and nothing is being recorded. Has anyone ever seen this before? Does the Master.csv fill up? It's current size is 8591056 but there are no other files in the directory ( such as Master.1 or .2 ) as one might expect with traditional logging.Does anyone know what could be going on?Richard
Evnin'
Already asked that a while ago (o;
Has someone an explanation why a 7970G running
SIP firmware 8.0.2 can't correctly register
with asterisk 1.2.11?
It registers quickly but asterisk marks it
right after registration as unreachable:
-- Registered SIP '1002' at 62.x.x.x port 5060
It's excellent home phone. I wouldn't use it in a business environment. No
hold, no one-touch voicemail. However, it works great!
/R
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Monday, September 25, 2006 10:25 AM
To:
Trying again
Has anyone an explanation why this error happens?
Only hear my echo and not the other side anymore...
and the other side can't hear me...
Version asterisk 1.2.9
-- Executing Macro(SIP/1001-9c43, stdexten|1010|SIP/1010) in
new stack
-- Executing Dial(SIP/1001-9c43,
Evnin'
Has someone experienced the same with the FreePBX frontend?
After changing a SIP extension and pressing the red
bar on top in the browser I only see on the CLI:
sip*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Form Hold
Last Message
62.x.x.x
Tzafrir Cohen wrote:
On Tue, Sep 19, 2006 at 09:58:45PM -0700, mitcheloc wrote:
You are incorrect. The GUI you are referring to is the framework I already
mentioned. The webpages are static html javascript (AJAX functionality).
Asterisk has a simple built in HTTP server in trunk now which
Iam experiencing the same
problem.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: Monday, September 18, 2006 10:03
AMTo: asterisk-users@lists.digium.comSubject:
[asterisk-users] Chanspy crashing the server, again
I upgraded to 1.2.12.1 - the
Richard,[EMAIL PROTECTED]
Is there something wrong with my config?
thx in advance
__
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Evnin...
Googled around for this strange error meesage with no
helpful results at all...
Does somebody has any idea what this means?
Forcing Marker bit, because SSRC has changed
At the same time I only get inbound audio but other
side can't hear me...sometimes I just hear my echo
and
Jan Fousek wrote:
Hi all,
is there any possibility of setting the global variables from outside of
asterisk?
Like manager api or something like that.
Thanks a lot
not sure about current svn trunk,
but in the past you could set a channel var with
action: SetVar
channel: Zap/49-1
Hi all,
we plan to install several IAX softphones.
http://www.voip-info.org/wiki-Asterisk+IAX+clients
lists a lot of IAX phones for Windows and Linux. Which
one would you recommand? We will install IAX client on
Linux and Windows.
thx richard
Hi Jason
loadInformation6 model=IP Phone 7970SIP70.8-0-4SR1S/loadInformation6
1. Stick with the 8.0.2 SIP image as it works best with asterisk...
at least for me (o;
- Here are TFTP server logs to illustrate that I'm using the correct
case'd XmlDefault.cnf.xml file:
Sep 10 21:57:55
hello
If I want to use asterisk to hookup to a SIP account
I just use the register line in sip.conf with the
extension number at the end...
But how about if I want to use a SIP trunk from a
provider which gives me 10 DID numbers with the same account?
thanx in advance
rick
And now for something completely different (o;
Is there a way out of a problem when registering
2 times with different account with same host?
I've setup 2 seperate peers using seperate
context in sip.conf...but as soon I change one
extension in one context it influences the other
as well and
MF wrote:
Has anyone got a clue about this?I need to know which operator to
send a message to, prior to the queue command ringing him, (just
after he is assigned)
Anyone knows if I can get to know the operator ACD choosed to send
the call by using Realtime Queue, or maybe via the
Hello (o;
Ist there a way to remove the trailing @domain from
the displayed caller id on the Cisco 7970G?
No problem dialing a number from the missed call
directory with the domain attached...just looks
weird (o;
cheers
rick
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Hi Dean
Dean Collins schrieb:
I don’t know if I’m mistaken or not but I noticed in a iax2 show peers
command that it is showing my iax2 connections as netmask 255.255.255.255
/32 are hosts addresses...which is correct.
All of my lan traffic is supposed to be running on 255.255.255.0
This
Tomislav Parčina schrieb:
According to this thread
http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=990forum=3
Cisco 7970 (SIP 8.0.2) sends wrong request to http server and that is why Cisco
7970 IP Phone doesn't show phone directory or services. It seams there is the
same problem
Tomislav Parčina schrieb:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
My 7970G running 8.0.2 SIP firmware works perfectly with
the Open XML 79xx directory frontend...
I have never tried Open XML 79xx, although I have hear of him.
http://www.asteriskpbx.de/index.php?open79xx
?
thanx in advance
rick
Richard Klingler schrieb:
Evenin' (o;
Following strange problem:
7970G SIP phone - asterisk - SIP provider
In sip.conf I register to my SIP provider to receive
calls from them...but as soon the numer rings I
see as CallerID the configured outbound number
from my SIP
Tomislav Parčina schrieb:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Where did you find 8.0.3 SIP image?
Cisco website...
I didn't noticed 8.0.3 SIP firmware there...
Also on their ftp:
-rwxrwxr-x1 518 201 8136838 Mar 6 2006
cmterm-7970_7971-sip.8-0-2-0.cop
EHLO (o;
Anyone succeeded with hooking up a Matra 6501 PBX to * ?
cheers
rick
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Evenin' (o;
Following strange problem:
7970G SIP phone - asterisk - SIP provider
In sip.conf I register to my SIP provider to receive
calls from them...but as soon the numer rings I
see as CallerID the configured outbound number
from my SIP account and not who is actually calling...
So I
Morning (o;
What would give me less headache for integrating a Nortel PBX
to VoIP?
a) Hook up with a Cisco which handles the SIP stuff
and E1 to telco failover?
b) Hook it up to an asterisk box instead?
If I would go with plan (b)...is there an option I can
sort of pipe through the E1
Dave Fullerton wrote:
I just verified it here as well. Running Asterisk 1.2.11 and two polycom
I'll throw in a me too here, with the addition that it also occurs
with canreinvite=no.
Regards,
Richard
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Tomislav Parčina schrieb:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Does the 8.0.3 image has the same flaws as 8.0.4?
Where did you find 8.0.3 SIP image?
Cisco website...
Wasn't even able to register with * at all since
most configuration examples from voip-info.org
Does the 8.0.3 image has the same flaws as 8.0.4?
Wasn't even able to register with * at all since
most configuration examples from voip-info.org wouldn't
work...
Do you have any example config for me to try with SIP
image on 7970G?
Only tried 8.0.3 on my 7970G and had to switch to SCCP
Is there a way to find out if a channel is currently being
recorded/monitored via the Asterisk Manager API.
Currently, if I issue a Action: Status, it lists all channels as
unmonitored, regardless if they're being recorded or not.
(In my setup, I'm not doing automatic monitoring, I have a
Jeremy McNamara wrote:
Douglas Garstang wrote:
Oh, and I see nufone caters to residential. We only cater to business
customers, who's needs are a lot more demanding.
Apparently you haven't actually gone to our website which, since you
brought it up, will be re-launched on September 5th, 2006
Hi,
i have xlite too and it works without any problems.
ps: what about ekiga? (www.ekiga.org)
rich
--- Joao Pereira [EMAIL PROTECTED] wrote:
Hello to all
can someone recommend me a nice SIP client with
video for windows??
I tried X-Lite 3.0 but it's a lousy piece of
software.
Thanks this was exactly what I was looking for.
Thanks
Richard
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I just bought a grand stream 2000. It appears that it will not dial any
number with a leading * (*70,*71)
So I can not dial any of my Apps in *
Can anyone point me in the right direction?
Thanks,
Rich
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Here is the software version:
Program-- 1.1.0.16Bootloader-- 1.1.0.1
When I pick up the line and dial *70 it just disappears and never dials.
If I enable early dial it does dial *70 but then it breaks my outbound
routes.
Thanks
Rich
I just bought a grand stream 2000. It appears that
Tom,
Disabling the features worked. Thanks.
Richard
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Hi all,
i have following setup
[]--[asterisk]--[oh323]--[HiPath]--[8000]
is my voicemail access
exten = ,1,Answer()
exten = ,2,VoiceMailMain()
8000 is an Optiset phone registered on the HiPath.
When 8000 calls i have no voice (depends on the
setting of FastStart). When
Can anyone tell me how to configure the grandstream gxp-2000 for 4 line
apearances. I have the the sample conf from the website and the phone
is getting its config from my TFTP server. But it does not have any info
for the other line apearance butons
The real thing that would help is a complete
I agree with a bad phone or config problem. I use Polycom extensively
and have had very few problems. Can you give us more info on the config
so we can try and help?
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Stefan Reuter wrote:
Johannes Zweng wrote:
Although I can associate every incoming event to a specific channel on
Asterisk (because of the Uniqueid field) I see no possibility to identify
without doubts which channels were created as a result of my Originate
action.
add an ActionId
Tony Mountifield wrote:
*snipped
Comparing with 1.2, I see there were originally two calls to manager_event(),
one for OriginateFailure and another for OriginateSuccess.
They have now been combined into one, with a conditional event name,
which may have given rise to the mistaken impression
*snipped
Patrick, yes, this is a literal portion. I have no reason to believe that
spsaces between the priority, and the command cause problems, so I haven't
tried that yet. Just trying to make the horrible assembler-like Asterisk
dialplan language more readable.
*snipped
this doesn't
antonio wrote:
I have a problem: when i make i call from a device h323 to sip, i have no
cdr, and i don't see cdr variables for the channnel ooh323.
Anyone can help me ??
Thanx
On my system, this lives in /var/log/asterisk/cdr-csv/ast_h323.csv.
Regards,
Richard
I am thinking of using this machine to run asterisk. Has anyone had any
experience with this machine?
Thanks for any info.
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Bill Schaffer wrote:
I don't know the Cicso equipment, but an educated guess from the
configuration info you gave tells me that you need to use MFC R2
signalling.
Read about it here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+MFC+R2
And pay heed to the big disclaimer at the front
I got most of it working. The sendDtmf() does not seem to work. However I can register and make calls using the call(...) method.I cannot compile the sources (native) so I just got the jar files from http://www.hem.za.org/jiaxclient/and played around with the java code to fit with my app.
I think you have to set where to get the libraries (jiaxc*.jar files).Setup a webserver somewhere and put the jar files there. Then in your code before initialize client.setCodeBase("your URL to the jar files"); HTH, richard Enrique Sanchez [EMAIL PROTECTED] wrote:
For those enquiring last week about ooh323 not compiling with the svn
version of asterisk, the module loader changes have just been checked
into the svn version of asterisk-addons and so should now work with svn
asterisk.
Have not yet tested this.
Regards,
Richard
Martin Joseph wrote:
Do you just mean the tar balls of 1.2.9 and latest addon?
Yes. I believe the svn addons package will be updated soon.
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to reproduce the issue after
that type:
START-HISTA:RTYPE=SEARCHB,STIME=2006-06-27/09:00,ETIME=2006-06-27/09:30;
adjust the start time and the end time in a way that
the test is in the range between STIME and ETIME...
regards rich...
--- Josué Conti [EMAIL PROTECTED] wrote:
Hi Richard.
Thank you very
2006/6/26, Josué Conti [EMAIL PROTECTED]:
Hi Richard.
Thank you very much for its attention. In the
reality what is occurring is
that in some originated calls of the HiPath with
destination to the Asterisk
they are being without the dumb and rings. I do
not have this parameter
Hi,
which Hicom and which version is installed?
Hicom 300 or Hicom100?
rich
--- Lito Lampitoc [EMAIL PROTECTED] wrote:
Hello all,
I'm new to asterisk. Our company wants to setup an
asterisk server and will
eventually move to IP centric phones, but they don't
want to just throw away
hi all,
The HG3550 V1 and HG3550v1.1 only supports H.323 V.2.
I'am not sure but i thing that the feature CallerID
Name was introduced in version 3 of the H.323
standard. More informations about the owerviews at
http://www.packetizer.com/voip/h323/.
-Concerning HiPathv3.0.
In version 3.0 the
PBXs.
Something strange is that in the HISTA you have
severals CIRCUIT EXT DIALTONE ERROR from the TM2LP
(analog trunk line).
i will compare tomorrow the COT with the one we have
configured at the office...
rich
--- Josue Conti [EMAIL PROTECTED] wrote:
Hi Richard
Thank you very much for its
production versions of both.
Regards,
Richard
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Hi Josué
if the Siemens phone calls Asterisk, it didn't get a
dial tone from Asterisk? Is it correct?
if yes, this is depending of Asterisk which didn't
generates a ringback messages as it expexts dial ton
generation localy. So try this workaround for HiPath
local dial ton generation:
- Add
providers cover the
entire area.TIA,W___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Richard C. Schroeder
[EMAIL PROTECTED
# This script asks asterisk to rotate its logs on its own.
postrotate
/usr/sbin/asterisk -rx logger rotate
endscript
is what we use and it seems to be just fine.
(logger reload reopens the log files, where logger rotate,
rotates then then reopens)
Matt Florell wrote:
Welcome
]mode=filesdirectory=/var/lib/asterisk/moh/default turby@ www.canistec.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard ReinaSent: Wednesday, June 07, 2006 5:30 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Music On Hold
you very much. Richard __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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version ?, asterisk needs a specific version in order to work... - Original Message -From:Richard Reina To: asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 6:02AM Subject: [Asterisk-Users] Music On Holdnot working with new 1.2.7.1 install I
Hi,
maybe http://www.oreka.org
--- Vic [EMAIL PROTECTED] wrote:
Hi, I was wondering if anyone knows of a opensource
SIP
voice logger.
I need to simultaneously record around 150 to 200
sessions.
I figured that if I just set a mirroring port on the
switch and just send all RTP
hi,
maybe http://www.oreka.org
--- Vic [EMAIL PROTECTED] wrote:
Hi, I was wondering if anyone knows of a opensource
SIP
voice logger.
I need to simultaneously record around 150 to 200
sessions.
I figured that if I just set a mirroring port on the
switch and just send all RTP
Hi all,
i am playing around with several optipoint4x0 and run
into trouble trying to get hint functionality to work.
I notice that there is no status notifications. But
afaik this should be implemented via the
SUBSCRIBE/NOTIFY mechanism.
I can see INVITE, TRYING, RINGING, ACK, BYE but no
Hi,
first of all, sorry for this long thread... I have
changed my extensions.conf like you suggested and
delete the line with subscribecontext=notify. But
unfortunately i still don't see subscribe request in
the sip debug trace.
SIP Debugging enabled
kingcoco*CLI
-- SIP read from
configuration on
the IP-phone?
thx in advance
--- Avi Miller [EMAIL PROTECTED] wrote:
On 17/05/2006, at 8:27 PM, richard Coco wrote:
unfortunately i still don't see subscribe request
in
the sip debug trace.
Have you configured your phone to subscribe to the
extension? :)
cYa,
Avi
*snipped
I like the idea that I can just pick up the phone, dial an extension, record
an announcement, and be sure that announcement will play during extended
hold times.
Thanks for the ideas, Richard!
It just came to me, about using timeouts on the queues, play an announcment,
update
Hi,
i have change my sip.conf and my extensions.conf but
unfortunately nothing change. Should i not see the
hint priority in the CLI?
richard
--- Steve Davies [EMAIL PROTECTED] wrote:
On 5/12/06, Jerry Jones [EMAIL PROTECTED] wrote:
I believe the hint priority must be in the same
context
A.J. Paxson wrote:
Hi All!
I've really been struggling trying to get around this. Instead of the same
announcement being played over and over again, I want to be able to play
more than 1 announcement in a queue.
Does anyone have any brainstorming ideas on how I can try this?
Once a caller
A.J. Paxson wrote:
On 5/15/06 10:55 PM, Richard Lyman wisely said:
A.J. Paxson wrote:
Hi All!
I've really been struggling trying to get around this. Instead of the same
announcement being played over and over again, I want to be able to play
more than 1 announcement in a queue
A.J. Paxson wrote:
On 5/15/06 10:55 PM, Richard Lyman wisely said:
A.J. Paxson wrote:
Hi All!
I've really been struggling trying to get around this. Instead of the same
announcement being played over and over again, I want to be able to play
more than 1 announcement in a queue
/forum.php?thread_id=10291962forum_id=43045
Regards,
Richard
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Hi all,
i am desperating, trying to configure an OptiPoint410
with the hint priority.
Here what i have...
OptiPoint410std- exten 2001
X-Lite - exten 2002
But unfortunately no LED ON on my OptiPoint410
sip.conf
[2001]
type=friend
context=local
host=dynamic
dtmfmode=rfc2833
incominglimit=1
Patel.
Regards,
Richard
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by a codec
mismatch, but I'm sure there are other factors that will give the same
result.
Sorry I can't offer any more.
Regards,
Richard
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codec;
disallow=all
allow=codec of choice
at the asterisk end and whatever you need to do at the legacy end.
Regards,
Richard
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is that current PCI-X boards are 3.3V only now.
The board you mention would appear to have PCI 5V slots.
Regards,
Richard
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http
try http://sourceforge.net/projects/web-meetmeChris Blunt [EMAIL PROTECTED] wrote:Hi List, Is it possible to store meetme config in a MySQL table?If so, any pointers would be appreciated.ThanksChris --Chris Blunt Entropy IT Ltd
Hi all,
i have an Asterisk box with an Eicon 4BRI with
chan_capi-cm and every thing works fine. We now plan
to install a new Asterisk using a Dialogic BRI/2VFD.
Is the Dialogic card supported and can i use
chan_capi-cm? Has anyone managed to install this card?
Unfortunately i was unable to find
?
Cheers
Richard Dutton
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them, one would expect Cisco's to work well also.
Regards,
Richard
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Hello All,
We have an installation that has Aastra analog phones
connected to the asterisk server with Sipura ATA devices. (It was done this way
in order to use existing wiring).
Is there any way to implement a page all by
turning on the speakers in these phones (like you can do by
progress passing it to Zap/46-1
-- IAX2/VoIP-1 stopped sounds
-- IAX2/VoIP-1 answered Zap/46-1
Richard G. Cavanna
Information Technology Manager
SyChip Inc.
P - 972.202.8840
F - 972.633.0327
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Please post pertinent config files and a CLI output so the list can help
with the 10 sec delay
You set codec selection in SIP.conf. This selects preferred codec from
top to bottom as well as jitter buffer settings and the RTP timeout.
Sip.conf
disallow=all
allow=g729
allow=gsm
allow=ulaw
;); define ("HOST", "localhost"); // I think your MySQL serverdefine ("PORT", "3306");define ("USER", "root"); // user to DBdefine ("PASS", "mysqlpass"); // password to DBdefine ("DBNAME", "DB_CDR");d
Hi list,
i am playing around with asterisk manager interface
(and astriskjava) and i notice that the login is not
case sensitive.
so i can use
username: admin
secret: admin
---
# telnet localhost 5038
Trying 127.0.0.1...
Connected to
Do you have an idea when this new submission will be available?Thanks. Dan Austin [EMAIL PROTECTED] wrote: Sorry for the late reply, I was away on vacation.Version 1.2 was created by Areski and I extended it to include the scheduling functions. I guess I should get an account on
I had this same problem with SX2000. I think you have to configure the Mitel to "Speed Dial" to your Asterisk server. What I think is happening is that the PBX grabs a trunk but does not "dial" into Asterisk.Ask the person configuring the Mitel PBX to setup an extension (e.g. 1234) when
or the version that they have installed?
Richard
On 3/29/06, Dovid Bender [EMAIL PROTECTED] wrote:
snip
wonderful place to start. Nothing against Asterisk or Linux. My build fromscratch issues are only due to my lack of Linux experience...
/snip
the only way to learn is by playing. a little over a year
I am able to provide local installation, configuration, and troubleshooting in the northwest region (Portland to Seattle and surrounding).
Please feel free to inquire for further details.
Richard Amerman
7 Tech NW
360-931-2721
On 3/25/06, Bart Fisher [EMAIL PROTECTED] wrote:
Maybe I could help
Check you voicemail.conf file and make sure
'delete=yes' next to
the appropriate mailbox
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= yestransfer = yescancallforward = yescallreturn = yesechocancel = yesechocancelwhenbridged = yesechotraining = 400rxgain =
0.0txgain = 0.0callgroup = 1pickupgroup = 1immediate = nochannel = 1-23 Thanks in advance.richard ___
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. It just did not create the channels.Learning a lot. It's painful but fun.Thanks.richard"Kevin P. Fleming" [EMAIL PROTECTED] wrote: Richard OSS wrote: rxgain = 0.0 txgain = 0.0 callgroup = 1 pickupgroup = 1 immediate = no channel = 1-23Where did you find any example that sugge
If you are using connecting the card to a smart jack (incoming line from Telco), then you need a straight T1/E1 Cable, which is identical to a straight Ethernet cable. If you are doing a back-to-back configuration, or connecting the card to another PBX or channel bank, then you need a cross-T1/E1
Kenige Ho wrote:
the ooh323 is from Asterisk-addon-1.2.1. Is there a bug on this version
for the ooh323 and also how can i get the newer version of the
ooh323(0.8.1) to compile with? Many thanks to you all.
You will find 0.8.X in the asterisk-addons svn branch.
Regards,
Richard
firmware 1.0.2.8 is a beta version, the latest beta version is 1.0.2.13.
You may want to try the latest beta.
http://www.voip-info.org/tiki-index.php?page=GXP-2000
I am using stable firmware 1.0.1.12, and it has no major problem.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hello,I am trying to connect a TE210P to a SX-2000 but zttool shows RED ALARM. The SX-2000 provides the internal T1 to the * server.Telco - SX-2000 TE210PMy config file is below which I got from http://www.voip-info.org/wiki/view/NFAS /etc/zaptel.conf span=1,1,0,esf,b8zs
.richard ___
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/perluse Asterisk::AGI-new();my ($contno) = $ARGV[0];open (TEST, "/home/richard/TEST.txt") or die "Can't open" print TEST "CONTNO " . $contno . "\n"; close (TEST);use DBI; my $dbh =
DBI-connect("DBI:mysql:database=accounting;192.168
Hi Jordan,We are planning on building the same thing. We are still waiting for the hardware. We are using a Dell PE 2850 3GHz with 2G of RAM and a TE210P.I asked Digium support if this can suupport 50 users in one conference and the tech support guy said yes. Here's also a response
Hi,
if yo are looking a way to interconnect Asterisk with
a HiPath 4000 via IP, so you have 2 ways to do it.
- via oh323 (for HiPath 4000 version 1 and 2)
- since HiPath4000 version 3 you are able to
interconnect using sipQ (SIP Trunking)
--- Viktor Tatianin [EMAIL PROTECTED] wrote:
Hello
Hi again,
i don't think that the HiPath2000 is an Asterisk based
system. AFAIK the HiPath2K is only configurable using
a Web-based tool (no console access). For the moment
the HiPath2K will only be release with CornetIP (HFA).
No SIP (panned in a second step) and unfortunazely no
IAX are
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