I am looking for cards or boxes which can be plugged in between 1 or 2
ISDN and a legacy PBX system (ISDN).
Same with 2xPRI (E1)
DSS1 Euro ISDN
Either an asterisk box with such cards will be put between the ISDN
lines and the PBX system or just a box and connect to a remote asterisk
box.
Hello,
I am implementing asterisk in my office in India. I am
facing a problem of caller ID. Any kind of help is appreciated.
As a call comes to asterisk console it shows
ERROR[27863]: callerid.c:260 callerid_feed: fsk_serie made mylen 0 (-7)
WARNING[27863]: chan_zap.c:5434
Michiel van Baak wrote:
Some of our customers are asking us about DECT solutions for
their asterisk install. Some others will not go to asterisk
if there won't be a DECT solution.
They now have a Siemens or a Samsung PBX. Those PBX-es come
with a DECT basestation and optionally repeaters etc.
Yoann Le Bihan wrote:
2005/8/17, Michiel van Baak [EMAIL PROTECTED]:
Is there any other solution like this out there that works
with asterisk ?
Why don't you use WiFi VoIP phones like ZyXEL 2000W (which is not such
expensive compared with Cisco ones...) ?
Because if you have a network of
Rich Adamson wrote:
I have been reading with great interest the posts on trouble shooting
echo cancellation with *. Is it just coincidence that all of this
discussion has been with analog lines. Are PRI's susceptible to echo
problem like POTS lines.
Keep reading. Echo _can_ occur
Hi!
I have the Kirk IP600, technically it's great and *the* best solution for
wireless phones and VOIP. WiFi phones plainly suck for a dozen of reasons
and IMHO are totally useless.
However, here comes the bad part. Currently you need to use the chan_sccp
driver to talk to it and that is
Hello,
I have seen that SpanDSP supports V.17 faxing, however when I tryed to
send pages, I eneded with very ugly pages (unreadable). Did anybody else
try that?
Yes, I checked frame slips and clocking on PRI, everything has to be OK.
Regards,
Tamas
I want to let Asterisk read its configuration from a mysql database. I
configured everything according to the wiki page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+res_config.
However it doesn't work. I am using 1.0.8 asterisk version and here are
my config files:
Extconfig.conf:
Hi Michiel,
I have a Kirk set which should be able to do H323, but I haven't had time yet
to try it. They have SCCP and H323 types, and ofcourse there are sets which
can be connected via an E1 link.
I tried H323, it has some severe problems (at least with asterisk). First
of all it is
Hi,
I have an E1 ISDN connected to the local telco.
The office line answers, plays the intro message and expects the caller to
press a key (for instance after the beep press 1 for reception or the 3
digits extension).
I wonder if it is possible that if one is calling from an a
Jaco, its probably in the IP's you allow in the /etc/asterisk/manager.conf
permit= ?
Otherwise look up in http://www.asternic.org/
Regards
Rob
On 8/17/05, Jaco vd Westhuizen [EMAIL PROTECTED] wrote:
I am running asterisk at home but have a strange phenomena that is going on
with my flash
is your Asterisk compiled from cvs head?
Kun
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Frank
Aartman
Sent: Thursday, August 18, 2005 3:10 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk configuration from database
As more I read the manuals, as more I am unsure how to do it ;-(
I have an * box and I have on a remote place a phone line.
I want to use this remote phone line for dialing out.
All boxes I read the manuals from have an FXS and FXO and connect these
directly if the box is without power. It
Is there an agent status that can be set or read for a user to be
unavailable for another call.
eg. Agent takes a call, the call is ended, but they have some data entry to
perform before they can take another call. Then let asterisk know they are
ready for another call.
I'm guessing this is
Am Mittwoch, 17. August 2005 17:06 schrieb Craig Guy:
Hi Bartek, I posted the exact same problem last week - I found that if I
connected two Asterisk systems together via a PRI crossover cable and talk
txfax to rxfax then you get a T4 state timeout. I tried connecting ports
one and two
Am Dienstag, 22. März 2005 00:04 schrieb Chris Tuska:
Hello All,
I am looking to get asterisk to receive faxes but I really don't know where
to go from here, any help pointing me in the right direction would be
great. What I would really like is for this second line to answer faxes,
but if
[EMAIL PROTECTED] wrote:
Nice looking device.
Does it support DECT repeaters?
I cannot rely on 1 basestation for my handset when I walk
around in the location. The Siemens stuff has 10
antennas/repeaters/extenders in the building. It's a bit
overkill, but the Siemens guys tend to love doing
Hi,
I do not know of an existing application which does this out of the box.
But what you want should be 'simple' to implement with AGI script(s).
You need to have some programming knowledge and knowledge of AGI though.
For the auto dial out there is already info on the site:
On Thu, 18 Aug 2005, Andreas Sikkema wrote:
[EMAIL PROTECTED] wrote:
Nice looking device.
Does it support DECT repeaters?
I cannot rely on 1 basestation for my handset when I walk
around in the location. The Siemens stuff has 10
antennas/repeaters/extenders in the building. It's a bit
This phone works fine, however the initial firmware it came with was
awkward. Once updated no problems, even NTP works!
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: 11 August 2005 01:23
To: asterisk-users@lists.digium.com
I found it!! (Heh, probably to easy to mention for most of you but
being unable to read any code at all I'm quite happy) :)
After applying florz patch it seems that the Makefile is ignoring the
linux26 option. The Makefile should change the $KSRC to the proper
directory i.e.
It sounds like web_hostname in your /var/www/html/panel/op_server.cfg is set
to your external ip. If you change it to your internal ip, I think you'll
have the opposite of what you describe. I couldn't find a decent solution to
this dilemma. Any one?
- Original Message -
From: Jaco
Hello there.
Is there somebody with this connection working? I can't seem
to make this work at all. Could someone
please share some .conf files?
Cheers,
Vedran.
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Nope, I got the stable 1.08 release from cvs.
Frank
From: Wei Kun [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk configuration from database
withres_config
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL
I am running AGI
script written in C.My script gets triggered but there is no action.
I have taken the samplec file from http://home.cogeco.ca/~camstuff/agi.html.
it says
launched AGI script /var/lib/asterisk/agi-bin/sample.cexec format error. Did I miss
anything?
I
am running
Armin Schindler wrote:
What kernel do you use? If it's 2.6.10 or newer, then make sure you use new
chan_capi-cm from sourceforge.net. Older chan_capi is buggy.
Thanks for the hint. I am using kernel 2.6.12. Changing to the
chan_capi-cm solved the problem... Looks like I could have saved
Hi
Is it possible to start recording a call during the conversation?
--
Eric Smith
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hi all,
I known taht this ML is about *, but lot of you are using BT telepnones.
I'm using FW 1.0.6.7 for all phone. This firmware support VLAN tagging for
QoS on the Layer2. I use it to separe the PHONE network form the PC network,
which are in PC connector in the BT.
And I also use the TFTP
We currently use v1.0.9 here, Monitor( ) seems to work fine, shows us
the dialplan.
It should be working if you configure the dialplan correctly.
Since you are using v1.0.9 have you gotten Queue( ) and Agent channel
dialing to work with it?
On 8/12/05, Victor Alvarez [EMAIL PROTECTED] wrote:
Hi All,
How can I lock the extension in Asterisk?
For example , my extension is 1000 and I am away for business trip. I
want to lock my extension during my absence.
Can it be done in Asterisk?
regards,
Stephen
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Hello all,
i am using * 1.0.9 and chan_capi-cm 0.5.4. So far everything works good.
But if i want to initiate an call over an capi channel either from the
manager api or from an call file * quits with an segfault. If i do the
same thing over sip channels everything works perfectly. Had anyone
Hello,
I want to see the output that goes to STDERR...Anyone knows where this
output can be found?
Thanks
Marios
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.338 / Virus Database: 267.10.12/75 - Release Date: 17/8/2005
I've this setup :
CiscoAta186 - asterisk with oh323 chan - gsmgateway
dtmf doesn't work, tryed inband, with g711a and g729 codecs
CiscoAta186 - gsmgateway works, even with g729, so it seems the problem
is in *
oh323.conf has inBandDTMF=yes, what else may I need to tweak ?
Quoting Eric Wieling aka ManxPower [EMAIL PROTECTED]:
[EMAIL PROTECTED] wrote:
Thank you W, asking differently:
(Suppose) I have a very reliable hardware,motherboard,power
supply,bios,kernel,configuration, I have reliable and fast everything.
Now what is the maximum number of
Hi,
A note for people spending hours to get ISDN equipment work together
with Digium TDM equipment
I have 1 OctoBri and 1 TDM40B and tried bristuff-0.2.0-rc8f-cvs.
modprobe qozap works well, but modprobe wctdm does not find a board.
(while wctdm works fine with zaptel-cvs-head)
So, I just
hello
i am trying to use res_odbc for sipuser. my connection
is working. i have checked using isql. even cdr_odbc
is working but i hav problem in res_odbc. i have
created user in sip_buddies table but asterisk is no
getting user from this sip_buddies table.
/etc/asterisk/extconfig.conf
Yes, there are different ways depending on how you use your system.
The most unobtrusive way to the call is to use a computer app like a
webpage or simple GUI app to send a Monitor command for the channel
you are on(astguiclient does this: http://astguiclient.sf.net)
or
If you are using
I got it working, below it's the oh323.conf, I'd done
nothing in particular to make it work, just define it as node in the
gnugk
ListenAddress=0.0.0.0listenPort=1720
connectPort=1720
tcpStart=1tcpEnd=2udpStart=1udpEnd=2
fastStart=yesh245Tunnelling=yes
h245inSetup=yes
Remarque : message transféré en pièce jointe.
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur
As more I read the manuals, as more I am unsure how to do it ;-(
I have an * box and I have on a remote place a phone line.
I want to use this remote phone line for dialing out.
All boxes I read the manuals from have an FXS and FXO and connect these
directly if the box is without
On Thu, 18 Aug 2005, Tobias Wolf wrote:
Hello all,
i am using * 1.0.9 and chan_capi-cm 0.5.4. So far everything works good. But
if i want to initiate an call over an capi channel either from the manager api
or from an call file * quits with an segfault. If i do the same thing over sip
I would like to know if gsm codec is supported on the 79xx series of Cisco
Voip Phones.
It seems not to be supported, is it right ?
I am looking for phones supporting that codec (AT 320 on example)
Thanks in advance,
Andrea
Chi ricevesse questa mail per errore e' gentilmente pregato di
Thanks for the ideas Chris; most appreciated.
Hugh
On 8/18/05, Chris Coulthurst [EMAIL PROTECTED] wrote:
If you want to use samba for their checking, you COULD use smb.conf to
control which users get in to which folders. You might make
/var/spool/asterisk/voicemail/default readable by all,
Title: Message
Hi,
Is
there a module within the Asterisk standard distribution that provides SNMP
features?
Is
there any third party software for that purpose?
Regards,
Stojan
Sljivic
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Asterisk-Users mailing list
Hi!
Still trying to get fax to work in a reliable manner.
I have some dumb fax machines that do not have a setting to limit tx/rx
speeds. For some reason when training they often think the line is good
enough for 14k4 or higher after which the fax will fail.
Is there any other way of
I would like to know if gsm codec is supported on the 79xx series of Cisco
Voip Phones.
It seems not to be supported, is it right ?
The Cisco 79X0's do not support gsm.
I am looking for phones supporting that codec (AT 320 on example)
Don't know.
hello
try http://www.faino.it/en/asterisk.html
harry
--- Stojan Sljivic - GDS
[EMAIL PROTECTED] a écrit :
Hi,
Is there a module within the Asterisk standard
distribution that provides
SNMP features?
Is there any third party software for that purpose?
Regards,
Stojan Sljivic
Hi,
Can i set Password to Conf rooms?Please Let me know the procedure.
Thanks
Sharadindu MohantySharadindu Mohanty
To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre.___
Asterisk-Users mailing list
Hi Harry,
I have tried to install it, but it gives compilation errors on Fedora Core
3.
Also, patch file for Asterisk was not in sync with version of Asterisk I
had, but I managed to apply changes manually.
Did anyone succeeded to run that package on FC3 or Red Hat Enterprise 3?
Regards,
On Aug 17, 2005, at 4:18 PM, Ryan Stark wrote:
So one of my employees just got married and is taking her husband's
last name. As soon as her business cards run out she will be using
the new last name for business as well. So for example is there a way
to make it so that if someone goes to the
yes
to set password in conf, edit archive meetme.conf
or meetme_addtional.confthen in number of conference add ",xx"
exemple:
conf = 8200,123
- Original Message -
From:
Sharadindu Mohanty
To: Asterisk
Sent: Thursday, August 18, 2005 9:26
AM
Subject:
I solved all my problems on eyebeam videophone, and now they work.
All I had to do is to enable only H263 protocol (Basic H263 Protocol).
I also succeded in configuring them behiand a firewall /NAT so now is all
ok
Thanks all
Andrea
___
As everyone else has said, you *might* be able to make it work, but
it is extremely unlikely. You want to buy a T1 interface and a
channel bank.
Tom
On Aug 17, 2005, at 7:38 PM, [EMAIL PROTECTED] wrote:
Is it possible to use 24 FXS/FXO lines(on 6 PCI slots) at the same
time on the
On Thursday 18 August 2005 06:34, [EMAIL PROTECTED] wrote:
Because 800$2000$ (interms of hardware).
(For 12 Ports: The min. price of T1(500$)+channel bank($1,500)=2000$.
While the min. price of 3 TDM400P with 24 modules is just=800$.
Your solution is still 400$ more expensive for 24 ports.)
On Wednesday 17 August 2005 3:04 pm, Tzafrir Cohen wrote:
On Wed, Aug 17, 2005 at 01:27:08PM +0300, [EMAIL PROTECTED] wrote:
Hi!
I'm trying to start asterisk at boottime. Since SuSe
It was SuSE (the old way). Now it is SUSE. Was it ever SuSe?
Nope, but it was S.u.S.E. before SuSE:
What problem are you trying to solve with this? Just stepping out on a
limb but it sounds like you are trying to swat a fly with an F-16.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christoph
Eicke
Sent: Wednesday, August 17, 2005 4:34 AM
When I attempt to compile the zaptel driver (latest CVS HEAD) I get this
compile error:
You do not appear to have the sources for the 2.6.11.4-20a-default kernel
installed.
make: *** [linux26] Error 1
When I load YaST I see kernel-source 2.6.11.4 as installed version. So why
do I get this
On Wed, 2005-08-17 at 17:27 -0500, Mojo Jojo wrote:
I don't mind buying an appliance to get something solid but IP Cop just
looks better than he appliances I see out there.
I am only concerned if it is stable for a production environment. It says
it's designed for a SOHO environment, we
Well thanks I have changed the ip address now as you said the opposite is
true however is there a command in asterisk that we can make both visible in
var/www/html/panel/op_server.cfg ?
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I'm trying to solve my Nikotel problem (see previous post) where the problem
is that I get a hangup after 2 minutes, therefore I need some number that
doesn't cost anything and gives me some audio for a long time...
On Thursday 18 August 2005 15:08, Jonathan k. Creasy wrote:
What problem are
Hi im using [EMAIL PROTECTED] with 30 phones and 50 users, I want the users to
utilize pins when making external calls asterisk supports this, but I want
to take this one step further I want each user to have his /her own pin that
is connected to the mysql databases so in the monitoring system I
Hello,
I hope someone can help me with this. I have come across a
few other people who seem to have experienced this problem but the answer was
never posted.
I am trying to listen to voicemail by dialing for the
main voice mail menu..However I hear nothing. The Asterisk console
Tamas J wrote:
Hello,
I have seen that SpanDSP supports V.17 faxing, however when I tryed to
send pages, I eneded with very ugly pages (unreadable). Did anybody else
try that?
Yes, I checked frame slips and clocking on PRI, everything has to be OK.
Regards,
Tamas
V.17 is disabled
I too am starting out with Andy Powell's Little Asterick Handbook
following it letter by letter and having the 403 forbidden issues with
my XLite client.
Before I start to work this out... did you find anything?
John Gruber
___
Asterisk-Users
Ma Zhiyong wrote:
I'm sure use my telco as clock src, and my libtiff is also v3.5.7,
while problem still exist.
Shall I contact with my telco for timing?
in zaptel.conf, I set
span=1,1,0,ccs,hdb3
span=2,2,0,ccs,hdb3
span=3,0,0,ccs,hdb3
span=4,0,0,ccs,hdb3
To trace rxfax, just turn on debug
Unfortunately as of right now this unit is designed as more of a home use
device. Repeaters are not supported currently. We are working with the
vendor in either creating a firmware upgrade that supports more
functionality that offices need or designing a higher end version that will
support
Just call tech support for a large company. Your always on hold longer
than 10 minutes!
Adam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christoph
Eicke
Sent: Thursday, August 18, 2005 8:15 AM
To: Asterisk Users Mailing List - Non-Commercial
We use the Areskicc calling card system as an authentication system. It
does everything you are asking and can generate great reports and
graphs. I like it very much.
That being said, areskicc is tough to get going, but there is plenty of
info here:
In theory, using a URL instead of an IP address in web_hostname should solve
this issue. But last time I tried it didn't work (you may need to set DNS
records of your ADSL modems). I should try again.
- Original Message -
From: Jaco vd Westhuizen [EMAIL PROTECTED]
To:
try calling comcast.. they are always good for at least 15 minutes of
hold 18778242288
qworst(qwest) works too.. 1800244
On Thu, 2005-08-18 at 06:28, Adam Vocks wrote:
Just call tech support for a large company. Your always on hold longer
than 10 minutes!
Adam
-Original
Gurminder Arora wrote:
Hello,
I am implementing asterisk in my office in India. I am
facing a problem of caller ID. Any kind of help is appreciated.
As a call comes to asterisk console it shows
ERROR[27863]: callerid.c:260 callerid_feed: fsk_serie made mylen 0 (-7)
Dear Friends,
Iam useing the latest CVS-HEAD of the asterisk
when i use the agentcallbacklogin for the Agent
side apart from the password it is askeing the new
location ?
and its not accepting the any extension
it gives the error message like this
i could not able to found any related
ln -s /usr/src/linux-2.6.11.4 /usr/src/linux-2.6.11.4-20a-default ?
On Thu, 2005-08-18 at 06:02, Angus Comber wrote:
When I attempt to compile the zaptel driver (latest CVS HEAD) I get this
compile error:
You do not appear to have the sources for the 2.6.11.4-20a-default kernel
Because 800$2000$ (interms of hardware).
(For 12 Ports: The min. price of T1(500$)+channel bank($1,500)=2000$.
While the min. price of 3 TDM400P with 24 modules is just=800$.
Your solution is still 400$ more expensive for 24 ports.)
Sorry but to have 24 ports you need 6x TDM cards, which adds
Hello,
I have installed festival (the rpm package came with fc4).
But getting this:
client(1) : rejected from myserver not in access list
whenever I try to access it from asterisk .
I found in documentation:
If you see a message such as:
client(1) : rejected from myserver.mydomain.com not in
John Gruber wrote:
I too am starting out with Andy Powell's Little Asterick Handbook
following it letter by letter and having the 403 forbidden issues with
my XLite client.
Before I start to work this out... did you find anything?
John Gruber
in your sip.conf entry for your phone.
Sorry, the file I am talking about is in right place !!
but not sure what to add in festival_server script.
Thanks,
Hello,
I have installed festival (the rpm package came with fc4).
But getting this:
client(1) : rejected from myserver not in access list
whenever I try to access it from
Todd Weiser wrote:
We recently changed our asterisk system to begin using G.729a as the
primary codec. We have a Cisco 1700-series router which connects to the
PSTN via FXO ports, along with Cisco 7940 SIP phones. Everything is
working great, except... When an inbound caller calls into our
Pudenz, Duane wrote:
We are testing our Asterisk server prior to deployment. The server has
a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and
one PRI for local calls.
We are using sipp from two different stations routing a test number out
the LD lines and another test
festival_server.scm -- should be in /usr/share/festival
grep for '(localhost) and replace with nil (don't use double quotes).
Innocent Evil wrote:
Hello,
I have installed festival (the rpm package came with fc4).
But getting this:
client(1) : rejected from myserver not in access list
i guess you are trying to run a non compiled C program, you have to
compile it first.
# gcc sample.c -o sample
then from asterisk run the executable 'sample' without the extension '.c'
best regards
On 8/18/05, someshwarak [EMAIL PROTECTED] wrote:
I am running AGI script written in C. My
Stojan Sljivic - GDS wrote:
Hi Harry,
I have tried to install it, but it gives compilation errors on Fedora Core
3.
Also, patch file for Asterisk was not in sync with version of Asterisk I
had, but I managed to apply changes manually.
Did anyone succeeded to run that package on FC3 or Red Hat
Asterisk Supporter wrote:
How real time is realtime? If the extensions.conf is stored in the
database, does * query it row by row or is it cached? In other words,
given the following exerpt:
If you store the .conf file using realtime, then yes it is cached. If
you simply use realtime
Hello,
Do you succes to apply changes manually ?
Harry
//
from Andrea Fino:
harry gaillac wrote:
Remarque : message transféré en pièce jointe.
Do you update ast-ax-snmpd according asterisk
releases
?
harry
No, it's some time I don't work on
That wiki page is old, ugly and out of date. There are many like it and
if I only knew how to delete wiki pages, I would clean it up some.
The easiest way, Frank, to do what you want is to download CVS-HEAD and
use ARA to store your config files. Also download addons from HEAD and
you can use
Have you looked in the debug log for any errors? Have you checked the
ODBC connection using Asterisk CLI? I believe if you type 'odbc ?' then
you will see a command to check the status of a connection.
-Matthew
Kamran Ahmad wrote:
hello
i am trying to use res_odbc for sipuser. my connection
Matthew
Could you tell us the differences between your project
and ast-ax-snmpd .
Regards
Harry
--- Matthew Boehm [EMAIL PROTECTED] a écrit :
Stojan Sljivic - GDS wrote:
Hi Harry,
I have tried to install it, but it gives
compilation errors on Fedora Core
3.
Also, patch file for
Hi Harry,
Yes I have succeeded to apply them manually.
Regards,
Stojan Sljivic
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
harry gaillac
Sent: Thursday, August 18, 2005 16:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi Matthew,
Nice to hear that.
What will be the license type for your SNMP module?
Are you going to include it in the Asterisk CVS or it will be independent
product?
Regards,
Stojan Sljivic
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew
Hi,
Armin Schindler schrieb:
Can you please provide a verbose log level 5 with 'capi debug'?
A backtrace could be of help too.
thx for your answer.
i hope these information gives you some insights. i have asterisk write
an core dump and after that i executed:
gdb asterisk core.8301
which asterisk release do you apply it ?
Harry
--- Stojan Sljivic - GDS
[EMAIL PROTECTED] a écrit :
Hi Harry,
Yes I have succeeded to apply them manually.
Regards,
Stojan Sljivic
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
From: [EMAIL PROTECTED]
Sent: Thu, 18 Aug 2005 08:23:45 -0600
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Festival error
festival_server.scm -- should be in /usr/share/festival
grep for '(localhost) and replace with nil (don't use double quotes).
Isn't this change
Does anyone on-list have any experience with this device? It's a 6 port
FXOSIP gateway.
Michael Graves
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. [EMAIL
I'm testing one now. We got the 2x and the 4 port add-on license.
It seems to be a nice product. It's a basic VOIP PBX. We are using it to
provide a customer site with 2 SIP Trunks, 6 phones, voicemail, failover
to PSTN and 2 FXS ports.
We are evaluating whether we want to deploy asterisk to
I thought that I would try this on iax.conf as well however I still get
asterisk asterisk as the callerid name and num.
I have the latest CVS as of 8/17/05
Has anyone have this working with iax incoming?
Thanks
Ben
That worked. The following line also got rid of asterisk without
entering
Wow that is overkill (3 bases every 20 meters) could have/should have
been better planned for by use of overlapping cells covering not only
office but walkway areas as well.
The most I've ever seen is 8 base stations stacked 1 on top of the other
on a trading platform where all of the
Title: CRM software
Can anyone recommend CRM software with a link into Asterisk? I would like a pop up on caller ID if possible. I've played with the FOP and SugarCRM but can't get them working together.
Regards
Lee
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Asterisk-Users mailing
Stojan Sljivic - GDS wrote:
Hi Matthew,
Nice to hear that.
What will be the license type for your SNMP module?
Are you going to include it in the Asterisk CVS or it will be independent
product?
Regards,
Stojan Sljivic
Same as asterisk. It will most likely be in asterisk-addons.
On 8/18/05, Lee Archer [EMAIL PROTECTED] wrote:
Can anyone recommend CRM software with a link into Asterisk? I would like a
pop up on caller ID if possible. I've played with the FOP and SugarCRM but
can't get them working together.
Regards
Lee
harry gaillac wrote:
Matthew
Could you tell us the differences between your project
and ast-ax-snmpd .
Regards
Harry
Not really many differences at all. It's basically Andrea's code cleaned
up. His code was scattered across about 10 different files/headers. I
combined them into one
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