On Wednesday 13 August 2003 03:24 pm, Jerk Face wrote:
/usr/src/usr/include/mysql/errmsg.h
The version of MySQL that I'm running is 3.23.57-1
What distribution are you running? That's a pretty braindead place
to put the mysql header files. You'd think someplace like
/usr/include,
Hi,
First off, a big thanks to Digium (Mark, John, and Martin) for helping
sort out a BellSouth config issue on our PRI. T100P working like a
champ!
Now it's back to tweaking the configuration on our SIP phones (7960s).
The message_uri parameter in the phone's configuration file is working
On Wed, 2003-08-13 at 18:36, Andy Powell wrote:
On 13/08/2003 at 17:46 Dave Cotton wrote:
in the file wcfxo.c the following structure is initialised as below
which would suggest that FCC is wrong for France or pretty well all of
Europe.
errm,
FCC mode is for the US. CTR21 is for
Hello Martin,
Yes, I have span configure in zaptel.conf:
span=1,0,0,esf,b8zs
I dont have a PRI plugged in to the card. Would it be an issue? Reason is I
am current only testing the call
originating from H323 endpoints.
Firewall shouldn't be a issue since the call works fine with ztdummy loaded.
Andy Powell wrote:
FCC mode is for the US. CTR21 is for Europe - you even pasted the info
in your message!
Exactly, the question really is how do you change it?
modprobe wcfxo opermode=1
HTH
Andy
This switch (opermode=1) is redundant with the current X100P cards, as
it changes register
We use rfc2833 for everything and have no trouble. Make sure your 7960
is sending the right indications.
Jeremy McNamara
Jay Sakata wrote:
I have the same problem that Michael describes below does anyone have any recommendations?
Jay
I made a mistake of buying it so that I can have a low-bandwidth
well-tested codec for use on an IAX2 link. Then I've caused Digium lots
of unwanted trouble, because hair stood on the back of my neck after
reading the licensing agreement and seeing the .so library. Let's hope
it gets better
Chee Foong wrote:
I wonder is there a way where I reload asterisk on CLI without
disconnect any call that is currently taken place.
Type help into the console and read.
canopy*CLI help restart gracefully
Usage: restart gracefully
Causes Asterisk to stop accepting new calls and exec()
It can be a bad module. Contact [EMAIL PROTECTED]
regards
Martin
On Wed, 6 Aug 2003, Eduardo Goncalves wrote:
Martin,
With KFLAGS+=-DNO_CALIBRATION uncommented, the module loads fine. Without the
calibration.
But I have no dial-tone on port 4. All the three other ports works
This is starting to sound like a feature request, perhaps by using
the same method that Cisco phones use (comparison using the Via:
header, and re-registering if the Via: header is different than the
known IP address.)
JT
At 11:02 PM -0700 8/12/03, Terence Chan wrote:
Wasim:
Hi! Thanks a
It does, but you have to use IAX2 (or IAX) which is a single-socket,
sanely designed protocol which penetrates any NAT/PAT which does not
explicitly block outbound UDP connections on port 4569 (or 5036 for old
IAX1)
Mark
On Tue, 12 Aug 2003, Dave Cotton wrote:
On Tue, 2003-08-12 at 15:29,
On Wed, 13 Aug 2003, Devon Henderson wrote:
[...]
We have agents who work both from home and from the office.
Some agents are
always in the office, some are always at home, and some
alternate between
the two.
[...]
I guess my big question is: is it possible to have extensions mapped
Only if you have another FXS port and a real modem connected to that and
you bridge the call between FXS and FXO in asterisk.
regards
Martin
On Thu, 14 Aug 2003, Dan wrote:
Hi,
There is any possibility to define a virtual extension on the asterisk box
to act as a local modem?
This is the
I guess my big question is: is it possible to have extensions mapped to
people, not to phones?
Yes, you just need to link the user/extension to a technology/channel
when logged in, and to a bogus value when not so that your dial will
fail quickly and fall through to voicemail. Also you
neither does agentlogin, see
http://www.digium.com/asterisk_handbook/agentlogin_queues.html
remember, you can define members as devices/types (like
agents)/local so you can create some pretty wicked setups G
Brian West wrote:
Nope.. sure doesn't.. You call the AgentLoginCallback extension
Oh my why do that? Customers/Users will have a hard time hearing and
understanding in some cases.
bkw
On Wed, 13 Aug 2003, Stuart Hirst wrote:
Does anyone know if it would be possible to play music on hold in the
background whilst playing IVR prompts. I am hoping that this would have
the
Hey thanks. Much appreciated!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andy Hester
Sent: Wednesday, August 13, 2003 9:23 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] IP phone recommendation
Nathan,
I am using the
On Wed, 2003-08-13 at 15:06, Jerk Face wrote:
I'm trying to compile the cdr_mysql module, but I am receiving error
messages.
I have installed mysql-devel.
Here is the output of make cdr_mysql:
cc -fPIC -I/usr/local/mysql/include -I/usr/include/mysql -c -o
cdr_mysql.o cdr_mysql.c
On Wed, 13 Aug 2003 17:56:46 -0500
Tilghman Lesher [EMAIL PROTECTED] wrote:
The CNG tones are sent by the sending fax machine, not the receiving
fax machine. Those tones are sent from the moment that the fax
machines dials and continues until either a timeout or the receiving
fax machine
Maybe get on IRC and try to debug it with IAX2. SHouldn't be any
different peering as long as your gateway provider supports it.
Mark
On Thu, 14 Aug 2003, Dave Wilson wrote:
mark wrote:
Can you try iax2?
We tried that, but couldnt seem to get the peering to work on IAX2. We being
On Wednesday 13 August 2003 03:06 pm, Jerk Face wrote:
I'm trying to compile the cdr_mysql module, but I am receiving
error messages.
I have installed mysql-devel.
Here is the output of make cdr_mysql:
cc -fPIC -I/usr/local/mysql/include -I/usr/include/mysql -c -o
cdr_mysql.o
It could work if it would be coming over g711 and you'd have
dtmfmode=inband set for that call
regards
Martin
On Thu, 14 Aug 2003, James Golovich wrote:
On Thu, 14 Aug 2003, Eduardo Goncalves wrote:
I'm using G.711alaw.
My extensions.conf:
===
[globals]
Hi Brian,
ATA is in SIP mode, and RFC2833 is used.
Something else to check?
Thanks,
Dan
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 14, 2003 6:16 PM
Subject: Re: [Asterisk-Users] '#' doesn't work for me
Accually it will work
No i don't think so..
- Original Message -
From: George Lin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 14, 2003 9:33 PM
Subject: [Asterisk-Users] CODEC DTMF
Dear all,
I like to know if the DTMF option is related to the codec or not. Can a
SIP
phone with g729
If you add t to you out-going trunk Dial lines:
exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED]||t)
exten = _NXX,2,Congestion
so that you can still use park to park a call or transfer
the phones, You have a problem of not being able to use
# on external IVR systems. Is there any solution
to
On Wed, 2003-08-13 at 09:46, Dave Cotton wrote:
I've had a few problems with my system holding the line after a call has
been made, just now I rebooted and noticed the following in
/var/log/messages
When you say holding the line, do you mean that asterisk still
believes a channel is in use
Hi,
I exactly got the same problem on the Belgian network. I have tried to recompile the
wcfx0 driver with the FCC line
commented and I have created a zone for Belgium in zonedata.c (see below with the
values I know. I'm not sure of
call wait, dial recall and record tone). Everything works fine
On Thu, 2003-08-14 at 12:24, Andy Powell wrote:
Can't find the message in a search.. but below is a msg retreved from my
archive..
this is what Mark sent a little while ago
I have no idea if it actually does anything to the card, but on a modprobe I
do get a msg saying it's using
Hi Martin,
I use ATA-186 (G.711) with two analog phones.
I can transfer using Flash, but nothing happen when press on '#'...
There is something else I have to check?
Thanks,
Dan
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 14,
Chris,
Your project sounds right up our alley! Your price constraints are
quite tight, but we may be able to work something out. I have a few
questions I need clarified before we can provide you with a quote. I'm
assuming we're getting you 4 phones instead of 8.
- Does the $2500-3000 price
I am trying to compile the cdr_mysql module but I am getting errors. I have
MySQL version 4.0.11a installed on my box (Mandrake 9.1).
As far as MySQL packages, I have installed:
MySQL-4.0
MySQL-client
MySQL-devel
MySQL-common
libmysql
I have the latest CVS source for Asterisk.
When I run make
Is this not just a case of a new entry in sip.conf
EXTERNIP = external IP
with the code for the contact header modified to use it (if present). Then the
external firewall is set to forward the rtp and 5060 to * ..
I know many people either have sip aware firewalls (as i do) or their * box
Brian West wrote:
Correct me if i'm wrong but doesn't the cdr modules log the call
duration?
If you look at the last sentence of my post:
Storing stuff using the cdr isn't really an option.
This is because I want to add other things to my call log that CDR
doesn't support (for custom IVR apps
On Wed, 2003-08-13 at 11:13, Emmanuel Bergmans wrote:
In order to test CTR21, I was forced to comment the line in the source file as I did
not find a define or a
zaptel.conf directive. It's really bad but... In my case this change has not solved
the problem (see previous
posting)
Well, I'm
I'm trying to compile the cdr_mysql module, but I am receiving error
messages.
I have installed mysql-devel.
Here is the output of make cdr_mysql:
cc -fPIC -I/usr/local/mysql/include -I/usr/include/mysql -c -o
cdr_mysql.o cdr_mysql.c
cdr_mysql.c:30:26: mysql/errmsg.h: No such file or
I am running Mandrake 9.1, and MySQL 3.23.57-1; and yes, I would think that
/usr/src/usr/include/mysql is not the right place for errmsg.h.
What can I do to get around this?
I changed the cdr_mysql.c file:
#include mysql/errmsg.h
Changed to
#include /usr/src/usr/include/mysql/errmsg.h
But I get
Hello all,
I am sorry to bring the old question to the community. But I cannot find any
answer in the google.
I want to deploy multiple SIPs phone in our office. And we have shutdown the
firewall at our office router(with ip 211.x.x.x). we have deployed the
asterisk with IP 218.x.x.x.
All SIP
In order to test CTR21, I was forced to comment the line in the source file as I did
not find a define or a
zaptel.conf directive. It's really bad but... In my case this change has not solved
the problem (see previous
posting)
in wcfx0.c
[...]
fxo_modes[] =
{
/* { FCC, 0, 0, 2, 0, 0, 0,
For those that are using chan_capi in the US, how do you have your line
provisioned (ordering code)? Are you using CACH EKTS?
thanks!
--Justin
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Errm, no...
does that mean you'll personally check to see if my line is busy or not ;P
will try it now...
Andy
*** REPLY SEPARATOR ***
On 14/08/2003 at 09:58 Martin Pycko wrote:
Did you try BUSYDETECT_MARTIN in asterisk/Makefile ?
regards
Martin
On Thu, 14 Aug 2003, Andy
FCC mode is for the US. CTR21 is for Europe - you even pasted the info
in your message!
Exactly, the question really is how do you change it?
modprobe wcfxo opermode=1
HTH
Andy
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At 08:14 13-8-2003 -0500, you wrote:
Has anyone had the opportunity to use a PingTel phone with Asterisk?
No, I have used the Pingtel softclient though, and it's supposed to be very
similar. Works pretty well, although I seem to remember something about
DTMF modes...
Met vriendelijke groet,
I'm getting the following message when I start Asterisk:
WARNING[1024]: File codec_g729b.c, Line 413 (load_module): Unable to
initialize va stuff: -1
Did I mess up the registration key or is something else wrong?
--Eric
--
BTEL Consulting
850-484-4535 x2111 (Office)
504-595-3916 x2111
Hello
I am a newbie to Asterisk. We have set up Asterisk
on a PC with Redhat 9.0. We have installed H323 openphone on our PC's. We are
wondering what a gatekeeper does. It seems we need one but what I have seen in
this group is that agatekeeper must be installed on another box on the
I have a quick call routing question that I'm sure is simple, but for the
life of me I can't figure out.
For example, someone dials 94162384000 asterisk trys our first call route
(our sip trunk) as per the extension rule below:
exten = _9NX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
However,
Vonage got Cisco to include a password protect the config in the
latest version of the firmware, and as far as I know now all the Vonage
ATAs are forever destined to be used with Vonage and only Vonage.
Cell providers do the same, but they help you unlock the phone after
a set period - one or
This is what I have found on the Cisco web site:
CallerIdMethod (a parameter from the web interface)
Description
This 32-bit parameter specifies the signal format to use for both FXS ports
for generating Caller ID format. Possible values are:
Bits 0-1 (method)-0=Bellcore (FSK), 1=DTMF, values 2
OK. Thanks - I think :-)
I'll go trolling on Ebay, see what comes up. Given that most of my projects
take 6 months or so to get off the ground, I hate to put a bunch of money into
this anyway. So, for $1,000, I can put a 6 x 18 unit in my office play with
it to see if this is a product
Lorenzo -
I've submitted a feature request with this patch
(http://bugs.digium.com/bug_view_page.php?bug_id=052). Your
patch isn't completely descriptive, since I still don't know how you
set the hidecallerid value from within a dialplan. Can you explain a
bit more, please? Have you
Hi,
Have someone tried to use the same trick with the PCPhone application (soft
SIP phone) from iConectHere which can fully support Actiontec's Iinternet
Phone Wizard USB phone interface?
I have tried without any success.
Thanks,
Dan
- Original Message -
From: Martin Pycko [EMAIL
And to answer Wade's question: to limit outbound calls on a
particular path, you'd use a local db set routine. In other words,
every time a call is created to that particular SIP peer, you'd add 1
to the counter, and every time a call was hung up out of that pool,
you'd subtract one.
JT
At
Hi Roy,
always use latest chan_capi. the bug is fixed in 0.2.4a.
today 0.2.4b is online which fixes some issues with sending
dtmf and a small enhancement to capiECT.
capi on!
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:
Chris:
Try not to be so worried about sound card, analog
(FXO/FXS), digital (ISDN, BRI, PRI) and what is available by connecting device.
The channel drivers take care of making the devicesavailable to Asterisk.
In turn Asterisk makes all the features such as voice mail, call parking, and
Beautiful. Thanks!
Jim Friedeck
TC wrote:
Jim
I added a patch that mark got into cvs last night
use
ackcall=no
in agent.conf
-Original Message-
From: Jim Friedeck [EMAIL PROTECTED]
To: [EMAIL PROTECTED] [EMAIL PROTECTED]
Date: August 6, 2003 1:46 PM
Subject: Re:
Hello,
I downloaded the chan_oh323. I experience few
problems:
When I dial from console I get all the object
creation and deletion message, and when a call get connected it gives me the
following output.
Wrong Pitch 1st subfr. ! ! Wrong
Pitch 1st subfr. ! !Wrong Pitch 1st subfr. ! !
Try to set the frames option in section [codecs]
to a reasonable value, say 20 for G711, 2 for G7231,
4 for GSM.
Also, do you get segfaults when you try the same
with just one codec enabled?
Michael.
Sip Rtp wrote:
Hello Michael,
Here is the BackTrace of the program which i forgot
to attach
Installed Asterisk on Redhat 9.0 - and not channeled to PSTN/PLMN
networks (no XP100 or special hardware) yet
When I use * with a softphone (SIP) - Asterisk answers the call but
voicemail or other playbacks are STOTTERING for the first 30 secs
(approx.)It happens more often when I start
Hello Michael,
Here is the information which you asked for.
Please look into it..If you need more info tell.
I am using the following call scenerio..
I am dialing to PBX from openphone by dialing a PSTN
number connected to *
through development kit of digium.
then i press 12 as the extension to
I live in Japan and last Sunday I bought my first X100P to see
if it really works for my H323 application.
How long time it should take to be delivered?
Isamar
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Jan Rychter wrote:
Please try to find a better solution.
The DSP Group owns G.729. There is nothing anyone can do, they have us
by the family jewels.
We use iLBC and found it to be very acceptable in quality and bandwidth
usage and its free.
Jeremy McNamara
http://www.bkw.org/~brian/ata.html
Pay attention to connectmode and audiomode Its how I set it up and it
works.
bkw
On Thu, 14 Aug 2003, Dan wrote:
Hi Brian,
ATA is in SIP mode, and RFC2833 is used.
Something else to check?
Thanks,
Dan
- Original Message -
From: Brian
Can you send a trace from your screen after you turn of the debug in
/etc/asterisk/logger.conf
console = blabla,debug
regards
Martin
On Tue, 5 Aug 2003, Ricardo Villa wrote:
Is it possible to know what application? The extension I'm daling is very
simple:
exten = 1001,1,Dial(SIP/1001,15)
Q: What's the difference between Asterisk and a softswitch
A: About $100,000
Soft switch - Hard to afford!
Regards,
Steve
Bruce Ferrell wrote:
I've been working in the VoIP industry for just a bit over a year
now... Mostly taking care of the underlying systems. I've now reached
the point
well if you ask me, the leastrecent part would work if you reversed the
logic on the metric.
my other last_used mod would do a time_t on that agent the last time it
was 'tried' (ast_request'd) then (i was using arrays) qsort so that (new
agents) '0' would be on top, and the agent that got the
I have an opportunity for a 50 seat call center requiring outbound
dialling, inbound call queuing, agent management, call recording,
call/skill matching, call list management, reporting, IVR, management
call whisper, etc. Are there any * resellers on this list who are
capable of handling a
Hi folks, where can I find the R2 beta code for Asterisk?
Best,
PauloHM
Here is the last mail that I recall seeing on the subject:
From: Steve Underwood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] E1 R2 on Asterisk
Reply-To: [EMAIL PROTECTED]
Date: Fri, 18 Jul 2003
On Thursday 07 August 2003 11:10 am, Steve Meyers wrote:
On Thu, 2003-08-07 at 10:01, Justin Carlson wrote:
unsubscribe
Has anyone ever been on a mailing list where you could unsubscribe
simply by sending a message with unsubscribe in it to the mailing
list? I swear, every list I've been
If I am understanding correctly your setup looks like this..
{Asterisk}--[NAT]--Internet--[NAT]--{X-Lite}
If this is correct then you are going to have major problems getting it to work.. Your
RTP traffic is going to get very confused..
You need to get Asterisk onto a Public IP address..
I
No need for the pri debug span, the problem is the duration of the tones
when using dtmfmode=rfc2833. It is way to short. A lot of IVR's just
don't get enough of the tone to work. The info method still has the correct
duration.
Simple to test just deal another phone and hit keys, you will see
Garry,
yes this is possible although it would end up being quite convoluted.
No, simpler than that...
Voicemail comes into the asterisk machine,
* Calls me at work
Plays message for me to enter PIN for voicemail
Retrieve Voicemail
Hangup.
However, if it got my voicemail at work (due
When connecting an analog phone (Siemens Gigaset) to * via a WX100USB,
the phone displays Out of area first, and then the caller id. The two
displays alternate, making the caller-id hard to see.
Is there any way I can tell the phone to just display the caller id? Out
of area is a flag that gets
Mine has been working well, but the only problem is that it doesn't
support callerid (from the POTS side).
-Original Message-
From: John Schmerold [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 5 August 2003 12:37 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Zhone Zplex 10 units
The x100p does get the CID in France. It is now a question of how to break it down.
I changed callerid.c line 278 to :-
ast_log(LOG_NOTICE, Got this:- %s\n, cid-rawdata);
and the result on August 8 at 10:06 from 0490233081 was:-
File callerid.c, Line 278 (callerid_feed): Got this:-
For smaller systems, you'd have to go a NetJet ISDN BRI card ($150? for
two lines)
Try BT Speedway BRI ISDN, ~20$ on ebay
Peter
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Hi again.
On Mon, 11 Aug 2003, Rainer Jochem wrote:
I've played around a little bit and discovered the following:
with
services_url:
http://xxx.xxx.xxx.xxx/xmlservices/vm/index.php?user=1234amp;pin=1234;
the phone tried to get
GET /xmlservices/vm/index.php?user=1234?pin=1234name=...
Hey,
Have i done something wrong or is there something wrong with latest CVS
and cdr_mysql, cause after checking out latest CVS today, I got warning:
[cdr_mysql.so]WARNING[1074424544]: File loader.c, Line 226
(ast_load_resource): /usr/lib/asterisk/modules/cdr_mysql.so: undefined
symbol:
Martin Pycko wrote:
well should be ok if you cvs update now.
Many Thanks !
Martin
On Wed, 6 Aug 2003, Rhys Hopkins wrote:
Martin Pycko wrote:
You're looking for libncurses-dev and in libpri you can remove -Werror
from libpri/Makefile or cvs update libpri (it should be fixed)
Thanks for
Hello Michael
My extensio.conf are as follows:
I have try it with H323 phone, it works ok all digits detected. Only when
call is coming from pstn cause the problem
Also, the console output when digit is press is:
Invalid extension '1 ' in context...'
There is a space after the 1, I
hi ..
I have an asterisk system with three TDM100P (single port FXO) cards
and 10 Grandstream 100 phones connected to it ..
The TDMx00P cards are FXS cards.. :)
1st question:
when i phone out
or receive a call from one of the SIP phones onto the PSTN, there is
a LOT of local echo in
http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.4b.tar.gz
the downloads dir is browseable, but i probably should update the
website a bit
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:
See answers in-line.
At 4:14 PM -0400 8/7/03, Wade Weppler wrote:
From: Wade Weppler [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Sip Trunk config / Least Cost Routing
Reply-To: [EMAIL PROTECTED]
Date: Thu, 7 Aug 2003 16:14:51 -0400
Ah, good idea! I assume even a global
Instead of using a PCI card is it possible to use an outside
SIP service for CO lines?
Title: RE: [Asterisk-Users] queue / agent documentation
My configuration is with a X100P (incoming) and TDM400P w/ 2 ports (agents) and the calls will distribue just as perscribed with ringall and leastrecent. Those are the only two I have used thus far. CVS was a check out from last night.
Why use an AGI? This seems to be easily done with the dialplan,
unless I'm missing some additional sophistication that you're not
mentioning.
Our local area (Toronto) has some extreme overlapping areacode
problems that require some logic to decipher. I've been able to pull
exchange
Hi,
I am having trouble building and installing libpri and asterisk on my
system. Zaptel seemed to install OK.
I am running SuSE 8.2 ( Linux 2.4.20-4GB )
I have made sure I have the prerequisites ( rpm versions shown below )
rhys2:/usr/src/libpri # uname -a
Linux rhys2 2.4.20-4GB #1 Fri Jul 11
for me it takes between 2 and 3 weeksstandard delivery.
BR,
Dan
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 07, 2003 6:23 PM
Subject: [Asterisk-Users] X100P delivery
I live in Japan and last Sunday I bought my first X100P to see
First of all I would like to thank Mark for getting roundrobin to go
roundrobin. Good job.
Exactly, the whole queue system seems significantly better than it was not
so long ago. Thank very much!
Now we have some options here for leastrecent and fewestcalls strategy. It
needs some work on
Hi Martin,
Together with another list member we try to find a solution now.
We'll keep you in touch if something will be solved.
Thanks,
Dan
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 06, 2003 5:54 PM
Subject: Re:
It doesn't make much sense to me, but it appears Robertson intends to
make money just selling pre-configured phone hardware. The sample
units from Grandstream were $60 a while ago, and $75 MSRP. Doesn't
seem like much markup, so I'm curious to see how this plays out.
I would assume they
Hi,
It's really a problem for new Asterisk users. I am new to Asterisk and do
not know * history, which applications are stable, which are in
development, and who do what? It's really hard for new users to keep the
pace with CVS.
So can you recommend more stable Asterisk versions, which are
I'm a bit interested in an intercom system as well. I'm using asterisk
with analog phones. Is there any way I can do this?
AJ
On Fri, 8 Aug 2003, cwitte wrote:
There was a thread a few months ago that tossed around some ideas for
using a cisco phone for intercom or paging. I don't have any
Look in /usr/src/asterisk/include/asterisk/frame.h and scan down to
where you see all the codecs listed. Then, take that number use it
as a power of 2. In other words, if the number for G729A is 8, then
you need to do 8^2 = 25, so 25 will be the number shown in sip show
channels under the
My dial statement is (for testing purposes):
123,1,Dial(H323/192.168.1.55|20|tT)
When a caller dials extension 123 I can connect and talk without difficulty.
Both the caller and the callee can press # to drop back to asterisk.
The caller can dial an extension and transfer the callee.
When the
Hi,
Does anyone know if there is a way (and sample .conf would be very
helpfull) to start and stop call recording(Monitor) while the call
is in progress??
Maybe by transfering the call to a special extention which will
enable the recording and then connect the call back to the phone..
You can
This is some routine that comes with older versions of MySQL. You need to
find out what happened to it .. maybe they substituted it with somehting
else ...
regards
Martin
On Thu, 14 Aug 2003, Jerk Face wrote:
I updated asterisk this morning cvs update -dA
When I try to run Asterisk (asterisk
I was pondering on this question, and have to agree, splitting mailing list just means
yet another list to join (since there may one day be something relavant) and filter
locally. What might appear to be a good solution is a privately run newsgroup on a
digium server eg news.digium.com the
Several people have asked about integrating asterisk in a company that
currently has a PBX installed. Considering the capital outlay for most
PBXs, depending on the model and age of the system, most companies do
not want to look at replacement.
So, what should one look at in a PBX in order to
On Fri, 2003-08-08 at 01:28, Armand A. Verstappen wrote:
Hi Martin,
On Fri, 2003-08-08 at 00:16, Martin Stubbs wrote:
Unfortunately the present x100p driver code will not decode the callerid for 2
reasons
1) the UK protocol is different to the US system.
As it's French this is
Excuse me for jumping in here. I just subscribed, so I might be asking
something that has already been answered. I tried searching the archive,
but didn't see an answer.
I was trying to compile under Redhat 9. I think I got everything
installed that I need to compile, but I get the following
Exactly, Steven. However, we also want the following logic to work:
Agent 1 (ext. 1234) can login (either via the telephone, or preferably, a
web interface) from either of the below locations and have calls from one or
more queues routed to their phone:
- Any phone in any of our offices
- Any
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