Hi,
Perhaps there's something wrong in my config...
I did some tests connecting Asterisk to an Ericsson MD110 PBX by setting
up an h323 trunk. When dialling into asterisk I got some problems when
the entire number is not in the setup message, i.e. I'm dialling digit
by digit on the ericsson
It liiks like a motherboard problem. It's failing the initial boot.
You say it booted again after two hours. Was the machine powered down
during that interval. If so, I suspect you have a temperature problem.
I've seen very similar problems from a defective CPU fan.
Bill
On 5/30/05, Ronald
hello all,
now, i want to do configuration to make sip client
have extension on my asterisk.but i have a problem
with registration of sip client.
*CLI May 31 13:58:01 WARNING[4927]: chan_sip.c:886
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for
seqno 115 (Critical Request)
Hello,
I'm setting up AST Post Paid application, is there
anybody who set up astpp ?
I followed the directions, i visited
the astpp admin page in my web browser. But i couldn't setup the brands and
routes etc. "Database unavailable -- please check configuration" appeared on the
top of the
Have you updated with the lastest firmware..
It now does an on-hook forward to asterisk
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop
Sent: Tuesday, 31 May 2005 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Ronald Wiplinger wrote:
One of our remote user's phone reports frequently:
Got SIP response 481 Call Leg/Transaction Does Not Exist back from IP
What can I do ???
Turn on SIP debug, set verbose to 4, debug level to 4 and trace what
happens. If we can't see that, an error message out of
Tim P wrote:
I have multiple Sipura ATA 2100s attached to normal analog phones that
are all configured as extensions in *
When I call an extension it rings and will go to voicemail if no one answers
it.
When I call the same extension a second time after no answer (went to
VM) the phone
Manjit Riat wrote:
Hi,
I prevoiusly has asterisk on a public static ip and had a phone from
a different location registering to the asterisk box. But now we have
dropped the previous connection and the current connection has a
dynamic ip. Is there any way for the phone to register to
Hi-
How to configure MGCP in asterisk.
I want to connect my asterisk to MGC gateway.
Best Regards
Ibrar Ahmed
Project Manager.
Comcept (Pvt) Ltd. Islamabad Pakistan
www.com-cept.com
[EMAIL PROTECTED]
[EMAIL PROTECTED]
Ph # (Off) +92-51-111784784
Ph # (Res) +92-51-2271283
Ph # (Mob)
Hi!
I'm trying to build gnugk with asterisk. Asterisk is working well with chan_h323
built with needed PWlib v.1.5.2 and open H.323 v.1.12.2.
But gnugk' s installing instructions says that I need latest PWlib(1.17.1) and
openh323 to get gnugk work. Now, with installed pwlib and openh323 gnugk's
I am doing some testing using FOP (Flask Operator Panel) and so far, its
going great! Been able to do callerid and also open a SugarCRM screen.
All without having to install anything on the computer, just open a FOP
browser screen and that's it!
More later when I debug some ideas.
In article [EMAIL PROTECTED],
Anton Krall [EMAIL PROTECTED] wrote:
I just installed the latest cvs head and seems a lot of commands haven been
depricated.
Where can I see the changes on all cvs head versions in order to keep up
with the changes needed on my side.
I checked the wiki and it
In article [EMAIL PROTECTED],
Tom Fanning [EMAIL PROTECTED] wrote:
Hi
Can anyone provide me with a Manchester (0161) UK DID number, preferably
IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume
will be low.
The critical thing is that DTMF must be correctly passed
Yeah tried it.
Unfortunately I need this feature in reverse. I need the call to stay
on hook when going from Asterisk to Sipura. Staying onhook from Sipura
to Asterisk workd fine.
On 5/31/05, David Phelan [EMAIL PROTECTED] wrote:
Have you updated with the lastest firmware..
It now does an
Good day all
I remember some time ago I tried recording on asterisk
But it did not work because the sox app was broken and by downloading a
older one it worked
Now things have come and go and version change
What sox version will work with asterisk 1.0.7
Thanks
Altus
In article [EMAIL PROTECTED],
Tim Connolly [EMAIL PROTECTED] wrote:
I recently began using the curl cmd to do an external callerid
lookup on my own customer database. I've noticed certain lookups will cause
a crash and not show anything in the messages file or the console.
It is failed
Hello All
I'm using asterisk 1.1.X and MFCR2 lib version 0.03pre2. when i call to E1 (connected with asterisk), chan_unicall don't detected event incoming call and show error.
error messages:*CLI Warning, flexibel rate not heavily tested!Rx CAS bits 0x9 [ 1/ 0/ 0]Line unblocked -- R2 Channel
I am looking for a provider that accepts BYOD that has good rates to UK
NCFA (+44 0870 ..._. If anyone knows of a provider that they use that
has reliable service I would greatly appreciate hearing from it. Feel
free to reply private since this isnt directly asterisk related.
--
Trixter
I have an Asterisk PBX behind a manually-built
IPCHAINS firewall machine. Can anyone tell me what I need to allow/build
QOS packet rewrites through this simple NAT barrier? What do I need to
pass to IPCHAINS to let QOS out to the next outside network hop?
I ask this, because I have
Slightly OT, but I think this is of possible interest to many of you,
I need to get a UPS for my asterisk box. They are rated in VA but I
can't quite figure out how that converts to real life.
I have a PIII-800 box with two X100P and one TDM400P plus graphics
adapter, an IDE hard drive etc. Will
You can either download the executable version of gnugk or you can reinstall
the other versions of the pwlib and openh323 as they are only needed during
the compile.
RG,
Gentian
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 31,
Hi ;
Have two handytone 486 and want to use them as digium TDM400
fxo-fxs card...
I mean is it possible to direct pstn calls from astersik (extensions)
to handytone line port directly and
vice versa ?...
Thanks in advance
Betul
Onemli not : Bu e-mail iletisi, sadece adreste
Title: Asterisk install error ...
Hi;
It is my first time to use asterisk I have TDM400 wildcard and 4 FXO Modules when I install asterisk an error occurred
Chen_zap.c 2772 : error : zt_event_dtmfdigit undeclared
Can any body help why this error ..
Thanks;
Ghassan M. Lama'
Title: Asterisk install error ...
Hi;
Thanks for replay;
I have used the latest CVS and the stable version .
I am installing the software on Fedora core 2 Kerenl 2.6
I do have zaptel instaled and configured
Regrds;
___
hi, I am new to asterisk
I have a client who wants a help desk type of application. the asterisk
tool kit seems to fit the bill nicely. is there anything already implemented
that is available or is all the asterisk implementations custom?
attributes of the project include a sequence of
I've been looking out for the Uniden UIP1868 for a while now, but I
haven't seen it anwhere that I'm used to buying things from. According
to froogle, a couple of places (that I've never heard of) have a small
number in stock (small = 10 in this case). I'm doubly suspicious
because even
Hi,
Let me try to answer this one.
Assuming your P3-800 is using a 300watt power supply, then in a full
load condition, convert to VA, it will be 300/0.6=500VA. So, it is
greater than your small 400VA box. So, you need a bigger ups. Of course,
if your power usage is actually much lower than
Dyndns.org seems like a good choice. Just make sure you put in the
hostname in the phone configuration, not the IP address :-)
Also, when the ip changes, users will usually need to reboot their
phones. I added a mail alert that sends a heads up to users and also
some stuff to reprovision the
GS phones don't need to reboot.
On Tue, 2005-05-31 at 10:58 +0200, Wilson Pickett wrote:
Dyndns.org seems like a good choice. Just make sure you put in the
hostname in the phone configuration, not the IP address :-)
Also, when the ip changes, users will usually need to reboot their
Another thing to consider regarding the ups is the runtime, depending
on the hours and minutes you want the ups to supply power to your
asterisk box, you may need to add more batteries to the ups.
Regarding this, I have done this hack yesterday:
- Remove the battery from an existing UPS
-
Assuming your P3-800 is using a 300watt power supply, then in a full
load condition, convert to VA, it will be 300/0.6=500VA. So, it is
Thanks for that info. Where does the /0.6 come from? I've always
wondered about VA which looks like VoltAmps.
There are 400, 500 and 600VA models. The
On Tue, 2005-05-31 at 13:22 +0400, Jean-Michel Hiver wrote:
Another thing to consider regarding the ups is the runtime, depending
on the hours and minutes you want the ups to supply power to your
asterisk box, you may need to add more batteries to the ups.
Regarding this, I have done
Tzafrir,
We need to send an email with the fax number for astfax to fax.
eg:
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: ...
attach fax image
How to configure postfix to understand and deliver this?
I've tried putting this line in /etc/aliases :
fax:
I have a very simple question .
I have 2 internal extension 301 and 300 sip phone . I want to these extesioncan call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal.
How can I do that ?
[x1]exten = 300,1,Dial(SIP/300)
include = pstnlocal
[x2]exten =
I'm trying to setup DATA calls with Dial(Zap/g1d/12345678), but with PRI
DEBUG SPAN 1 on, it seems to connect a regular SPEECH call.
I'm using 1.0.6. Is this feature broken in stable release?
There seems to be support in the source, but it doesn't work.
Does the Telco set what each PRI channel
Normally, the power factor is taken as 0.6, thus to convert watt to va,
just divid the wattage by 0.6 to get the va rating.
cheer
Wilson Pickett wrote:
Assuming your P3-800 is using a 300watt power supply, then in a full
load condition, convert to VA, it will be 300/0.6=500VA. So, it is
Hi,
you will need app_settransfercapability to make this work properly. This
is part of CVS-HEAD. I have backported it for the asterisk stable
version of bristuff (see www.junghanns.net/asterisk/) and also fixed
some bugs in Asterisk that will make ISDN data calls unreliable (or in
some cases
On Tue, May 31, 2005 at 05:38:55PM +0800, Eddie wrote:
Tzafrir,
We need to send an email with the fax number for astfax to fax.
eg:
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: ...
attach fax image
How to configure postfix to understand and deliver this?
I've tried
Title: Asterisk compailation Error Chan_zap.c
Hi;
It is my first time installing an asterisk PBX system I do have a TDM400 wildcard with 4 FXO moduls on a PC with 3.0GHZ HT CPU and INTEL 915 moatherboard
Fedora C2 Linux as O.S. and I have the latest CVS astreisk , Zaptel and Libpri
Another thing to consider regarding the ups is the runtime, depending on
the hours and minutes you want the ups to supply power to your asterisk
box, you may need to add more batteries to the ups.
No worry there, since the modems (upstairs) will be unpowered as well.
Although the asterisk
Betl Gzlkolu wrote:
Hi ;
Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card...
I mean is it possible to direct pstn calls from astersik (extensions) to
handytone line port directly and
vice versa ?...
On my 486 I can't dial out on the FXO port, it's just a
asterisk asterisk wrote:
I have a very simple question .
I have 2 internal extension 301 and 300 sip phone . I want to these
extesion can call each other, and ext 300 can call outside to pstn, and
ext 301 to call internatonal.
How can I do that ?
You read the samples and the guides
asterisk asterisk wrote:
I have a very simple question .
I have 2 internal extension 301 and 300 sip phone . I want to these
extesion can call each other, and ext 300 can call outside to pstn,
and ext 301 to call internatonal.
How can I do that ?
include pstnlocal at either [x2] or
Hi,
I'm trying to configure Sipura 2000 (behind NAT) which connects to
Asterisk (public IP, no NAT) and having interesting results. When Sipura
is behind Linux/NAT firewall it works great and no special NAT settings
on Sipura are necessary. The issue I'm having is when Sipura is behind
Hi,
I am struggling with a problem where calls are being created on ZAP
channel, but no call exists.
I have 2 X100P cards 1st = BT, 2nd Telewest, with no problems on BT
line.
I have carried out various test on signalling types, Kewlstart,
Loopstart, Groundstart and EM. Only Kewlstart and
Dear All,
I have installed Asterisk everything is OK until I tried to configure
meeting room, configuration was simple enough when I try I get a message
that it's not a valid meeting room, Now I don't have a Zaptel device on
my machine, so I found that you will have to use ztdummy to make
a
I'm trying to configure Sipura 2000 (behind NAT) which connects to
Asterisk (public IP, no NAT) and having interesting results. When Sipura
is behind Linux/NAT firewall it works great and no special NAT settings
on Sipura are necessary. The issue I'm having is when Sipura is behind
You can build it alone.
Then try to 'modprobe zaptel' and then 'modprobe
ztdummy'
If you do this without errors it must
work.
For more info read the wiki.
RG,
Gentian
- Original Message -
From:
Mohamed A. Gombolaty
To: asterisk-users@lists.digium.com
Sent: Tuesday,
Hi,
I am looking for a way to let * choose the voice codec relying to the
used communication channel.
Example
I am using a Polycom 500 which supports G729 and G.711.
When I am doing internal calls (with my LAN) or calls over the PSTN
(ISDN) I want to use the G.711 codec because there is
Dear Ghassan,
I never used fedora but in the link below you will find a step by step
installation for fedora platform check it out and see if you are missing
anything.
http://www.voip-info.org/wiki-Asterisk+Linux+Fedora
Thx
MAG
Ghassan Lama wrote:
Hi;
It is
my first time installing an
I have canreinvite=no already, below is my sip.conf entry.
[1360]
username=1360
callerid=Phone 1 1360
secret=mysec1
host=dynamic
auth=md5
qualify=1000
dtmfmode=rfc2833
context=from-sip-unrestricted
mailbox=1360
type=friend
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g726
nat=yes
Hello list,
I put together a quick note about how to see oh323 calls while they are
handled by your * box.
http://www.oinko.net/astrecipes/index.php?n=89
The article is just a draft with usage examples; I'd love to hear your
comments and updates if there is something I got wrong.
Thanks
FYI
If using oh323 v0.6.5 then the oh323 show info has been replaced by
the command oh323 show channels
Thanks
Giles
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lenz
Sent: 31 May 2005 12:51
To: asterisk-users@lists.digium.com
Subject:
Yes Pstn local start with 9 and pstn international starts with 00 .That is ok.
I can make call form 300 to pstn local, and from ext 301 to pstn international, that is ok .But in this exemple I can not call formext 300 to 301 and form 301 to 300.
It is possibile to have2 diferent group of
you can try use variable preffered_codec in dial command (if you now
the prefixes/dial numbers, for which to use eg. g729)...
PJ
Kib Eki wrote:
Hi,
I am looking for a way to let * choose the voice codec relying to the
used communication channel.
Example
I am using a Polycom 500 which
Nardis Dome wrote:
try eyeBeam, it works fine for me...
[]
type=friend
secret=
auth=md5
callerid=myCallerId
canreinvite=no
host=dynamic
disallow=all
context=default
allow=alaw
allow=ulaw
allow=speex
allow=gsm
allow=h261
allow=h263
Thanks, I bought eyeBeam for two computers
Eric
A completly off topic response (and not even a response in that I'm
asking you a question - sorry)
you say that you have several 3000 devices and you show your dial string as :
Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Is the sipura1 section referencing a single sipura or the group of
several.
could you please give more information concerning this setting?
Pavel Jezek wrote:
you can try use variable preffered_codec in dial command (if you now
the prefixes/dial numbers, for which to use eg. g729)...
PJ
Kib Eki wrote:
Hi,
I am looking for a way to let * choose the voice
Hi,
did you enable the right video-codecs in eyeBeam?
settings-media-video-Advanced-Codecs
--- Ronald Wiplinger [EMAIL PROTECTED] wrote:
Nardis Dome wrote:
try eyeBeam, it works fine for me...
[]
type=friend
secret=
auth=md5
callerid=myCallerId
canreinvite=no
How would one go about having asterisk insert a faint 'beep' once every
ten seconds or so whilst a call is being recorded? I can't see any flags
in the Monitor appliction for doing so.
This is a legal requirement in many jurisdictions when calls are being
recorded (as an alternative to an
Hallo,
we have started playing with asterisk about one month ago, and we do like
very much what we are experiencing.
Now we would like to take some step further towards standardizing
installed modules, functionalities, tools etc.
The wall we are facing now is: choosing the right tool for *
Nardis Dome wrote:
Hi,
did you enable the right video-codecs in eyeBeam?
settings-media-video-Advanced-Codecs
I have here
1. H.263++QCIF 128
2. H.263+
3. Basic H.263
and in asterisk
allow = 'ulaw;alaw;speex;gsm;h263;h263p'
--- Ronald Wiplinger [EMAIL PROTECTED] wrote:
Nardis Dome
There is nothing wrong with your config, it is just unimplemented
functionality.
Michael.
Alexander Topolanek wrote:
Hi,
Perhaps there's something wrong in my config...
I did some tests connecting Asterisk to an Ericsson MD110 PBX by setting
up an h323 trunk. When dialling into asterisk I
On Tue, 2005-05-31 at 15:08 +0200, [EMAIL PROTECTED] wrote:
Hallo,
we have started playing with asterisk about one month ago, and we do like
very much what we are experiencing.
Now we would like to take some step further towards standardizing
installed modules, functionalities, tools etc.
This is off the top of my head - never tested
For the end user device (ie polycom in your case) your sip settings
would be something like
[5000]
username=5000
SNIP
deny=all
allow=ulaw
allow=alaw
allow=G729
which would give you both
Then if you in the Trunk set the following
[Trunkroute]
Hi,
I'm a newbie on Asterisk and I'd like to know if it's
possible to connect two or more asterisk together.
In fact, I'd like install and connect some asterisk
together.
Thanks for advance,
Cyril
[EMAIL PROTECTED] wrote:
We tried AMP, very powerful but incomplete (CAPI is very important to us);
The 1.10.008 version of AMP supports Custom Trunks. Text from the AMP
tooltip:
-begin-
Define the custom Dial String. Include the token $OUTNUM$ wherever the
number to dial should go.
[EMAIL PROTECTED] wrote:
Hallo,
we have started playing with asterisk about one month ago, and we do like
very much what we are experiencing.
Now we would like to take some step further towards standardizing
installed modules, functionalities, tools etc.
The wall we are facing now is:
I've google'd this to death, is there a simple way to make MWI work from *
for my Cisco phone ??? Examples ???
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
Yes, via IAX
Adam Collard
General Manager, ER Wireless
(800) 757-5669 x4861
(810) 496-0161 Fax
(517) 242-1800 Cell
Nextel DC 131*256784*19
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cyril SIMON
Sent: Tuesday, May 31, 2005 1:29 AM
On 31/05/05, cyril SIMON [EMAIL PROTECTED] wrote:
Hi,
I'm a newbie on Asterisk and I'd like to know if it's
possible to connect two or more asterisk together.
In fact, I'd like install and connect some asterisk
together.
As usual, Google and the Wiki are there for your convenience.
I am connecting to a local nortel PBX. The person setting up the PBX did
not
define a user name and password (dont ask why). So I presume my register
command
can leave off the username and password and look like:
[sip.conf]
register localpbx/5551212
I presume that is good enough then so calls
Problem 1 - Outgoing:
I am able to call out of the * box using the analog line attached to
the sipura 3000 but when the person being called answers there is no
audio from either end. * registers that the call was answered but
passes no audio.
Problem 2 - Incoming:
When calling into the 3000
The most common problem is that the web server does not have permission
to write to the config files in /var/lib/astpp. Try changing their
ownership to be the same as the web server owner.
Darren Wiebe
[EMAIL PROTECTED]
Erdem HAKI wrote:
Hello,
/I'm setting up AST Post Paid application,
Ronald Wiplinger wrote:
Nardis Dome wrote:
Hi,
did you enable the right video-codecs in eyeBeam?
settings-media-video-Advanced-Codecs
I have here
1. H.263++QCIF 128
2. H.263+
3. Basic H.263
Try 261?
--
Cheers,
Matt Riddell
___
I already have the Cisco 7960 and am thinking of getting a Polycom 501 or
600. Anyone got these? Quality? One phone I really like is the Mitel
5240 but unfortunately there is no SIP image available for them. Anyone
know of a phone similar in looks/functions that works with * ?
Thanks in
The chan_sccp page at
http://www.voip-info.org/tiki-index.php?page=chan_sccp2 has been
updated.
See the bottom of the page.
Thanks.
Comments welcome.
--
respectfully, Joseph ===
-= ** =
___
Asterisk-Users
Hi Gavin,
I installed the cvs Asterisk CVS-D2005.05.28.22.00.00-05/31/05-14:25:23 and i
added this rowd in the features.conf
[featuremap]
blindxfer = #1; Blind transfer
disconnect = *0 ; Disconnect
automon = *1 ; One Touch Record
atxfer = *22
Dustin Wildes wrote:
I feel there is nothing wrong with having a web-based configuration
utility, if set up correctly. Look at the WRT54G Linksys router, plus
other countless devices that use an embedded browser for configurations.
Just a nitpick, if I may. They have embedded http servers,
Has anyone seen a situation where, upon connecting two asterisk servers
together with IAX registration, outgoing/incoming calls that route through
both servers are choppy and jittery? I don't have this problem when I call
out to teliax (my ITSP) directly, but if I try to make the call through the
so what should astpp db be exactly, where can i find its name? what
should i write there?
Thanks again..
The Database field should contain the name of the astpp db, something
along the lines of astpp is what I would put in there. Here is a fixed
version of the script. It did not post
Chris Mason (Lists) wrote:
When I reload the config, I see this error in the CLI. However, I don't see
what I have done wrong:
== Parsing '/etc/asterisk/zapata.conf': Found
May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring
signalling
-- Reconfigured channel 1, FXO
Sorry, no support for rates with time limits yet. You can file a bug @
http://www.aleph-com.net/astpp/ if you wish.
Darren Wiebe
[EMAIL PROTECTED]
Erik Versaevel - Infopact Netwerkdiensten BV wrote:
What happens if the rate changes mid call?
IE, call starts @ 18.30 and lasts till 19.15
Rate
[EMAIL PROTECTED] wrote:
I've google'd this to death, is there a simple way to make MWI work from *
for my Cisco phone ??? Examples ???
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
What is it you feel is missing in AMP?
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Tuesday, 31 May 2005 9:09 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Tools for effectively
Did you install php-pgsql?
Check if the register_global is On in php.ini file (reload apache)
Regards, A.
On 5/31/05, Alexandre Charles [EMAIL PROTECTED] wrote:
Hi Everybody,
I have tried to make AreskiCCV2 work on RH9.0 but it does not work.
More precisely, I have followed the guide as
[EMAIL PROTECTED] uttered the following thing:
I've google'd this to death, is there a simple way to make MWI work from *
for my Cisco phone ??? Examples ???
Message waiting? Sure...
If you're using SIP, then it will work as long as you have the right
'mailbox=' line in your sip peer config.
So, I'm wondering does anyone have real-life
comparisons on the failure rate of a PC compared to the failure rate of
some of these options??
Obviously, an embedded PC or something that is designed such as a Sokeris is
made to last a *long* time, but in my experience, a Tier 1 PC (older Compaq,
Some ISP's provide a static hostname on a dynamic host, which you can use to
your advantage. Ask them if it is possible. For example, up where I am an
extremely large ISP is Telus Communications. They require you to register
the host's MAC address with an online tool and when you do, the tool
Jason,
thanks a lot for the info.
Is there any way to separate AMP stuff from asterisk, in other words to
have AMP, apache and so on on a different pbx than asterisk?
Tia brgds
Francesco Pellegrini
Frame srl
[EMAIL PROTECTED]
in your sip.conf:
[general]
videosupport=yes ;
in your eyeBeam settings- try to enable all the h.263
codec.
hope it helps...
--- Ronald Wiplinger [EMAIL PROTECTED] wrote:
Nardis Dome wrote:
Hi,
did you enable the right video-codecs in eyeBeam?
settings-media-video-Advanced-Codecs
On Tuesday 31 May 2005 14:41, Giordano Grandis wrote:
Hi Gavin,
But...how atxfer work ?
Ehm, just the way I explained yesterday :) Just make sure you include the 't'
option to the Dial application, in the same way you need for the old-style
'#' blind-transfer to function.
gdh
Just got it working with eyebeam:
in sip.conf under general:
videosupport=yes
allow=h261
allow=h263
shouldn't need per phone config.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matt Riddell
Sent: Tuesday, May 31, 2005 9:43 AM
To: Asterisk
Has anyone seen a situation where, upon connecting two
asterisk servers
together with IAX registration, outgoing/incoming calls that
route through
both servers are choppy and jittery? I don't have this
problem when I call
out to teliax (my ITSP) directly, but if I try to make the
Works for me, make sure you're not sending the voicemail to an e-mail
account, no point in setting the MWI in that instance.
Here's my Voicemail.conf...
format=wav
serveremail=asterisk
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
sendvoicemail=yes
[zonemessages]
I have a PIII-800 box with two X100P and one TDM400P plus graphics
adapter, an IDE hard drive etc. Will a small 400VA box be enough for
this?
It's tricky sizing UPS'es to be bang on the money. The rule-of-thumb
calculation for VA is watts/.6 . So, for a 200 watt power supply / .6 is 333
VA.
I am currently running stable, CVS-v1-0-05/25/05-12:07:15, with Polycom
SIP phones, running 1.4.1.
Too many of our transfers using the Transfer end up with zombie channels
after a REFER. As such, I implemented # transfers, and all is well.
Sort of.
I have a reproducible issue. Take a call from
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jean-Michel Hiver
Sent: Tuesday, May 31, 2005 5:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] UPS rating for SOHO asterisk box
[...]
Regarding
Peter - I speak with the folks at Uniden regularly, the UIP1868
currently has an ETA of late June, although I expect it might be into
July before these are widely available. Unless there are eval units
floating around, to my knowledge, these are not available in the channel
yet.
Cory
I've read the Wiki on using asterisk's built-in transfer options (#8 and
#6). They work fine but how does one cancle an attended transfer? Example: I
have person on phone, I hit #6 to being att-transfer. I enter Sally's
extension. I let it ring for a few seconds. Sally never picks up but her
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