On Wed, 2005-03-16 at 16:20, Razza wrote:
Chris Blake wrote :
-%-
If anyone can help I`ll send the call file to you, or is it ok to
clutter the list with it ?
-%-
'Clutter' the list I'd be interested and at least it is pertinent to *
;o)
Howdy Razza and Stefan, thanks
On Thu, 2005-03-17 at 11:34, Alexander Lopez wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohammed
Firdosh Nasim
Sent: Tuesday, March 15, 2005 11:08 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] How to register two
Kong wrote:
Is there any application that actually work like Background, but instead
of playing a specified file, it plays the streaming music from music on
hold?
the reason i am asking this because i come across a dialplan that goes
this way,
if a person gets to an extension that is busy, it
Hi,
I need the option to Record certain conversations through * on our help desk.Id like toarchive these for later access.Please can someone point me in the right direction as search has brought up nothing.
Thanks
Mike
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Hi there,
I'm using ser and asterisktogether. Asterisk
for voice mail etc and ser forregistration of the users
usig database.I can restrict forwarding
callsfrom another sip proxy to ser(using proxy_authorize) but how
can I restrict access to asterisk ... Now everyone can forward calls to
On Thursday 17 March 2005 06:13, Geoff Nordli wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of David Uzzell
Sent: Wednesday, March 16, 2005 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Pavel Siderov - Hostmates wrote:
I can restrict forwarding calls from another sip
proxy to ser (using proxy_authorize) but how can I restrict access
to asterisk ... Now everyone can forward calls to my asterisk and
can place pstn calls.
Use iptables on the asterisk machine to only allow
I need the option to Record certain conversations through * on our
help desk.Id like to archive these for later access.Please can someone
point me in the right direction as search has brought up nothing.
Check out the Monitor option. (just search the wiki for monitor) Loads
of examples and
Hello Everyone,
I am trying to find a single port isdn pci card in the usa for
asterisk,
but it seems everything is abroad.
Does anyone know a good place to find a BRI S/T and U card for north
america?
Perhaps it could be possible if you get an NT1 box giving you an S0 bus
and then using a
Hi,
I have just published my last few weeks of hard work: IPSwitchBoard BETA.
Please let me know what you think and post comments on the Wiki.
http://www.voip-info.org/wiki-IPSwitchBoard+BETA
I've installed it and tested it, it works great, the idea is great, works
simple enough
for users to
Maybe I've missed it but I'm wondering if there has been any movement
towards getting t.38 support into asterisk.. has there been any news?
Where is t.38 support at? will it even happen?
Steve Underwood is working on it. There's a bounty at
I tried wiki, but I got too many pages (I think all of them), ...as
answer.
I want to write an agi.
I need a HOW-TO, is there anything available?
see the perl agi package from http://asterisk.gnuinter.net/, the
agi/agi-test from the asterisk source and
hello list (3rd try as my first post seems to have gone astray in the endless
realms of tcp/ip and in my second i accidentially replied to another post not
related to the problem),
i searched for nearly a week for a solution to this problem, as there is:
analog fax machine -» grandstream ata
I've installed it and tested it, it works great, the idea is great, works
simple enough
for users to understand :)
One problem though:
If a phone registered under Monitored Extension has two
calls, one Active and one On Hold and if I attempt a transfer via
IPSwitchBoard
it works fine.
On Thu, 10 Mar 2005, Kevin P. Fleming wrote:
http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view
Enjoy!
Anyone have pictures from the Heart show? :( My camera phone just wigged
out. I thought I had like 60 pictures right from the stage, but apparently
it didn't save
On Thu, 10 Mar 2005, Kevin P. Fleming wrote:
http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view
If you look closely, you'll see me at the booth doing some troubleshooting
for Digium during one of my session breaks. We actually setup an IAX2
connection from the main
Hi,
Does anyone have Carrier Access Corporation (CAC) Access Bank I Manual ? Could
you please email it to me off list ?
We have a FXS channel bank and the framing Error Led is blinking and I have no
clue on what could be the problem .
Is there command line utilities available in Linux to
hi
any one tell me how to make a dialplan
my extensions.conf
exten = _40,1,Dial(OH323/${EXTEN})
i want to dial to 40 number.
could be any number like 923335224005 or
92512213248
at the moment when i am trying to dial 40923335224005
asterisk is dialing
Bug or feature, I will look at it and try to solve it.
One more thing while you are at it:
I just installed .NET 2.0 Beta, so I don't know
if it's a problem with your app or with new .NET,
but after a few minutes of running your app and not doing anything (the app
runs idle)
a new window
On 02:42, Thu 17 Mar 05, Kamran Ahmad wrote:
hi
any one tell me how to make a dialplan
my extensions.conf
exten = _40,1,Dial(OH323/${EXTEN})
i want to dial to 40 number.
could be any number like 923335224005 or
92512213248
at the moment when i
On Thu, 17 Mar 2005 02:42:27 -0800 (PST), Kamran Ahmad [EMAIL PROTECTED]
wrote:
hi
any one tell me how to make a dialplan
my extensions.conf
exten = _40,1,Dial(OH323/${EXTEN})
i want to dial to 40 number.
could be any number like 923335224005 or
Kamran Ahmad wrote:
hi
any one tell me how to make a dialplan
my extensions.conf
exten = _40,1,Dial(OH323/${EXTEN})
i want to dial to 40 number.
could be any number like 923335224005 or
92512213248
at the moment when i am trying to dial 40923335224005
asterisk
Hi *,
I want to integrate the Eicon Diva 4Bri Card to Asterisk.
Eicon drivers and capi is installed. I use the latest dev version from
eicon compiled and installed for my fedora 2 system.
I found the chan_capi for asterisk from www.junghanns.net. Also loaded
the patch and applied to the chan_capi
Hello Everybody,
This is Bharat here. I am on the way of learning Asterisks, and I just wished
to know how I go about if got to write dailplans for outbound calls and inbound
calls. If you could provide me with a simple example, I could get thru.
Waiting for your response
On 16:51, Thu 17 Mar 05, Bharat M. Sarvan wrote:
Hello Everybody,
This is Bharat here. I am on the way of learning
Asterisks, and I just wished to know how I go about if got to write
dailplans for outbound calls and inbound calls. If you could provide me with
a
I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I
consistently get one of the following errors:
PRI got event: HDLC Abort (6) on Primary D-channel of span 1
or
PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1
My zaptel.conf file:
Bharat M. Sarvan wrote:
Hello Everybody,
This is Bharat here. I am on the way of
learning Asterisks, and I just wished to know how I go about if got to
write dailplans for outbound calls and inbound calls. If you could
provide me with a simple example, I could get
[EMAIL PROTECTED] wrote:
change the sntp_mode: from directedbroadcast (the
default) to unicast. This will cause the phone to poll
your NTP server. This solved the problem for me.
This fixed my problem as well! Thanks
Doug
___
Asterisk-Users
Greetings *`s,
Further to the above thread, the problem has been solved.
Stefan was correct in stating that it was a permissions error, but we
were only able to catch what permissions the file carried over from its
source directory to /var/spool/asterisk/outgoing by completely stopping
the *
Hi,
Some time I did not touch a cisco.
At a previous job, I managed a 53xx
If I remembered well, you can define dial-peers at ingress and outgress.
The trick is the add a very specific header at ingress and remove it at
outgress.
Also, by then, not all traffic directions where possible on the
Hi,
I have T.38 over UDPTL with SIP signalling kind of working-ish within
Asterisk. I hope to be passing code around for some serious testing by
other people in a couple of weeks, or so. Certainly within a month. Once
I have it stabilised with UDPTL and SIP I will get it working with IAX.
Then
[EMAIL PROTECTED] wrote:
Hi there,
Anybody on this list knows where I can obtain Hong Kong DID's from ?
Cheers,
Sahil
You get them when you subscribe to a T1 or E1. However, if you want
blocks bigger than 200-300 per T1/E1 it is a problem these days, unless
you are a telco.
Regards,
Steve
I just installed .NET 2.0 Beta, so I don't know
if it's a problem with your app or with new .NET,
but after a few minutes of running your app and not doing anything (the
app
runs idle)
a new window appears with a following message:
Timer2: Object reference not set to an instance of an
[EMAIL PROTECTED] is believed to have said:
Suse 9.2 uses udev. Look for README.udev in you zaptel source directory and
follow the instructions.
Regards,
Alex
Thanks Alex!
Aldo
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[EMAIL PROTECTED] is believed to have said:
I have a fairly current CVS build of asterisk running on SuSE 9.2. You
need to get rid of the stuff that gets installed with the system and
then install the zaptel stuff. Works fine for me, but I do get warnings
about unsupported modules and
[EMAIL PROTECTED] is believed to have said:
I've posted this question twice without a single reply. Does that mean no
one knows the answer, or no one cares to answer?
I've been having an issue with an IAX2 trunk setup in Asterisk. Setup the
trunk fine and it registers and works fine.
Hi,
Welcome.
Read the samples *.conf files
(in /etc/asterisk)
extension.conf, sip.conf are
some good places to start.
Read search the wiki.
Many info there (also not always very clear)
success
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201
Am Donnerstag 17 März 2005 12:26 schrieb J Thomas:
I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I
consistently get one of the following errors:
PRI got event: HDLC Abort (6) on Primary D-channel of span 1
or
PRI got event: HDLC Bad FCS (8) on Primary D-channel of
All the samples are on your system
/usr/src/asterisk/configs/ the files have a .sample on them.
Also there is allow of information on the
Wiki http://www.voip-info.org/wiki-Asterisk
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bharat M. Sarvan
Sent:
I've posted this question twice without a single reply. Does that mean no
one knows the answer, or no one cares to answer?
I've been having an issue with an IAX2 trunk setup in Asterisk. Setup the
trunk fine and it registers and works fine. I'm able to make outgoing calls
from any
Dear All
:
We need to use the
Conference Room Capability from Asterisk to use it with our IPT Solution which
based on Cisco Call Manager..
Also we need to use most
of Asterisk features in our IPT Network ..
How can I do this ? Any help will be grateful ..
Mohamed Farid ,,
On Thu, 2005-03-17 at 06:33 -0500, Doug Lytle wrote:
[EMAIL PROTECTED] wrote:
change the sntp_mode: from directedbroadcast (the
default) to unicast. This will cause the phone to poll
your NTP server. This solved the problem for me.
This fixed my problem as well! Thanks
Carrier Access generally have all of their manuals available for
download. You just have to request a free login. they also provide
excellent dialin support - also free. If your framing LED is blinking I
would double check that both ends of your span are set for ESF.
zttool is the tool for
Hello,
I need some help setting up statistics per call. I need to store in a
database call quality details such as jitter, packets lost and other
informations. Is there any way to do this?
I'd really appreciate some links or any other kind of info on this.
Thanks,
Calin.
On Wed, 16 Mar 2005 19:58:55 -0800, Luki [EMAIL PROTECTED] wrote:
Anyway, if anyone ever needs this info, they can Google it now :-).
Might be a good thing for the wiki too. ;)
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Hi there,
Newbie questions on ZAP channel numbering (forgive me if this was asked before):
1. How are channels numbered if I have multiple FXS/FXO cards in the
system? Is there a fixed mapping between PCI slot id and the number
range allocated for all the ports on that PCI card?
2. Same
Hello,
can anyone using astgui client i have a problem in installation phase
everytime i try to create database from MySQL_AST_CREATE_tables.sql it
gives error in phone table
ERROR 1064 (42000): You have an error in your SQL syntax; check the
manual that corresponds to your MySQL server version
Good Day List,
I am looking to see if anyone is willing to share their working
configs with me.
I would be happy to add to wiki and document steps to get it to
work with asterisk.
I am looking for both Welgate configs as well as sip.conf and
extension.conf snippets.
Hi Andreas,
it's impossible to use iptables due to the reason that audio flows through
asterisk and users
won't be able to communicate w/ *...
I've tried that.
Regards,
Pavel
- Original Message -
From: Andreas Sikkema [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Thanks you !
I'll try!
Regards,
Pavel
- Original Message -
From:
Rod Bacon
To: braincrew.com ; [EMAIL PROTECTED] ; asterisk-users@lists.digium.com
Sent: Thursday, March 17, 2005 3:24
AM
Subject: Re: [Serusers] ser+asterisk -
security
Do some reading
Roy Sigurd Karlsbakk wrote:
I tried wiki, but I got too many pages (I think all of them), ...as
answer.
I want to write an agi.
I need a HOW-TO, is there anything available?
see the perl agi package from http://asterisk.gnuinter.net/, the
agi/agi-test from the asterisk source and
Hello
I am using Astwind under Debian Linux as my first trial of Asterisk. Somehow
I managed to restart (!) Asterisk automatically when linux starts. (I did a
make install, do you think making asterisk, will do it?)
Ok, it is not a big deal, but, unfortunately, it stops with an exit code 127
Title: Comparing Callmanager to Asterisk
Callmanager does nothing than construct and tear down calls and the actual RTP stream does not flow through the Callmanager but is direct from IP device to IP device. How does this work with Asterisk? I read something that lead me to believe that
[EMAIL PROTECTED] wrote:
it's impossible to use iptables due to the reason that audio
flows through asterisk and users won't be able to communicate w/ *...
I was thinking of just the SIP port. I am assuming that asterisk
protects its RTP ports from processing traffic from a third party.
--
Hi
Im using Asterisk @ Home 0.6 running on VM ware virtual
machine. I have no interface card and have configured ztdummy as best I could.
I know that the usb timer is installed correctly on the machine.I changed the
following files to try getting it working:
/usr/src/zaptel/Makefile
hi
Any one give me any hint how to start radius with
asterisk.
Is there any addon available for asterisk+radius.
Please provide me helpfull link which could help me.
i am new to radius.
regrads
kamran
__
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Yahoo! Mail - Find what
That doesn't work. I was trying to do it yesterday, there is a patch
that fixes the problem. google for it or if ur too lazy:
http://lists.digium.com/pipermail/asterisk-users/2004-August/059709.html
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Hi List,
As most know, Chan_Spy and consequently, the MOH
patch that used Chan_Spy disappeared around version 1.0.2 (or so). I know
the native MOH patch works well and doesn't require the mpg123, which as proved
problematic, at least for me. However, I know of no method to "listen in"
or
[EMAIL PROTECTED] wrote:
I need to deploy some quasi-virtual-PBXes, and I'd like to avoid having to
be hands on for each new phone number deployed... so I would like to set
up some administrative extensions that can record greetings... lets say:
[admin]
exten =
Hi,
What do I need to do to get Asterisk to allow me to use codec G-726?
I've already tried allow=all in my sip.conf config.. didn't work...
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Hi
I have a Cisco 2801 with CME and I'm trying to connect it to Asterisk through
SIP. Can you please send me the Dial-peer configuration that creates a trunk
between the Cisco router and Asterisk.
Thank you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf
had the same thin with 729
I had to go
disallow=all
allow=g279
On Thu, 2005-03-17 at 16:37, Matt wrote:
Hi,
What do I need to do to get Asterisk to allow me to use codec G-726?
I've already tried allow=all in my sip.conf config.. didn't work...
___
Steve Underwood wrote:
The bigegst holdup has really been the poor state of T.38 support in
current equipment. Few ATAs do it. Fewer do it right.
Care to share which ones do it right? We purchased 2 ATA's all which
claim to do T38 and they don't. (Azatel, WorldAxx)
-Matthew
Kamran Ahmad wrote:
hi
Any one give me any hint how to start radius with
asterisk.
Is there any addon available for asterisk+radius.
Please provide me helpfull link which could help me.
i am new to radius.
regrads
kamran
If you are new to radius then I will suggest and highly recommend
Matthew Boehm wrote:
Steve Underwood wrote:
The bigegst holdup has really been the poor state of T.38 support in
current equipment. Few ATAs do it. Fewer do it right.
Care to share which ones do it right? We purchased 2 ATA's all which
claim to do T38 and they don't. (Azatel, WorldAxx)
Hello!
I have bin trying to set up trunking between some of my Asterisk
boxes but had no luck...
I use Asterisk 2.0.6 on SuSE 6.2, but I have had the same problem with
erlier releases. I have a working connection and can place
multiple calls in both direktions. Than I set trunk=yes on
both
I've got echo problems.
*** I'm looking for paid support. ***
I'll accept free support, but don't mind paying if someone really knows
what they are doing.
I've read the wiki, etc.
Played with the settings in zapata.conf
Using V400P
PSTN-_T1-_ASTERISK-_BROADVOICE-_PSTNECHO ON CALLED PHONE
hi
thanks all who helped me in making this success.
i am using latest asterisk from CVS.
asterisk-oh323-0.7.1,
pwlib-Janus_patch4-src-tar.gz,
openh323-Janus_patch4-src-tar.gz
GnuGatekeeper
it is working asterisk is routing calls to GNUGK
successfully
extensions.conf
exten =
On Thu, 17 Mar 2005 14:53:27 +0800, XinTai Wang
[EMAIL PROTECTED] wrote:
who have been fabricated their own cards from Tormenta 2 PCI Card?
govarion.com
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you need to get openssl-dev package too
for most dependencies problems you need respecitive dev libriaries
regards
m.
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Gilbert Abboud wrote:
I have a Cisco 2801 with CME and I'm trying to connect it to Asterisk
through SIP. Can you please send me the Dial-peer configuration that
creates a trunk between the Cisco router and Asterisk.
You can try something like this:
dial-peer voice 900 voip
Steve Underwood wrote:
Matthew Boehm wrote:
Steve Underwood wrote:
The bigegst holdup has really been the poor state of T.38 support in
current equipment. Few ATAs do it. Fewer do it right.
Care to share which ones do it right? We purchased 2 ATA's all
which claim to do T38 and
Thank you John,
Max Blackmer
I would like to create an Intercom extension that will dial a group of
extensions which are connected to SIP phones. The SIP phones are setup
to auto answer a particular extension assigned to one of the lines in
the phone. All phones must answer and broadcast
has session border control been added to asterisk yet? i remember hearing
about it, but i haven't been able to find any information on it on wiki.
Thanks,
daniel
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Upgrade to kernel 2.6.9, there are supposed to be significant bugfixes for
CAPI support in 2.6.9.
All of my CAPI systems use FC2, 2.6.9. I tried to go 2.6.10 but had
problems.
Craig
- Original Message -
From: Kib Eki [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
I have tried everything to get BV working outbound. All worked fine
until the BV change last week. I called BV and they changed me to sip
gen with a new password. I stripped my Asterisk server to one phone on
Zap/1 until I get this working. The same BV account works fine with a
SPA-3000 so I
i have written app for billing with asterisk. what is
the problem in using radius.
kamran
__
Do you Yahoo!?
Yahoo! Small Business - Try our new resources site!
http://smallbusiness.yahoo.com/resources/
As you can see from the SIP trace below (from the called phone),
intercom=true is being appended to the To: header as per requirements.
The intercom=true needs to be appended to the request URI, not to the
header as a whole -- your To: header should be:
To: sip:1011 at
As you can see from the SIP trace below (from the called phone),
intercom=true is being appended to the To: header as per requirements.
The intercom=true needs to be appended to the request URI, not to the
header as a whole -- your To: header should be:
To: sip:1011 at
Daniel,
Did you get much progress made on this?
I'm new to asterisk - but we are a heavily invested LDAP shop and
if I can demo an initial install that pulls telephony configs from LDAP
it would really be nifty.
I'd be happy to help in any way I can
- I'm not much of a developer - but have some
Am Donnerstag 17 März 2005 17:49 schrieb John B Dunning:
Daniel,
Did you get much progress made on this? I'm new to asterisk - but we are
a heavily invested LDAP shop and if I can demo an initial install that
pulls telephony configs from LDAP it would really be nifty.
I'd be happy to help
Hi List
I've been using Asterisk for quite some time with no major problems, but
I've been facing this bug from the beginning and now I want to see if that
is fixable.
We have a provider who terminates our USA LD traffic and the problem comes
when relaying the caller ID I send them from my
Kamran Ahmad wrote:
i have written app for billing with asterisk. what is
the problem in using radius.
kamran
Its a pain and redundant. Why run two seperate databases when 1 will do
what you need? There is no native radius support for Asterisk. There is an
addon, (search the wiki) but
Anyone out there using NetLogic DIDs? And have inbound working? I got
outbound working, but no joy so far with inbound. Here are the relevant
parts from my conf files:
iax.conf
[general]
tos=lowdelay
jitterbuffer=no
register = username:[EMAIL PROTECTED]
[netlogic]
type=friend
host=dynamic
Hi all,
I am working on building a new VoIP PBX. Looking at the current market
for phones it seems my best enterprise options are the Cisco and
Polycom phones. I have some experiance with the Cisco 7940G, but the
process of flashing the phone with the SIP firmware left a bad taste in
my mouth
Anybody knows how to patch the music on hold bug on a
bristuffed-0.2.0-RC7j 1.0.6-asterisk ?
Thanks
maxx
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On Thu, Mar 17, 2005 at 11:59:06AM -0500, Oswaldo Arratia wrote:
I send a call with valid caller ID info (areacode+number); my provider gets
the call and routes it properly, the end receiver gets the call and does not
see the caller ID I sent, they just get 'Unknown Number'.
This remains
Thanks for the pointer, makes sense.
Although I am not using zaptel, I am sending the calls via SIP to my Cisco
AS5300 which is connected via PRI to my provider and it happens to be set
like NI2.
I will test and will let you know.
Thanks!!!
Oswaldo
-Original Message-
From: [EMAIL
Hi,
I've got an Asterisk latest CVS head with oh323 installed. There is one
thing I can't understand about the codec negociation. I receive calls in
G723G729, and send them to another gateway who can handle both codecs
too. So all I want to do is just passthrou, for both. It seems that *
only
Oh this is sad.. I'm familiar with radius.. and was hoping to be able
to use asterisk with freeradius to be able to do call accounting and
billing.. so you're telling me this is now not a good idea?
Am I better off (for now) parsing the csv report each month?
On Thu, 17 Mar 2005 11:00:09 -0600,
I think Cisco VoIP phones are absolute works of art. The first time I
saw one, I wanted them,
That being said, I use Polycom IP 500s and I absolutely love them.
The speakerphone is excellent, configs are pretty simple once you know
what you are doing with them, and the phone is very
If you don't want to proxy the media through * the put this setting:
canreinvite=yes
this will allow the 2 end points to connect directly for the RTP
bypassing
you. otherwise I have noticed the same when I try to proxy I have to
make sure everyone is using the same codec or it doesn't work
The lack of full SIP suport and the cost of Ciscos license plus the
added base cost of their phones moved us away from Cisco and over to
Polycom. They have been working extremely well. Software updates are
free and the update process is relatively simple. I have found the
IP600 is a great desk
I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines
connected via TE405P. Everything works great, except MOH. I added an
exten with MusicOnHold(30), and it plays just fine. Conferences have
music when no one is in. I have SIP phones. When I place a call on hold,
the CLI give no
What would be a minimum sound card/microphone combo for good voice
quality recording on a budget? This would be for * voice prompts.
Would a soundblaster live and a good mic do the job?
Chris
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I did it and it worked. The problem was the national plan!
Thank you very much for your tip.
For those who run into this, here is the configuration of the voice port in
a Cisco AS5300 series:
interface Serial2:23
no ip address
isdn switch-type primary-ni
isdn incoming-voice modem
isdn map
On Fri, 21 Jan 2005, Greg Boehnlein wrote:
Hello,
I've got a mixture of SPIP 300 and 500 phones in production for
various clients. I've got the XML settings configured for local
conferencing, but I'm not seeing the expected behavior from the phone when
I attempt to conference two
I read about this option. But does it work on a h323 channel ?
(inAccessnetwork's one)
Brian C. Fertig wrote:
If you don't want to proxy the media through * the put this setting:
canreinvite=yes
this will allow the 2 end points to connect directly for the RTP
bypassing
you. otherwise I have
Hi:
I had asterisk with RealTime database working perfectly in a RH 9.0
machine. I used the sip cache so I even had MWI working. The problem
is that I decided to move to Fedora Core 3. I installed the lastets cvs
version of asterisk and the RealTime addon from asterisk-addons. I at
Not sure the rquirements for your receptionist. I have found that the
IP600 does have most everything required to function properly. If you
do have an office without DID and a lot of traffic then you may want to
look at the tools to display status on her computer. I do have a Snom
inhouse for
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