Of possible interest to people having various issues with TDM400 cards,
is that a fix has just been submitted to CVS for the issue where CPU
usage would regularly spike up to 100% with the wctdm driver loaded.
Regards,
Richard
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place a call from lcs to *.
thx in advance...
--- richard Coco <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> i have the same setup too.
>
>
[exten_3008]-[asterisk/TCP_SUPPORT]-[LCS]-[exten_20]
>
> Unfortunately i don't know how to configure the
> dialpl
Hi,
i have the same setup too.
[exten_3008]-[asterisk/TCP_SUPPORT]-[LCS]-[exten_20]
Unfortunately i don't know how to configure the
dialplan in my LCS. Can you please give me a hint
where to configure this.
thx.
--- Jacky <[EMAIL PROTECTED]> wrote:
> LCS 2005 just support SIP TCP or
On Mon Sep 12 10:32:47 CDT 2005 John Hill wrote:
> I have 1 x100p. Caller id works fine with the beta1 release. Cvshead
> releases fail with a combination of checksum and ss_thread errors?
> I'm concerned when beta2 or the 1.2 release comes out it will not
work.
> I have been through the configs
? If so, what are the
tricks?
Can macros have goto labels?
Thanks for any help,
--Richard Cook[EMAIL PROTECTED]T: 705-497-9320
x2010
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Are there any USB type devices(ex: analog modem) that can be used to
connect a local telephone company line to the Asterick PC? I've seen
some cheap cards on Ebay but they use a PCI connection. A USB
connection would help with packaging if there was one.
Thanks in advance.
Richard Davis
[
Hi,
The only thing i know is that you need a netbootserver
using five special files. So, if possible, ask Siemens
for the optipoint 600 netboot upgrade procedure. AFAIK
it is a known problem...
hope it helps...
--- Anthony Cox <[EMAIL PROTECTED]> wrote:
> Not strictly a problem with Asterisk bu
Matt wrote:
Am I correct in thinking that at this time the CVS-HEAD supports
Jitter Buffer for SIP on Asterisk?
No, but attached to "issue" 3854 you will find patches you may be able
to apply to the current CVS-Head to acheive this.
Regards
nterrupt :00:02.0[A] -> GSI 22 (level, low) -> IRQ 185
ACPI: PCI Interrupt Link [APCL] enabled at IRQ 21
ACPI: PCI interrupt :00:02.1[B] -> GSI 21 (level, low) -> IRQ 193
ACPI: PCI Interrupt Link [APCJ] enabled at IRQ 20
ACPI: PCI interrupt 0000:00:04
David,
Yes we got them and they caused huge problems. The echo training would
cause the line to mute and you would hear something like a dtmf tone briefly
and then you would be connected and talking again. This might happen once
or 50 times during a call. I spoke to Digium and they say there ma
Michiel van Baak wrote:
Is there any other solution like this out there that works
with asterisk ?
If you find something, I would be interested in the outcome.
I want something for the house here, at the moment I just have 2 analog dect
bases plugged into the same line, but you cant roam betw
am
> getting the following messages.
Hi,
When I got cause 88, this was caused by sending the wrong Bearer Capability.
Either Speech, or 3.1khzAudio .
HTH
Richard
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Checked modules and wct4xxp and zaptel are loaded. (if you check the makefile
wct2xxp is an alias for wct4xxp)
Then did a lspci and it is not sharing any IRQs
Now I am doing a zttest and it hangs on " Opened pseudo zap interface,
measuring accuracy..."
I have installed a new TE205P in my asterisk server. When I reboot the
box the error " ZT_SPANCONFIG failed on span 1: No such device or
address (6)"
When I "modprobe wct2xxp" and run "ztcfg -vvv" loads everything
correctly. So it seems that it is not loading wct2xxp at boot.
I then added the f
I don't know if my conf files are
screwed-up or if ooh323c code isn't working.
/var/log/asterisk/h323.log should give you some good information.
Regards,
Richard
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onsive and they are
thieves.
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ll to configure Netfilter, and wouldn't want to be causing
latency with it.
Richard.
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I am having a problem with my CDR web page (AMP). There is a banner on
the page saying "YOu MUST ACCESS THE CDR THROUGHT THE ASTERISK
MANAGEMENT PORTAL!" and it will not show any calls just "No calls in
your selection." I have checked the database and calls are being
recorded in the database.
I
I am taking it for granted that you have configured *
correctly for E-Mail notification.
If you are not receiving e-mail then I would look at your
sendmail configuration. The easiest way to configure it is to use the
Smarthost=xx in sendmail conf to point to your SMTP gateway
_
Hi,
I'm trying to figure out why my Sweex CA22 is not detected by wcfxo on my
SuSE Pro 9.3 running on a VIA EPIA SP motherboard. The card should work as its
an Intel 82537EP based card.
When Booting the IRQ assignments for the board indicate:
Bus Dev Func Vendor Dev Class
020 0
ztdummy is working and Meet
Me is working internally (until 2 min ago when I made the wrong change but I
will back track and make that right again).
Only the external users
coming in over a trunk can not access. IVR picks up but only allows ext
dialing.
Info:
Asterisk CVS-v1-0-06
ztdummy is working and MeetMe is working internally (until 2 min ago
when I made the wrong change but I will back track and make that right
again).
Only the external users coming in over a trunk can not access. IVR
picks up but only allows ext dialing.
Info:
Asterisk CVS-v1-0-06
Meet me .conf (s
I am trying to configure MeetMe so that external callers can
enter the conference rooms after an IVR menu. I have created Conf rooms for
all internal Ext’s with a prefix of 8. When I call into the system from
my vonage trunck the IVR picks up but will not let me dial a conf room. It
t
Ed,
Check this out:
http://turnkey-solution.com/asterisk-sphinx.html
That got me up in running in no time.
-Rick
-Original Message-
From: Ed Greenberg [mailto:edg at greenberg.org]
Sent: Friday, 8 July 2005 9:32 AM
To: Dean Collins; Asterisk Users Mailing List - Non-Commercial
Discu
Better read up on why a sip phone should register with asterisk. Do a 'sip show
peers' and that will be the list of phones that can "receive" calls.
---
I've double checked this. Everything is logging in fine, because I can
make calls, check my voicemail, everything exce
Is it just me that sees the post above as spam?
If we (tinw) even consider buying stuph from spammers, then we are
encouraging them in their sociopathic behavior, and as a consequence they
will do more spamming.
What is the consensus here?
It is a product announcement for a new product that is
[root at pbx sbin]# ps ax | grep asterisk
3371 ?S 0:00 /bin/sh /usr/sbin/safe_asterisk
3417 ?S 0:00 asterisk -vvvg -c
6846 ?S 0:00 asterisk -vvvg -c
6848 ?S 0:00 asterisk -vvvg -c
6849 ?S 0:00 asterisk -vvvg -c
6850 ?S
so I know I will work with Linux.
Thanks in advance.
Richard Davis
[EMAIL PROTECTED]
Business Systems Connection Inc.
9357 Watson Industrial Park
St Louis MO 63126
local: 314-918-7526
Toll Free: 877-271-9484
cell: 314-602-1326
Fax: 314-918-7527
www.bizsysco
If it does materialize, im up for 3 or 4 of them at that price.
Huddleston, Robert wrote:
Well poo - if I can use that word I'm one of those poor family guys who
loves to buy hardware on the cheap =)
smime.p7s
Description: S/MIME Cryptographic Signature
_
Sandy Thomson wrote:
Has anyone had any success with this card?
Thank you.
I am looking for a source for the clones in NZ - getting the real deal here isnt
an option (killer shipping) and at the moment I am just having a play with
asterisk and have given up on the internet linejacks I rescued
Hello Carlos,
Thank you for the reply.
It does appear to be the actual viewer that squishes the image, not SpanDSP.
I tried a different viewer on the workstation and the fax appears correct.
I like your suggestion to convert it to PDF, thank you. :)
--
Richard Cook
[EMAIL PROTECTED]
T: 705-497
Hello,
Has anyone had issues with faxes showing up
squished in the TIFF file?
Any ideas what could be causing
it?
--
Richard Cook
[EMAIL PROTECTED]
T: 705-497-9320 ext 2010
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Sorry about the lengthy post, I've searched high and lo for
information on how to do this but now I need your help...
Brief intro on problem and requirements ===
I'm hoping to use Asterisk in a Home environment where I'd like to
replace the current non-PC Answering Machine, an
I am about to add h323 to my system and although I have found
information on the Wiki, comparing the asterisk implementation to oh323,
I have not found anything about the new ooh323, which is included in the
addons.
Can anyone please compare this to the other two?
Thanks,
Richard
Ronald Wiplinger wrote:
app_addon_sql_mysql.so
app_intercom.so
app_saycountpl.so
cdr_addon_mysql.so
format_mp3.so
res_config_mysql.so
WARNING WARNING WARNING
I cannot remember that I have seen that before.
you must have checkout'd asterisk-addons and compiled it at some po
> I already have OH323 support in Asterisk, but have
> no clue how to
> configure the HiPath.
hi...
oh323 is the only thing you need for Astersik. For the
HiPath it depends on which version you have.
FOR HiPath4000 V1.0
---
for version 1.0 you need a HG3550 V1.1 Board.
-Configure
(TP'n to follow flow)
why is it not doing the 'uname -r' ? (meaning, what does it
matter, it would go after whatever was the current loaded kernel
(like it used too!))
Andrew Latham wrote:
check you /lib/modules/ for other kernel directories.
On 6/2/05, Pudenz, Duane <[EMAIL PROTECTED]
Hi all,
i am looking for informations about large installation
with Asterisk (~3000 users). Has anybody experience
with such a setup. Any comments, suggestions or
problems would be appreciated.
thx in advance...
__
Do You Yahoo!?
Tired of spam?
David Phelan wrote:
Anytime I receive a landline to anything over here in AUS, it comes up as
Overseas
I asked telecom why, and they said that the standard used doesnt support longer
then 3+7 digits, so international numbers may not fit. I would still like to be
able to send an NZ number with
Rod Bacon wrote:
We have antiquated caller ID schemes here in Australia. We barely
support numbers from other local carriers, let alone OS ones. Certainly
no names either.
When dialing out thru voipjet, I can put anything I like and it will come thru
to my mobiles in New Zealand just fine (on
You can get the report at
http://www.cisco.com/warp/public/707/cisco-sn-20050524-dns.shtml
Richard
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AddQueueMember
. remove a member from db via RemoveQueueMember
. callout to members from dynamic reading from db
It should have its own mysql connection, but ast_data is preferred. We
will spend an additional $50 for using ast_data.
Please reply me offline.
Thanks,
Richard
--- [EMAIL PROTECTED] wrote:
> I'm trying to setup Asterisk trunk to Siemens HiPath
> 4000 V2.01
i suppose you mean version 2.0 ;-)
> What would be the best way to do so? I am a bit
> confused because as far
> as I've understand this PBX doesn't support H323,
> but I saw somewhere
> someone who
Hi,
I am using ACD, i.e. application Queue(). Is there a way to use mysql
for the configuration file?
Thanks,
Richard
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28 PM, Richard Lyman wrote:
Waldo Rubinstein wrote:
Hi guys,
I'm testing the sending of a URL to an XLite softphone when a call
is in queue. See the output of the CLI below:
-- Executing Queue("Zap/69-1", "q_sample|tT|http://
www.google.com/") in new stack
-- Started
Waldo Rubinstein wrote:
Hi guys,
I'm testing the sending of a URL to an XLite softphone when a call is
in queue. See the output of the CLI below:
-- Executing Queue("Zap/69-1", "q_sample|tT|http://
www.google.com/") in new stack
-- Started music on hold, class 'default', on Zap/69-1
They are the only company that I have been ordering
from.
I've probably placed about 10 orders of items ranging from
1 to 10 pieces per order. Never ever had a problem of any kind. I
even speak to them on AIM and place orders directly.
Excellent recommendation.
--
Richard
G.Marshall wrote:
The rtp audio is going phone to phone, not via asterisk. This is one of
the reasons I am trying to set up SER with Asterisk.
I thought that canreinvite=no was supposed to force the audio to go via asterisk?
smime.p7s
Description: S/MIME Cryptographic Signature
I have an asterisk server behind NAT - no audio on the test external calls I
have tried making so far.
Read http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - No solution
evident from there, sounds like I have case 9. I would have thought that all I
would have to do is port foward and h
d
> Asterisk to these Cisco routers before?
Just in case you don't know, AS5350 supports SIP *and* H323 after IOS version
12.3 (maybe a little earlier).
It allows you to use both at the same time, without needing to set it up for
one system specifically.
Haven't tried it with A
best candidates. I read it many times
and my impression is that ast_data seems to be better. But can't find a
patch for latest cvs stable.
Any suggestion?
Thanks,
Richard
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Matt wrote:
Greetings,
Does anyone know if there is a cost effective way to interface an
older AT&T Spirit system into Asterisk.
I'm only interested in A) being able to offer voicemail and B)
possibly an AAT to callers.
I've thought about just stringing the FXO cards into the line1/2 slots
that go
e something else I must do to make these lines
operational?
Thank you very much for you help.
Richard
Yahoo! Mail
Stay connected, organized, and protected. Take the tour:
http://tour.mail.yahoo.com/mailtour.html
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Matt Darnell wrote:
These phones are mentioned in the Sip 1.5 manuals, anyone know what
the differences are?
Where are you getting SIP 1.5 from?
When I log into the Polycom download area, all I can find is 1.4.1.
Regards,
Richard
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route calls to the group. It would behave the same
way where it can choose the next available SIP entry from the
group.
Is this possible, as is?
--
Richard Cook
[EMAIL PROTECTED]
T: 705-223-2000 ext
2010
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Paul Hales wrote:
It now works - but only in the latest (1.5+) firmware releases.
Where are the 1.5 releases? I see only 1.4.1 on all the Polycom sites.
Regards,
Richard
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Duane Cox wrote:
Do you get 2-way audio that sometimes drops off to 1-way audio then picks
back up as 2-way? (Thats what I see)
Not sure if my problem is a lost packet issue as I am sending IAX off net.
Duane Cox
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Friday, April 29, 2
Guy Boehm wrote:
wau thank you it works!! but,
first it says that e loop is detected,
and secondary what must I do to hand over the new working channel to
my x-lite to use it???
DENGENS
Richard Lyman <[EMAIL PROTECTED]> wrote:
Guy Boehm wrote:
> fputs($socket,
ot;2"
tcpIpApp.sntp.daylightSavings.start.dayOfWeek="1"
tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth="0"
tcpIpApp.sntp.daylightSavings.stop.month="10"
tcpIpApp.sntp.daylightSavings.stop.date="1"
tcpIpApp.sntp.daylightSavings.stop.time="2"
tcpIpApp.sntp.day
st is installed.
./fxstest /dev/zap/1 regdump
will show you the contents of all the registers on Zap 1. If the
majority of them show the value "ff", contact Digium support.
I had modules marked "Rev C" that did this replaced with "X100B RevB"
ones and have not had a
Guy Boehm wrote:
fputs($socket, "Channel: 6159bfb47b9\r\n\r\n");
Response: Error
Message: Invalid channel
the Channel: var needs to be in the form of type/dev/numbertocall
like Channel: IAX2/user:[EMAIL PROTECTED]/14085551212
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et thevoice to change the prompts
> when you are at the first message just press previous (4) & it will do the
> last message
> and then do 4 again to previous again etc
Not exactly the same thing. My request is for changing the order from
playing last to first.
Richard
_
option for each mailbox.
. open source and we will release the code to public
. I'd expect there will be code modification in apps/app_voicemail.c. It
should be based on the latest cvs stable version.
Thanks,
Richard
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http://www.qwest.com/largebusiness/products/voice/callingcards/lb_dial_guide.html
based on that info, i'd say you are about to have a very crappy day.
sorry to reply to my own post, forgot to suggest trying to send calls
over another network.
http://www.thedigest.com/faq/picodes.html
bill black wrote:
Anyone have any ideas here?
We are using 8 channels of E&M Wink with a T100P for outgoing LD and
incoming tollfree numbers and are apparently connected to a Nortel
DMS-250 at the CO. We are receiving ANI & DNIS just fine and can
dial-out domestically with DTMF but have two is
just wanted to let those out there having a similar issue know that ...
envir: normal phone -> chanbank -> asterisk -> iax2 ->pstn ->normal phone
the chanbank side could hear the pstn side, but not vice-versa (this
happend everytime), and would happen with both ulaw or gsm codec's.
seems there
not
> the best PERL scripter in the world but it works.
>
> - Original Message -
> From: "Richard" <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> ; "'Asterisk Developers Mailing
ce must use PHP with mysql as a backend database
. last but not the least, a easily readable and maintainable code is
expected. If you can please send some sample code you wrote before, that
would be really appreciated.
Thanks,
Richard
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Have asterisk installed, and working with my 2 quicknet phonejacks and 1
linejack cards.
I cant seem to get my way thru getting the linejack to answer, and give some
choices to the callers. I cant get the phonejacks to work when I change there
entry to mode=dialtone in the phone.conf, thats all
ssible.
thx in advance...Franz Knipp <[EMAIL PROTECTED]> wrote:
Dear Richard,On Fri, 15 Apr 2005, [EMAIL PROTECTED] wrote:> The latest firmware for optipoint420 advance SIP seems to be version> 4.0.22A, released for HiPath8000.thanks for this information. I've contacted my custome
*snipped
I'm not saying its a bad idea, but some information about what they're
hoping to gain, the type of clientelle they have, and how much they're
willing to spend (i.e. would they see a benefit to tying this into
wiring ethernet to all the rooms for guest use and giving higher-end
business cli
Franz Knipp <[EMAIL PROTECTED]> wrote:
Hi,<
The latest firmware for optipoint420 advance SIP seems to be version 4.0.22A, released for HiPath8000. Unfortunately on the Siemens page the only SIP image that can be downloaded is for OptiPoint400 (www.hipath.de then ->download -> software/version 2.3
lenz wrote:
Hello,
you have to enter "/var/log-xcast/queue_log_live" as the file and "DPS"
as the queue (select it from the drop-down box) for the demo to find
actual data to process. In a real-world environment, you would preset
this information in order to be meaningful for your installatio
Rich Adamson wrote:
I've had a question related to this: what's the deal with frame
slippage on the Digium TDM analog cards? What would cause this? How
can one correct for this? I've recently seen a bad buzz every 6
seconds or so, heard by callers when calls are bridged with my TDM card
anal
I got recently.
Thanks,
Richard
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[EMAIL PROTECTED] wrote:
Matteo,
I don't know much about DIgium, but I am comparing the distribution policy with
what exists elsewhere in the market and other sectors.
Digium do sell online and so many other of their resellers do. The important
point is that they don't sell lower cost than their re
erver.
Cheers
Richard
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erver.
Cheers
Richard
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Thanks - I'll look at the other thread.
On Apr 5, 2005 3:12 AM, Cirelle Internet Products <[EMAIL PROTECTED]> wrote:
> Richard J. Sears wrote:
>
> >Hey Everyone -
> >
> >I am having a problem that is keeping me awake at night.ok, so maybe
> >no
Hi,
my setup
[pbx]---[oh323]--[asterisk]
calling from the pbx into the voicemail gives following output in the console
-- Executing VoiceMailMain("OH323/R1909", "") in new stackApr 5 19:05:46 DEBUG[11862037]: res_adsi.c:212 __adsi_transmit_messages: No ADSI CPE detected (0) -- Playing 'v
sterisk-Users@lists.digium.com
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**
Richard J. Sears
Vice President
American Internet Services
e_request:
Registration from 'GSynn ' failed for
'5.63.198.220'
Apr 4 17:36:48 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
from 'GSynn ' failed for '5.63.198.220'
Apr 4 17:36:52 NOTICE[27276]: chan_sip.c:7681 handle_request: Registration
fro
Steve Edwards wrote:
How can I configure "queue()" so that it does not hang up if the caller
presses "*" to exit the queue?
I want to continue the call so the caller can choose other services.
allow the agent to be able to transfer, then create an exten in
that context that does what you want. s
>
> Richard wrote:
>
> >A debug on the pri shows,
> >Ext: 1 Progress Description: Call is not end-to-end ISDN; further call
> >progress information may be available inband. (1) ]
> >
> >So maybe the inband information is not detected by *?
> >
>
g volts on FXS to 50V peak.
boostringer=1
Boosts ringing volts on FXS to 89V peak.
Regards,
Richard
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ig.
> There are subtle differences in packets. I would check the configuration
> on
> your carrier side and * side.
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Richard
> Sent: Saturday, April 02, 2005 1:2
mpare the results.
I am not sure if it is * or just my * configuration.
Your help is highly appreciated. I am really stuck here.
Thanks,
Richard
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;, contact Digium support.
I had modules marked "Rev C" that did this replaced with "X100B RevB"
ones and have not had any trouble since.
Regards,
Richard
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When connected directly to my
incoming lines they ring normally.
Normally I'd assume this was a power problem but at 0.1 REN?? Any other
ideas I can try?
Try adding the module parameter boostringer=1 when loading the wctdm
driver. This raises the ringing volts to 89V pea
Anyway, you should have this as your
> first line in the
> script.
>
> #!/usr/bin/perl
> ___
>
I had #!/usr/bin/perl5 -w
I changed it to #!/usr/bin/perl and now it works.
Thanks for the help
__
Do
Thanks very much for the suggestions. I've
implemented them, but the main problem seems to be
that the program send_clid.agi is not executing
despite the cli> saying that it is. If you have other
ideas let me know.
Thanks again,
Richard
--- Jean-Michel Hiver <[EMAIL PROTEC
>cli seems to indicate it worked:
Launched agi script
/var/lib/asterisk/agi-bin/send_clid.agi
AGI script send_clid.agi completed, returning 0
however I see no output from wall and if I do a cat
call_id_test it's empty. call_id_test has permission
set to
>
> Do you have the Adit600 configured correctly? It's
> not stuck in a test mode
> or anything?
>
I have no idea if it's configured correctly. We just
kind of hooked it up when the install was done a
couple months ago.
> -A.
> ___
> Asterisk-Users
This goes on continuously and no phones are ringing.
I am using a digium T1 card and ADIT 600.
Does anyone know what this means and if I should be
concerned about it?
Thanks,
Richard
__
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u get
asterisk to pass that into to your log file. That is
in essence the part I'm haveing the most difficulty
with.
Thanks,
Richard
__
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htt
Michiel,
Thanks very much for the resonse. I am confused
however by "fopen("/var/log/asterisk/my_agi.log""
my * system has not such log file only the Master.cvs
which only seems to log a call one its teminated?
--- Michiel van Baak <[EMAIL PROTECTED]> wrote:
> On
database.
>
> It's not really that hard to make.
DO you happen to rember the name of the agi command
that thansfers the record into the table? Or do you
know where I can find some sample sripts to look at?
Thanks,
RIchard
__
Do you Yah
someone tell me how
to make it pass this info to my database server?
Any suggestions would be greatly appreciated.
Thanks,
Richard
__
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http://smallbusiness.yahoo.com/resources
in voicemail.conf and
did you (maybe) compile asterisk to use asterisk_vm mysql db instead of
the voicemail.conf..?
On Fri, 25 Mar 2005 04:55:48 -0500
"Andy Stewart" <[EMAIL PROTECTED]> wrote:
> Richard,
>
> Yep, got that config'd in there:
>
> 1001 => 1
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