yes with anybody interesting in ip phones for
Asterisk
So you can send them a childish insult? I wrote you off list as
requested, and you wrote back something a 10 year old would say in
school. What is your point, Harry?
___
Asterisk-Users mailing
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Monday, June 27, 2005 8:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RTP session between two end users
Erdem HAKİ wrote:
However, it's not really passing the called number per say. What
it's doing is putting the extension I have in my register statement
into the To field. I'm assuming the To field is actually being
populated with whatever * set the Contact field to when it
registered.This seems to mean
On Tue, Jun 28, 2005 at 01:18:18PM +0800, Ronald_Wiplinger wrote:
Can anybody give me a hint, what I am doing wrong, please?
Asterisk and H.323
1. download all parts
-
wget
http://www.inaccessnetworks.com/asterisk-oh323/download/asterisk-oh323-0.7.1.tar.gz
Wilson Pickett,
I posted this mail for people interested in polycom
ip300 for asterisk.
Harry
--- Wilson Pickett [EMAIL PROTECTED] a écrit :
yes with anybody interesting in ip phones for
Asterisk
So you can send them a childish insult? I wrote you
off list as
requested, and you wrote
Hello
I am using Asterisk 1.0.5 with i4l, a HFC
ISDN BRI card, and some Grandstream products. The system works fine except that
some external telephone numbers, when dialed always give a busy tone whereas
other numbers are fine. Ive checked extensions.conf and even tried hard
coding the
Hello
I have asterisk running in Red Hat 9 with a cheap HFC card on i4l. I have
choppy sound problems sometimes, and echo problems often. I am using a 2
port Grandstream ATA, Grandstream BT and a Grandstream GPX-2000
I read that changing to BriStuff will fix the echo problems, but have also
read
In article [EMAIL PROTECTED],
Ronald_Wiplinger [EMAIL PROTECTED] wrote:
Can anybody give me a hint, what I am doing wrong, please?
Asterisk and H.323
1. download all parts
-
wget
http://www.inaccessnetworks.com/asterisk-oh323/download/asterisk-oh323-0.7.1.tar.gz
On 6/27/05, Andres [EMAIL PROTECTED] wrote:
However, it's not really passing the called number per say. What
it's doing is putting the extension I have in my register statement
into the To field. I'm assuming the To field is actually being
populated with whatever * set the Contact field
Looks like the router around them in the Denver area have been having issues
or overloaded. I have been getting choppy calls all day but seems to be
better tonight.
Rick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Megacz
Sent: Monday, June 27,
Tzafrir Cohen wrote:
On Tue, Jun 28, 2005 at 01:18:18PM +0800, Ronald_Wiplinger wrote:
Can anybody give me a hint, what I am doing wrong, please?
Asterisk and H.323
1. download all parts
-
wget
On 28 Jun 2005, at 01:34, Eric Wieling aka ManxPower wrote:
I have no idea. But since it's NOT the same part number, I would
assume no. Perhaps a call to Digium would be in order?
Florin Mandache wrote:
As is on that page :
D/300JCT-1E1 E1 + 30 voice so is compatible ??!??
Hi
As far as I know, only the server versions of Eicon work with
asterisk (using chan_capi).
There are a few other BRI cards that work with asterisk. Junghanns
cards seem to work the best from the little I have seen.
Good luck.
Regards
Clive
On 27 Jun 2005 at 23:19, [EMAIL PROTECTED] wrote:
in routes pattern i tried to write pattern to usa
destination and that was 1* it worked well but when i
wanted to specify the number of digits then i tried
1NXXNXX but i didnt work.so i dont know what to
write please help.
__
Do You Yahoo!?
I use AVM fritz card A1, I have install with instruction of
Voip-info.or, but it don' work.
capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.101-02 (49.18)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal
Tony Mountifield wrote:
I corrected according to your suggestion:
wget
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz
wget
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz
These are not just
Sorry
it's asterisk-users@lists.digium.com
--- harry gaillac [EMAIL PROTECTED] a écrit :
Luca,
you may find help here:
http://www.cs.colostate.edu/~somlo/CSU-SIP-notes/
http://www.asteriskdocs.org/
http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large
ask for help to
Hi all,
Sorry for this elementary question (I'm a newbie).
I'm trying to write an agi script (test.agi) and run it when I call
in. However, I'm getting an error that says application agi isn't
being found. I've put test.agi into agi-bin with permissions 755.
Do I have to compile agi support
Hi,
Bristuff works great with HFC card... your compilation problem may come
from your kernel configuration...
You should check this doc, at least, for the redhat config :
http://www.automated.it/guidetoasterisk.htm
Then, installation of Bristuff works like as charm !
Bye
David Masure
Hello,
How can i learn my asterisk how many simulyaneus calls support?
My configuration: 80 GB HDD, 1 GB Ram, P4 2,8 MHz processor,
Fedora Core 3 minimum installation, no digium cards, codecs g729 or gsm, 1Mbit
internet connection.
Thanks for your interest...
Erdem HAKI
Ronald_Wiplinger wrote:
Tony Mountifield wrote:
I corrected according to your suggestion:
wget
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz
wget
http://www.inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz
try using asterisk-oh323-0.7.2-pre1 it compiles with latest CVS
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Hi all,
If they are people on this list using this feature, I kindly ask them
to send me their feedback.
I am interested to support multiple GSM phones and standard voice
modems.
More, if it is someone with much more experience regardinig direct
using of bluetooth audio in an
application
Title: Spinlock with ZAPTEL
Hi, I'm using Fedora Core 3 and the 1.0.8 version of ZAPTEL. Why do I get spinlocks when I modprobe -r it but the HEAD version unloads fine?
Regards
Lee
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.323 / Virus Database:
On Mon, 2005-06-27 at 22:51 -0700, hank wrote:
mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/
I haven't tried this myself but if I put www.waixwave.com:8000 in
Firefox I get connection refused. Try another site that actually
streams music. Shoutcast.org should have a nice
On Tue, 28 Jun 2005, sylvain garcia wrote:
I use AVM fritz card A1, I have install with instruction of
Voip-info.or, but it don' work.
Which version of Asterisk and chan_capi do you use?
How does your capi.conf look like?
Armin
capiinfo
Number of Controllers : 1
Controller 1:
Hello
I have been buttling to get a Valiant VCL-30 E1 channel bank multiplexer
talking to asterik's E100P card.
The Spec for the VCL says this:
Coding: hdb3
Signalling: CAS
Timing: can be internal, external sync
Nominal Impedance: 120ohm balanced, 75ohm unbalanced
supports euroisdn
In the
Hi all the list,
I am searching how to insert few seconds of silence just before to send the
DTMF sequence via a FXO WildCard X101P to PSTN.
I remember that Hayes compatible modems knows a special character W that
do a 1 sec pause.
Is it possible to do something like this in DIAL line sequence ?
[EMAIL PROTECTED] wrote:
Hi all the list,
I am searching how to insert few seconds of silence just before to send the
DTMF sequence via a FXO WildCard X101P to PSTN.
I remember that Hayes compatible modems knows a special character W that
do a 1 sec pause.
Is it possible to do something like
Hello,
I seem to have a strange problem, which appeared out of nowhere.
Did anyone see something like this?
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel:
Hello,
I seem to have a strange problem, which appeared out of nowhere.
Did anyone see something like this?
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal
Armin Schindler a crit:
On Tue, 28 Jun 2005, sylvain garcia wrote:
I use AVM fritz card A1, I have install with instruction of
Voip-info.or, but it don' work.
Which version of Asterisk and chan_capi do you use?
How does your capi.conf look like?
Armin
On Tue, 28 Jun 2005, sylvain garcia wrote:
Armin Schindler a écrit :
On Tue, 28 Jun 2005, sylvain garcia wrote:
I use AVM fritz card A1, I have install with instruction of
Voip-info.or, but it don' work.
Which version of Asterisk and chan_capi do you use?
How does your
Hello,
I seem to have a strange problem, which appeared out of nowhere.
Did anyone see something like this?
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel: Illegal seek
2005-06-03 18:44:50 WARNING[10721] app_meetme.c: Unable to write frame
to channel:
Armin Schindler a crit:
On Tue, 28 Jun 2005, sylvain garcia wrote:
Armin Schindler a crit :
On Tue, 28 Jun 2005, sylvain garcia wrote:
I use AVM fritz card A1, I have install with instruction of
Voip-info.or, but it don' work.
On Monday 27 June 2005 23:04, David John Walsh wrote:
Hello,
does anyone know how to get the say number (say.c) agi application
to work in french [assuming that I have the French voice files]
I have looked in the code and about a 1/3 of the way thru there is :
} else if
my number with them is ok.
Jason
On 6/28/05, Erik Espinoza [EMAIL PROTECTED] wrote:
Nope, sixTel has been acting up big time recently. I've put in
requests for support online. Calling has gone to a message that says
go to the web site.
This really blows, the prices were good but unless they
I was able to raise the volume from inaudible to acceptable by
increasing the RxGain in zapata.conf by 5db. I'd rather not go the
uncomressed wav route, as it will chew up storage in my email system.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
General config might be ok - as long as national and international
prefix are 0 and 00
[interfaces]
; msn=50
; incomingmsn=*
;controller=1
softdtmf=1
accountcode=
context=incomingtest
;echosquelch=1
;echocancel=yes
it was the second one i needed - thank you. I only needed the numbers
in french on one b number and make it that the number could be
dialled from any extension (which is why option a was unsuitable.
thanks again.
On 28/06/05, Arvanitis Kostas [EMAIL PROTECTED] wrote:
On Monday 27 June 2005
To people who have replied to me about Fwd: JE TROUVE
QUE VOUS N'ETES PAS HONETE!,
Have a look at
http://lists.digium.com/pipermail/asterisk-users/2005-June/114066.html
Harry
___
Appel
What's your point?
On 28/06/05, harry gaillac [EMAIL PROTECTED] wrote:
To people who have replied to me about Fwd: JE TROUVE
QUE VOUS N'ETES PAS HONETE!,
Have a look at
http://lists.digium.com/pipermail/asterisk-users/2005-June/114066.html
Harry
Hi Hamish,
Sorry about being a day late... man, these lists are hell to keep up
with.
You could also try xlite - www.xten.com
Cheers,
Zoltan
Hamish Whittal wrote:
Hi Folks,
I am wanting advise on a good soft-phone on Linux. I have looked at
Gnophone but cannot seem to get it to
Is anyone using Asterisk as a Quality Monitoring Platform for random
recording of inbound calls that come into another ACD?
We use Aspect ACDs for inbound call routing and do all live Quality
Monitoring at the moment. I have looked at many Quality Recording
systems that run from $30K to $700K.
yea, i've been getting kinda crappy calls for the past day too
On 6/28/05, Rick Baranowski [EMAIL PROTECTED] wrote:
Looks like the router around them in the Denver area have been having issues
or overloaded. I have been getting choppy calls all day but seems to be
better tonight.
Rick
Hi,
I ve installed recently AAH 1.1
And I was wondering on how to use this conferencing feature ?
I have created extension 200.
and when I try to call 8200, it says that this is not a valid
conference number.
Is there something specific to do ?
Also, when entering MeetMe console,
I cannot see
On Tuesday June 28 2005 4:56 am, Tom Fielding wrote:
Hi all,
Sorry for this elementary question (I'm a newbie).
I'm trying to write an agi script (test.agi) and run it when I call
in. However, I'm getting an error that says application agi isn't
being found. I've put test.agi into agi-bin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Monday, June 27, 2005 5:26 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread
On Monday 27 June 2005 15:46, steve szmidt
Rich,
You must be a LiveVoip crony. Regardless, there is nothing wrong with
my post. Expressing that we should all learn from their mistakes is
hardly an objective. So whatever your issue, I think you can take your
own advise.
Thanks,
Wiley
-Original Message-
From: [EMAIL
Hi group!
Today I tested spandsp with asterisk and started to send a fax, but it dont
work :( Asterisk call and spandsp start sending - here is my console-log:
DIS with final frame tag
In state 10
Start tx document
CFR with final frame tag
In state 4
Start tx page 0
Start tx page 1
RTN with
Never uses AAH, but check for two things
if 8200 is mentioned meetme.conf
and if you have ztdummy initialised ...
~Vamsi
On 6/28/05, Jean-Marc Salsa [EMAIL PROTECTED] wrote:
Hi,
I ve installed recently AAH 1.1
And I was wondering on how to use this conferencing feature ?
I have created
Try _1NXXNXX -- don't forget the underscore.
-Original Message-
From: wassim darwish [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 28, 2005 2:59 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] HOW TO WRITE ROUTE PATTERNS DIALPLAN
in routes pattern i tried
Sorry, I meant I had never used AAH.
On 6/28/05, Vamsi Pottangi [EMAIL PROTECTED] wrote:
Never uses AAH, but check for two things
if 8200 is mentioned meetme.conf
and if you have ztdummy initialised ...
~Vamsi
On 6/28/05, Jean-Marc Salsa [EMAIL PROTECTED] wrote:
Hi,
I ve installed
On 2005-06-28, Rich Adamson [EMAIL PROTECTED] wrote:
It's probably a $2 decision. Just pick one or two and try them.
There are a fair number of people on this list (including myself) that
stay current with multiple itsp's. Every itsp is going to have a problem
now and then, so keeping a
Hello!
Started to use asterisk.
i'm connecting to it with 'asterisk -r -q' command and everytime people are
using it, i see following in my asterisk:
Jun 28 18:07:25 NOTICE[20564]: rtp.c:298 process_rfc3389: RFC3389 support
incomplete. Turn off on client if possible
Jun 28 18:11:12
That's incorrect, both versions represent valid syntax.
To the OP -- show us the contents of test.agi
-Original Message-
From: John Millican [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 28, 2005 8:23 AM
To: Tom Fielding; Asterisk Users Mailing List -
Non-Commercial Discussion
On 2005-06-28, r00t [EMAIL PROTECTED] wrote:
I'll second voipjet for outbound only. While many reported problems to
VoipJet bothers me for two reasons. First, their terms of service are
absolutely insane. Users are specifically forbidden to place calls
regarding medical or financial matters
Also, when entering MeetMe console,
I cannot see anything. Is that allright ?
meaning that if I have not started any conferencing, then, I shall
see
nothing in MeetMe :o)
You need to type the extension number in the box to see the conference
details
I did that ... 8200 ... still ... nothing ... :o)
On 6/28/05, Dean Collins [EMAIL PROTECTED] wrote:
Also, when entering MeetMe console,
I cannot see anything. Is that allright ?
meaning that if I have not started any conferencing, then, I shall
see
nothing in MeetMe :o)
You need to
So far my experience with TOS has been that most of them are pretty odd.
No one wants the liability of a stock trade gone foul or a call to the
doctor that gets disconnected. Essentially, those things in the TOS are
just a CYA. They are un-enforced but should someone decide to attempt
to sue
The 1 m internet connection will be the
limiting factor in your setup, you did not state what type of internet
connection, but given the speed of 1 mbit it must be DSL (or maybe fraction
t/e1).
Is the outbound speed also 1m? Is there
data on the line also? How much? What about voice
Is anyone using Asterisk as a Quality Monitoring Platform for random
recording of inbound calls that come into another ACD?
we have several clients that we are doing something similar with. we are
doing full call logging by tapping the telco lines.
snip
The connection to the Aspect ACD
Ok I have to get a vote of all the people that are going to come to
Cluecon so we order the beer keg's for the developers board room.
Anyone have any preference? (if you haven't registered for ClueCon
now is the time to register!)
Choices... choices... choices... I want Red Bull on tap!
On Tuesday 28 June 2005 09:44, Michael Di Martino wrote:
Their would not be so many newbie questions if their was 1. A fully
indexed searchable archive list and 2. Good solid documentation.
Alright, you've called my bluff.
http://www.mail-archive.com/asterisk-users%40lists.digium.com/
Is that
We are interested in how other people are handling NAT problems. We have
several customers all of which have some sort of firewall/NAT device at
their location. For simplicity sake, all customers' internal networks
are 192.168.*.*.
Our asterisk box is on public IP not blocked by any FW/NAT.
Yes it is DSL and outbound speed is aslo
1Mbit, its a dedicated server and we just use to talk. I look at the web
site which you suggested, but i want to learn how many calls supported practically?
Any information do you have?
Thanks
Erdem HAKI [EMAIL PROTECTED]
From:
you say busy because you hear the busy tone, or you say busy because
you see in the console that the $DIALSTATUS is busy???
post the verbose console output please (asterisk
-r)
best regards
On 6/28/05, vdasilva [EMAIL PROTECTED] wrote:
Hello
On Monday 27 June 2005 20:04, Robert Webb wrote:
I agree with that fact the same questions get posted, but
that problem is compounded by the fact the archives are not
really searchable. If the were as lease some users would search.
The archives need to be fully indexed.
In a Google
P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in
other real world, working, solutions.
Apparently, Jasomi does pretty good SIP/NAT far end traversal
solutions. From what I've read on the list, it's meant to be quite good
- although expensive.
We've been feeling our way along with the NAT stuff (using SIP) as well.
At this point we are fairly small, so the keep-alive packets are not too bad.
What type of user load are you at and what are the specs on your Asterisk box?
I'm concerned we may run into this as well.
We do have the luxury
On 2005-06-28, Wiley Siler [EMAIL PROTECTED] wrote:
So far my experience with TOS has been that most of them are pretty odd.
Not THAT odd :-)
No one wants the liability of a stock trade gone foul or a call to the
doctor that gets disconnected. Essentially, those things in the TOS are
just a
Erdem HAKİ wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Monday, June 27, 2005 8:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RTP session between two end users
Hello list,
I wonder if someone might be able to clear up something for me.
I recently set up asterisk and have now managed to get the MeetMe
application up and running.
When I dial the extension to access the conference/MeetMe application, the
only prompt I hear is:You are currently the
Great points Steve. I think the best we can do is all throw the newbies
a bone ounce in a while. Redirection to the content that is relevant is
enough to get most people on the path. Like you said, the hardest part
is not seeing the trees for the forest.
This is the whole teach a man to fish
Wiley Siler wrote:
What distro and kernel version are you using?
What version of Asterisk?
Thanks,
Wiley
RedHat 9
2.4.20-8smp #1 SMP Tue Dec 28 17:23:01 CST 2004 i686 i686 i386 GNU/Linux
Running CVS-HEAD as of June 27, 2005 around 9PM CST
-Matthew
Hi
I have a Digium TDM400 card with 4 FXO modules
connected tothe extension ports onaPanasonic KXTD816.
I'm using [EMAIL PROTECTED] v1.0, which has
Asterisk 1.07.
There's a problem that Asterisk doesn't detect when
the line is disconnected on the Panasonic. The Panasonic doesn't provide
Hi,
I have an TDM400 4 FXO module setup on my dual Celeron server
running asterisk 1.0.2 and I have had to restart the asterisk process multiple
times lately. I was wondering if anyone else has to restart the asterisk
process after storms roll through their area. Before I restart I
Might be worth asking the owner of voip-info.org if the mailing list
link can go on the left sidebar permanently?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: 28 June 2005 16:26
To: Asterisk Users Mailing List - Non-Commercial
Hello All
I'm using TxFAX and rxFax. this's work when my system connected with PABX. then when i connect my card with PSTN, my system don't work. it's don't send and receive any thing fax document.
Thanks
Please help me___
Asterisk-Users mailing list
Ray Van Dolson wrote:
What type of user load are you at and what are the specs on your Asterisk box?
I'm seeing loads on certain Asterisk threads reach in the upper 70%
periodically. Running a Quad proc P3 500Mhz with RedHat9 on 2.4.20 SMP
kernel.
We do have the luxury that each Sipura
Hi Aaron,
We have 2 products that do what you are looking for. It has the same
features has Nice and Witness including Voice recording, Screen capture,
Evaluation forms and scoring. One of the product is also integrated with
Genesys. The other one is fully based on asterisk.
Let me know, if you
Hello,
I'm trying to figure out why the asterisk service starts fine, but
when i try to connect by typing asterisk -r I get:
Unable to connect to remote asterisk
The service is running and lists under ps -ef as:
asterisk -vvvg -c
any help is appreciated
I cannot get this thing to work. Anyone know
of any tricks?
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Touche`.
Very good points...
But, hey... As long as it works as promised I am OK to let them make
strange TOS!
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Goerzen
Sent: Tuesday, June 28, 2005 8:24 AM
To: asterisk-users@lists.digium.com
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi List
I'm trying to create a conference room using H323 channels.
If i start asterisk normally (service asterisk restart) and connect to
cli using -vvvr options, when a user enters the Conference,
asterisk says You are the only ... and then
I have just installed [EMAIL PROTECTED] version 1.1,I have made a number of successful calls, but the bill cost remains "0".
I have kept everything simple:
Pattern
Comment
Trunks
Connect Fee
Inc. Seconds
Cost per additional minute
DANSAM
0
0
10
Trunk Name
Technology
Peer/Trunk
Interesting article, hopefully this might give some people
here some ideas about cool asterisk apps to develop.
Cheers,
Dean
Voice being heard in CRM
By Barney Beal, News
Editor
28 Jun 2005 | SearchCRM.com
SOUND
need latest zaptel source
_Mobilcomhttp://www.mobilcom.net
- Original Message -
From:
Steve Totaro
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, June 28, 2005 2:53
PM
Subject: [Asterisk-Users] Revision I
Board
Hello All
how to install functions allow called record current call by pressed any key to wave file. for examples.
the caller call to asterisk, then press 123 tone to switch to saler person. then two persons conversation, the saler want to save current call. saler may be press any key as *5 or
Hi Jay,
It's just the standard test script, agi_test.agi in
/var/lib/asterisk/agi-bin (pasted below).
So am I to assume therefore that I don't have to do anything special
during compilation and installation ('make' and 'make install') to
enable agi support?
Thanks,
Tom
[EMAIL PROTECTED]
Please ignore previous email:
I have just installed [EMAIL PROTECTED] version 1.1,I have made a number of successful calls, but the bill cost remains "0".
I have kept everything simple:
Pattern
Comment
Trunks
Connect Fee
Inc. Seconds
Cost per additional minute
44.*
TEST
0
0
10
i could not follow this message from the beginning, but by the subject
i may have an idea of whats going on.
does the command 'show applications' in asterisk console shows up AGI
in its output?
do you have load = res_agi.so in /etc/asterisk/modules.conf ???
best regards
On 6/28/05, Jay Milk
Hi All
I have a Junghanns BRI 4 port installed where only the first channel
of each line is working i.e. channels 1 and 4 work but 2 and 5 don't.
Our config is the same on this box as 15 other similar installations
where all works well. the only error I see is in /var/log/messages:
Jun 28
IAX2/[EMAIL PROTECTED]/996 at 1pm CDT on Thurday the 30th.
If you have any topics that need to be covered please direct them to me.
Thanks,
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”
___
I did a google search on 'voip speed test' - the first
site is very good. Here's the link:
http://www.talkswitch.com/voip/voip_test.php
It will test both your download and upload speeds and
will let you know how many concurrent calls at
different codecs your connection will support.
Try it a
Hilton Williams wrote:
Hi
I have a Digium TDM400 card with 4 FXO modules connected to the extension ports
on a Panasonic KXTD816. I'm using [EMAIL PROTECTED] v1.0, which has Asterisk
1.07.
There's a problem that Asterisk doesn't detect when the line is disconnected on
the Panasonic. The
Hello All
How to detect remote called offhook.
i make a context as below
i created call file. copy to /var/spool/asterisk/outgoing.
Channel: vpb/g0/9050718MaxRetries: 1WaitTime: 10Context: ext-calloutExtension: sPriority: 1
then when i copy to /var/spool/asterisk/outgoing. the asterisk auto call
I tried a Calgary DID with Link2Voip, but they never did get it working
correctly. My primary complaint with their customer service is that it was
basically non-existant. It took 2 weeks before a service guy even
responded to my problem, we fired a few emails back and forth, and then I
Hi all
Correction on my last mail, I found that line 1 both channels work
but on line 2 none work.
I have 2 BRI ISDN lines coming in on port 1 and 2 (4 channels) on a
Junghanns 4 port.
The setup by the Telco on this ISDN is different than our others, they
have 2 lines (4 channels) that are all
i think that the thing that really matters here is wich version of
Asterisk are you using exactly. I dont know wich version the latest
debian package is using, and i dont know wich version from CVS your
friend has compiled.
Also, its needed to show the extensions.conf configuration of your
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