originate script. So you can do whatever you want in that extension.
-Original Message-
From: lokotes [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 22, 2006 1:30 PM
To: asterisk-users@lists.digium.com
Subject:
sipsak is ok for that
Olivier
Abdul Lateef a écrit :
Hi all,
Is there any such as tools for multi call generation
to test, how much call can be done via Asterisk?
_
Best Regards,
---
Abdul Lateef
Nepal
__
Do
I wanted to get everyones opinion
on an issue I am having.
I am currently using linksys PAP2NA ATA adapters
to terminate analog calls from my auto dialer to the voip termination co. The
problem I have is when I call the PSTN everything goes fine until the person
being called hangs up
I wonder if some PRI vendors send different signaling and the bug fix in 2005
only dealt with a certain scenario.
--
--
Steven
http://www.glimasoutheast.org
Nabeel Jafferali [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Up here in Toronto, on a PRI to an Asterisk box with a
Very famous in SIP world is
SIPP - http://sipp.sourceforge.net/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abdul Lateef
Sent: Thursday, June 22, 2006 6:12 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Multi Call Generation
Interesting.
As I'm doing more debugging it doesn't seem like it has anything to do
with the first call triggering something. It seems more like 1800
actually means 1802! Yes, I checked the server clock and it is right,
however it seems the time based routing doesn't kick over for a minute
or
We're now back on 1.2.6 and running stable. Been running for over 17
hours. Something is wrong with 1.2.9.1
On 6/21/06, Matt [EMAIL PROTECTED] wrote:
I'm now wondering if the changes I made were NOT the cause of my problem.
I was just tweaking some settings (on a virgin 1.2.9.1 (no
On Thu, 2006-06-22 at 13:16 +0200, olivier.taylor wrote:
sipsak is ok for that
Olivier
sipp.sf.net is also not a bad product. They both work slightly
differently so it depends on what exactly you need.
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE
Not sure if this is the right forum for this... I need a reliable but "cheap" toll free number to forward to my Voicepulse connect account for incoming calls. Any suggestions? Also, is there anything I need to do in the asterisk configs (IAX) for forwarding toll free for incoming calls?
On 6/22/06, Matt [EMAIL PROTECTED] wrote:
We're now back on 1.2.6 and running stable. Been running for over 17
hours. Something is wrong with 1.2.9.1
Sorry. I may have asked this already, but are you running the tarball
releases or checkouts from SVN? I've seen some similar behavior on
some
If I have an incoming call which uses G.711u, which then gets
transferred to a phone which has G.729 selected as its first preference
(with 711 as a third).
Is it normal behaviour for asterisk to transcode the call to G.729
rather than keep it as 711?
Does anyone know if when T.38 support is
And once again I am reminded why I shouldn't bother helping
people here.
Not even a 'thanks'.
Thanks againm no problem, but I think that:
1) I already thank you in my previous message:
(Upgrade to freePBX 2.1.1, it's much better, really)
I upgraded to custom, 'made with vi' files,
BJ Weschke wrote:
On 6/22/06, Matt [EMAIL PROTECTED] wrote:
We're now back on 1.2.6 and running stable. Been running for over 17
hours. Something is wrong with 1.2.9.1
Sorry. I may have asked this already, but are you running the tarball
releases or checkouts from SVN? I've seen some
Hi Joao,
What is the T3 timeout set to? If it is short, try making it something
like 15000 (it is in milliseconds). Your T3 timeout problem could be due
to slow reponses from some destinations.
For your second issue, the following is definitely a protocol error:
UniCall/1 - 1 off
Dear
I am using [EMAIL PROTECTED] , and I have 2 hard disks on the
system ,how can I put the database (CDR) on the second hard disk .
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium
John,
I'm glad it worked for you.
Correct me if I'm wrong, but I believe that s will only workfor a macro.
Good luck with the outbound, if all else fails... givebroadvoice a call. They **may** have example conf files for you to look at for interconnecting with them.
bp
On 6/21/06, John
I am still convinced there is some sort of memory leak caused by either
heavy use of the manager interface or reloading repeatedly. I have cut
back on both and have not noticed any of this funkiness since (48 hrs).
My system never stops processing calls but it slows. On the CLI, it may
take
And the opposite is also true. I've tried a few Skype Certified
devices, and generally the audio works (though in some cases it's (IMO)
unnecessarily turned off unless Skype is in a call, perhaps so you don't
hear the beeps and squeaks a PC normally makes) And almost universally,
a Skype
I'm running just the tarball 1.2.9.1 now the tarball of 1.2.6.
On 6/22/06, BJ Weschke [EMAIL PROTECTED] wrote:
On 6/22/06, Matt [EMAIL PROTECTED] wrote:
We're now back on 1.2.6 and running stable. Been running for over 17
hours. Something is wrong with 1.2.9.1
Sorry. I may have asked
Yes, she is registered, and her status reads as ok. She receives calls
fine.
[EMAIL PROTECTED] wrote:
Message: 13
Date: Wed, 21 Jun 2006 17:48:34 -0400
From: Tom Vile [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: User Loses Ability to Make Outgoing
Calls
To: Asterisk Users Mailing
This worked great.
I made an extension 1 in context intercom and set my custom Goto
statement there (first added the SIPHeader mod)
[intercom]
exten = _XXX,1,SIPAddHeader(Alert-Info: Auto Answer)
exten = _XXX,2,Goto(from-internal,7${EXTEN},1)
and added that extension (and had it not
aha. I have it the other way around:
Dial(IAX2/iaxmodem/${CALLERIDNUM})
anyway, it works perfecly fine so I think I'll leave it, the administrative
overhead (other than setting it up in the first place) is low, and the
system is very reliable.
-Original Message-
From: Lee Howard
Hi all,
I use asterisk 1.2.6 with queues.
This is my queue entry in queues.conf:
[460]
strategy = ringall
servicelevel = 60
context = reception
timeout = 25
retry = 2
maxlen = 0
announce-frequency = 0
periodic-announce-frequency = 25
announce-holdtime = no
periodic-announce =
Welcome to Broadvoice. Get used to it. It will happen often. I loved the
price and was a customer for almost 2 years. Recently I changed over to
teliax where of course I connect using IAX and I have never been happier.
Its like 24.95 a month unlimited calling instead of 19.99 I was paying at
Hello
You have announce-frequency = 0
That would mean no announcements.
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Michiel van Baak
Sendt: 22. juni 2006 16:08
Til: asterisk-users@lists.digium.com
Emne: [Asterisk-Users] periodic-announce not
HiThis works fine in extensions.conf:exten = _0X./100,1,Dial(SIP/[EMAIL PROTECTED])exten = _0X./200,1,Dial(SIP/[EMAIL PROTECTED])This will just use different SIP channels for different Caller ID's.
If I write the same to a realtime table, Asterisk always uses sipout-a, no matter what Caller ID is
On 16:36, Thu 22 Jun 06, Jon Sch?pzinsky wrote:
Hello
You have announce-frequency = 0
That would mean no announcements.
But if I set announce-frequency it will tell the ppl waiting
in the queue where they are. We dont want that.
We simply want a message every 25 seconds: please hold or
press
On 16:57, Thu 22 Jun 06, Benjamin Stocker wrote:
Hi
This works fine in extensions.conf:
exten = _0X./100,1,Dial(SIP/[EMAIL PROTECTED])
exten = _0X./200,1,Dial(SIP/[EMAIL PROTECTED])
This will just use different SIP channels for different Caller ID's.
If I write the same to a realtime
2006/6/22, Michiel van Baak [EMAIL PROTECTED]:
On 16:57, Thu 22 Jun 06, Benjamin Stocker wrote: Hi This works fine in extensions.conf: exten = _0X./100,1,Dial(SIP/[EMAIL PROTECTED]) exten = _0X./200,1,Dial(SIP/[EMAIL PROTECTED]
) This will just use different SIP channels for different Caller ID's.
On 6/22/06, Michiel van Baak [EMAIL PROTECTED] wrote:
On 16:36, Thu 22 Jun 06, Jon Sch?pzinsky wrote:
Hello
You have announce-frequency = 0
That would mean no announcements.
But if I set announce-frequency it will tell the ppl waiting
in the queue where they are. We dont want that.
We
I am using Polycom 501s with asterisk 1.2.4.When transfering to call parking wih "#1" - 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" - blind - 700 . The user is not able to hear what extension the call was parked on.
Hi,
I'm having problems with an occasional disconnect from phone calls while
my phone is on mute. This is a problem with long conference calls, for
example. I've a GrandStream GXP-2000 and Asterisk 1.2.1. Anyone have
experience with similar issues?
Best, WILL
Title: Re: [Asterisk-Users] when I press transfer - blind - 700 . The user is not able to hear what extension the call was parked on
This is the way blind transfers work. The transferring party doesnt get to hear anything. For call parking, you have no choice but to use supervised transfer if
Hi guys,
Not a so technical question for the moment :) but I would like to share
some experience with people that have done some deployments on:
- big call center (300 call agents), 4 queues, high availability
- SER and Asterisk integration
- vPBX
Feel free to contact me directly.
I will be on vacation from 22/06/06 to 30/06/06.
I will not be reachable on my mobile. I will have limited access to mails, and
please expect a delayed response.
In my absence, please contact the following:
Ray Richard or Safeer Mohammed
Thanks
H.Gireesh
However, when I press transfer - blind - 700 .
The user is not able to hear what extension the call
was parked on.
It's blind - so it's working as expected. On a blind transfer the phone
set disconnects as soon as you press blind. Just make sure you do a
supervised transfer instead.
Hello,
I read ag-projects it does'not provide support for its
cdrtool pseudo gpl licensed .
It's one of the bad open source project even seen .
No documentation ...
A real bad project !!
Don't waste time with that
Anybody could advise me a billing system based on cdr
records with realtime
On Thursday 22 June 2006 11:24, sdgesa gaeharth wrote:
transfers or call parking. This problem just started happening a few weeks
ago. Before then , blind transfer worked fine. It must be a config issue
somewhere
What did you change? Can you roll back and get it to work properly again? If
sdgesa gaeharth wrote:
I am using Polycom 501s with asterisk 1.2.4.
When transfering to call parking wih #1 - 700 the user is able to
hear asterisk tell him what extension the call was parked on.
However, when I press transfer - blind - 700 . The user is not
able to hear what extension the
You got to be freaking kidding, a month of this?
Cant we get an easy process for the list owner to take care of these?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, 22 June 2006 11:45 AM
To:
On Jun 22, 2006, at 5:22 PM, BJ Weschke wrote:
Submit a feature request/patch to bugs.digium.com. There isn't
presently a way to do hold time announcements without queue position
along with it.
I dont want hold time neither.
I simply want 1 file played to the ppl waiting in queue every 25
Title: Re: [Asterisk-Users] when I press "transfer" - blind - 700 . The user is not able to hear what extension the call was parked on
I have blindxfer = #1 set in features.Doesn't this means #1 is the same as transfer - blind, correct? Both are blind transfers..Is so, why when I
Should only happen once if his email system is config'd in a standard
method. Otherwise just *plonk* his address.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Thursday, June 22, 2006 12:03 PM
To: Asterisk Users
Is it possible to get Joburg DIDs (probably need 4 at the moment), to be
delivered via SIP preferrably to UK.
If it's legal, please send pricing.
Thanks
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Actually, if his MTA is configured properly, it shouldn't happen at
all.
A.
On Jun 22, 2006, at 9:32 AM, Doug Geary wrote:
Should only happen once if his email system is config'd in a standard
method. Otherwise just *plonk* his address.
-Original Message-
From: [EMAIL
Does anyone out there have a recommendation for tools that
will monitor the quality of VoIP systems? I am looking for jitter and MOS
monitoring. I have a custom Nagios plugin that is alerting me if the jitter
jumps out of a 20ms but I am looking for a little more detail. I would not be
On Thu, Jun 22, 2006 at 05:47:47PM +0100, Steve Kennedy wrote:
Is it possible to get Joburg DIDs (probably need 4 at the moment), to be
delivered via SIP preferrably to UK.
If it's legal, please send pricing.
And that should have gone to the biz list, sorry.
Steve
--
NetTek Ltd UK mob
Hello,
We've released another update to our Asterisk GUI Client suite: 1.1.12
http://astguiclient.sourceforge.net/
The client suite runs on most modern web browsers on almost any
GUI-capable operating system, and it includes the astGUIclient
client-side web app which extends your phone's
Hi,
I've got some problems with bridged calls, the quality is extremely poor
(more or less blanks or one way voice issues). But if I do a normal call
with the same provider, there is no problem.
Reinvite is enabled, but what are the parameters in the dial command
that force asterisk to stay
He's probably using Exchange which has a global setting to either send OOO
replies to SMTP addresses or not. It's a dumbass Exchange administrator who
enables this option (it is actually on by default)
Same thing happened to the mac-asterisk list last week, except the OOO
message would reply to
On 6/22/06, Michiel van Baak [EMAIL PROTECTED] wrote:
On Jun 22, 2006, at 5:22 PM, BJ Weschke wrote:
Submit a feature request/patch to bugs.digium.com. There isn't
presently a way to do hold time announcements without queue position
along with it.
I dont want hold time neither.
I simply
Hey all. We have a DS3 circuit with GBLX split off into 7 systems with
a 4 port sangoma card (A104D) in the first 2 systems, and digium T410P
cards in the other 5. GBLX numbers their spans from 0 to 3 instead of
1-4 and we have a NFAS configuration with the d-channel on chan 96. All
of our
On Thu, 22 Jun 2006, BJ Weschke wrote:
On 6/22/06, Matt [EMAIL PROTECTED] wrote:
We're now back on 1.2.6 and running stable. Been running for over 17
hours. Something is wrong with 1.2.9.1
Sorry. I may have asked this already, but are you running the tarball
releases or checkouts from
Hi,
We run a small
switchboard using Asterisk and Free PBX.
We have two main
extensions and two ring groups. The first ring group rings the two internal
extensions. If the internal extensions do not pick up the call after 15 seconds
then the second ring group kicks in which should ring
Has anyone been able to get PHP-SNMP working on an asterisk box. I have
downloaded the net-snmp, utils,libs, perl and php-snmp but unable to get the
php to work wit it. It works via command line // without php. This is set
up on an AAH box.
___
Care to share your Nagios plugin?
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Jun 22, 2006, at 9:53 AM, Curt Shaffer wrote:
Does anyone out there have a recommendation for tools that will
On Jun 22, 2006, at 10:12 AM, Colin Anderson wrote:
He's probably using Exchange which has a global setting to either send
OOO
replies to SMTP addresses or not. It's a dumbass Exchange
administrator who
enables this option (it is actually on by default)
Same thing happened to the
Anybody have any more information on this Dial() d option for incoming calls?
On 6/19/06, John Klimek [EMAIL PROTECTED] wrote:
Thanks for the information...
After doing some reading it looks like I can use the d option with
the Dial() command to be able to enter a 1-digit extension while the
We use MS Exchange too and, as far as I am aware, it is cognizant of
mailing list headers and doesn't send OOO notices to mailing list
postings. The only mailing list from which I receive my own OOO notices
is one that doesn't have the proper mailing list headers set.
When you receive a lot
Once
I'm inside the the asterisk CLI and I'm on a call with another extension, how do
I play sounds from the CLI? It doesn't appear that I can run AGI commands
directly -- is there another way that I'm missing?
thanks,
JJ
___
--Bandwidth and
How can I get the external caller id to show on the polycom 501 phones. Currently, when someone calls our office, we only see the word "asterisk" in the caller id.This is our set up:VOIP(polycom)---Asterisk 1.2.4---PSTNThanks
Yahoo! Groups gets better.
Can someone
recommend the best way to view current calls in progress on the Asterisk
console?
Neither the 'show
channels' or 'sip show channels' commands are easy to read.
hestia*CLI show
channelsChannel
Location
State
Application(Data)
SIP/2944093-f9e2
(None)
Up Bridged
Andy Brezinsky wrote:
Hey all. We have a DS3 circuit with GBLX split off into 7 systems
with a 4 port sangoma card (A104D) in the first 2 systems, and digium
T410P cards in the other 5. GBLX numbers their spans from 0 to 3
instead of 1-4 and we have a NFAS configuration with the d-channel on
On Jun 22, 2006, at 7:18 PM, BJ Weschke wrote:
On 6/22/06, Michiel van Baak [EMAIL PROTECTED] wrote:
On Jun 22, 2006, at 5:22 PM, BJ Weschke wrote:
Submit a feature request/patch to bugs.digium.com. There isn't
presently a way to do hold time announcements without queue
position
along
d: This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to be answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered - see also
On 6/22/06, John Klimek [EMAIL PROTECTED] wrote:
Anybody
It is really just a play on the check_icmp plugin. You could accomplish the
same thing by doing the following:
$USER1$/check_icmp -H $HOSTADDRESS$ -w 80.0,80% -c 100.0,100% -n 1
Where in this example it is an RTA of 80ms or 80% packet loss for a warning
and 100ms or 100% packet loss for
hello to all,
I advice you to not use CDRtool from ag-projects :
Fisrt ag-projects talk about is product like a gpl
software however they don't provide at least some
documentation for non commercial users .
try to call them !!
i'll offer you some money .
You can not Call them for some advices
Tzafrir Cohen wrote:
On Wed, Jun 21, 2006 at 05:21:17PM -0700, Carlos Munoz wrote:
Tzafrir Cohen wrote:
On Wed, Jun 21, 2006 at 03:46:15PM -0700, Carlos Munoz wrote:
I'm unable to configure asterisk to provide dial tone, busy tone, detect
dtmf digits, etc to an analog phone
Do you have the Caller ID feature with your telephone service package?
sdgesa gaeharth wrote:
How can I get the external caller id to show on the polycom 501
phones. Currently, when someone calls our office, we only see the word
asterisk in the caller id.
This is our set up:
On 6/22/06, Michiel van Baak [EMAIL PROTECTED] wrote:
On Jun 22, 2006, at 7:18 PM, BJ Weschke wrote:
On 6/22/06, Michiel van Baak [EMAIL PROTECTED] wrote:
On Jun 22, 2006, at 5:22 PM, BJ Weschke wrote:
Submit a feature request/patch to bugs.digium.com. There isn't
presently a way to do
Any idea why it wouldn't work in my dial plan?
On 6/22/06, Peter Antonacci [EMAIL PROTECTED] wrote:
d: This flag trumps the 'H' flag and intercepts any dtmf while waiting for
the call to be answered and returns that value on the spot. This allows you
to dial a 1-digit exit extension while
Hi all, I have a Soekris net4801-50 board with OpenBSD 3.9 where I've
configured a dhcp server and tested it with a regular PC connected
directly via a crossover cable with success. The problem comes when I
try to connect my IAXy device instead of the PC. I can see with 'tcpdump
-nettti sis1'
I will agree that switching to the TDM card significantly helped my echo
and sound quality, I would take a second to point out that interrupt
sharing on your * server might cause crackling-like noises. Try
lspci -vb
and
cat /proc/interrupts
to see if you discern any hardware using the same
[EMAIL PROTECTED] wrote:
hello to all,
I advice you to not use
Harry!!
Only one post is needed for each of your silly complaints.
Please don't give people even more reason to relegate you to their
killfiles.
B.
--
This message has been scanned for viruses and
dangerous content by
Has anyone created a script that will download and install all of the
freepbx prerequisites in the INSTALL file automatically on a Centos 4 box?
TIA,
W
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
Whats wrong with show channels?
On 6/22/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Can someone recommend the best way to view current calls in progress on the
Asterisk console?
Neither the 'show channels' or 'sip show channels' commands are easy to
read.
hestia*CLI show channels
Channel
Great - thanks, Curt!
A.
On Jun 22, 2006, at 11:30 AM, Curt Shaffer wrote:
It is really just a play on the check_icmp plugin. You could
accomplish the
same thing by doing the following:
$USER1$/check_icmp -H $HOSTADDRESS$ -w 80.0,80% -c 100.0,100% -n 1
Where in this example it is an RTA
Hi,I would like to monitor in realtime the status of a given sip channel with the manager API and a web page. What is the better way to do that without using Asterisk
Flash Operator Panel ?Thanks in advance.Ronan
___
--Bandwidth and Colocation provided
replace the beep sound file with a silent one :) I think it's beep.gsm
James Harper wrote:
I want to have something for the kids to play with which just records
until silence is detected, plays back what was recorded, then repeats.
They are having fun with Echo() at the moment :)
I have
The options are not seperated by commas.
exten = s,1,Dial(SIP/50,23,r,d)
should be
exten = s,1,Dial(SIP/50,23,rd)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Klimek
Sent: Thursday, June 22, 2006 2:59 PM
To: Asterisk Users Mailing List -
Using this as an example:
hestia*CLI show channels
Channel Location State Application(Data)
SIP/2944093-f9e2 (None) Up BridgedCall(SIP/2944079-e7f2)
SIP/2944079-e7f2 [EMAIL PROTECTED]:2 Up Dial(SIP/2944093|36|tr)
Why does the first line
Crackling is usually a sign of IRQ issues, as Mojo wrote. Digium's full
documentation on solving IRQ issues are here:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
Regarding echo on POTS lines, I wish you the best of luck. Fixing your
IRQ problems may reduce the delay
On Thu, Jun 22, 2006 at 11:38:42AM -0700, Carlos Munoz wrote:
Tzafrir Cohen wrote:
On Wed, Jun 21, 2006 at 05:21:17PM -0700, Carlos Munoz wrote:
Tzafrir Cohen wrote:
On Wed, Jun 21, 2006 at 03:46:15PM -0700, Carlos Munoz wrote:
I'm unable to configure asterisk to
The first line has (None) as the location because a PBX is not running on it
as it was created by the channel below it using the Dial application. As for
the BridgedCall(SIP/2944079-e7f2) part that's to indicate it's bridged to
that channel. Anything could have been put in that space since no PBX
This post cannot be left without comment. People who don't know you or Adrian
might get a wrong impression.
I know Adrian quite well and know that he is one of the real experts in this
industry and he and his stuff does not deserve such a treatment.
I would recommend that you change your
On the asterisk1 I got this:
register = username:[EMAIL PROTECTED]
[eop]
username=username
secret=secret
type=peer
host=ipaddress1
auth=md5
on the second box I got this
this host is ipaddress2
[incommingiax2]
username=username
type=user
secret=secret
host=dynamic
context=from-internal-custom
Hi all
There seam to be a very short timeout waiting for digits being dialed. (about
6 seconds).
Is there a way to increase that time? I have a phone with integrated address
book and my fingers are just not fast enough to open the menue, select an
entry and hit 'dial'.
-Benoit-
On 22 Jun 2006, at 22:11, Christian Stredicke wrote:
This post cannot be left without comment. People who don't know you
or Adrian might get a wrong impression.
Honestly, I think it can. That post tells you everything you need to
know about the camplaining party ;)
jens
Has anyone created a script that will download and install all of the
freepbx prerequisites in the INSTALL file automatically on a Centos 4
box?
In a manner of speaking the trixbox guys have. Have you ever seen that
(or Asterisk @ Home)? There is a script, install.sh, that installs a
bunch
I am thinking of getting an asterisk user group together for either SE
Michigan or just Metro-Detroit.
How much interest in asterisk in Michigan is there on this list?
I am already on the board of glimasoutheast, with is a group for
technology professionals. (very broad range)
It is a spin-off
Please can anyone advise what these messages mean?
Jun 22 21:38:07 ERROR[2785]: chan_sip.c:11323 sipsock_read: We could NOT
get the channel lock for SIP/213.xxx.5.xxx-0816e1b8!
Jun 22 21:38:07 ERROR[2785]: chan_sip.c:11324 sipsock_read: SIP MESSAGE
JUST IGNORED: ACK
Jun 22 21:38:07 ERROR[2785]:
Kevin P. Fleming wrote:
I believe this is incorrect; all the RTP-using channel drivers supply
'ast_rtp_bridge' as their native bridge method, so assuming they also implement
the 'set_rtp_peer' method, then an RTP native bridge between dissimilar
channels should work fine. If the channel
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