Hi All
From wherewe can get the data for
1) ASR on various countries
2) Average Call drop on VoIP
3) Average Call Quality
This we require to get an idea of what types of
problem normally users use to face on voip and what is the average percentage of
those problems.
Pls. help me if
On Mon, 30 May 2005, Mike Price wrote:
On Thu, 2005-05-26 at 23:21, Armin Schindler wrote:
On Thu, 26 May 2005, Mike Price wrote:
Yes libcapi is installed. Here is a sample of the errors I am getting:
In file included from chan_capi.c:38:
chan_capi_pvt.h:92: syntax error before
Hmmmyes but last time I played with my FXO module on the TDM400 could
not detect hangup properly (that is on a London BT line). Has this been
fixed? I keep an eye on the CVS but I have not seen any fixes for that.
Maybe I missed it.
Vassilis
Well, the official line is as Mr. Spencer
On Mon, 2005-05-30 at 18:21, Armin Schindler wrote:
On Mon, 30 May 2005, Mike Price wrote:
On Thu, 2005-05-26 at 23:21, Armin Schindler wrote:
On Thu, 26 May 2005, Mike Price wrote:
Yes libcapi is installed. Here is a sample of the errors I am getting:
In file included from
Astcc works fine with me
--- Rodrigo Otavio de Fraga
[EMAIL PROTECTED] wrote:
Hi,
I liked to have a pre paid card in my asterisk
Server.
I saw some applications in the voip-ifo site, but
all are not complete.
Somebody has some tested and functioning solution ?
Ok, I have solved my problems by upgrading my
chan_misdn and downgrading my mISDNuser. Now I have
asterisk working with mISDN support.
My problem now is that no matter what I do always see
the link down.
I've plugged the BN8S0 adapter to get the 8 ports
working. When I plug to the ISDN box
On Mon, 30 May 2005, Mike Price wrote:
On Mon, 2005-05-30 at 18:21, Armin Schindler wrote:
On Mon, 30 May 2005, Mike Price wrote:
On Thu, 2005-05-26 at 23:21, Armin Schindler wrote:
On Thu, 26 May 2005, Mike Price wrote:
Yes libcapi is installed. Here is a sample of the errors I am
On Mon, May 30, 2005 at 09:03:29AM +1000, Gonzalo Servat wrote:
On 5/29/05, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, May 28, 2005 at 06:34:23AM +1000, Gonzalo Servat wrote:
[snip]
If Asterisk allowed me to configure up to 10 ringing patterns, I could
probably cover most of the
Hi, I use asterisk-1.0.3 and mysql 4 on redhat 9.0. Normal call is OK, but when I use voicemail, * crash.Any one can help me?When voicemail app runs, * crash and no more trace. -- Executing VoiceMail("SIP/222.44.32.92-08155eb8", "u51292029") in new stackvmfax1*CLI> Disconnected from Asterisk
On Mon, 2005-05-30 at 19:27, Armin Schindler wrote:
On Mon, 30 May 2005, Mike Price wrote:
On Mon, 2005-05-30 at 18:21, Armin Schindler wrote:
On Mon, 30 May 2005, Mike Price wrote:
On Thu, 2005-05-26 at 23:21, Armin Schindler wrote:
On Thu, 26 May 2005, Mike Price wrote:
Yes
Hello *'s,
I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone
integrate it with asterisk if anyone what is the scenerio? i have a
scenerio which is quite simple but i am confused about it whether it
is possible or not :
I integrate it with asterisk for intranet no PSTN at all
Can any one tell me what the mysql password, no its
not password.
[EMAIL PROTECTED] root]# mysql --user=root -p
Enter password:
Thx
Q
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Hi, I have my asterisk server inside a NAT.
When i connect a softphone SIP inside my net, all go well. Ok.
But when I try connect a SIP softphone client outside my NAT, I get:
NOTICE[6087]: chan_sip.c:7691 handle_request: Registration from 'Alex
sip:[EMAIL PROTECTED]' failed for '83.41.119.25'
Are you doing port forwarding on your firewall?
Just make sure your asterisk port is open...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Piqueras
Sent: 30 May 2005 10:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Problem
Hi,
Does anybody know how often does IAX2 registration happen ?
Also I'm getting a feeling that there is no way of changing it through
an iax.conf file ?
Thanks,
Ivan
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I'm running Postfix as my email server. When comes to configuring
Asterisk Fax, Inter7 recommends QMail. Does anybody knows postfix
equivalent for the qmail? I don't have any knowlegde in QMail.
This is the installation guide for astfax.
( http://www.inter7.com/astfax/INSTALL )
Has you redirected all the RTP ports? You must redirect the SIP and the
RTP streams. Take a look to the rtp.conf file of your asterisk
installation to configure the RTP ports that you want to use.
Best regards.
Rpr
Alex Piqueras escribió:
Hi, I have my asterisk server inside a NAT.
When i
Preston Garrison wrote:
Ever tried alot of sip devices on one asterisk box? You will see the
need real fast :)
Yes I know (I'm running SER with 400,000 user systems and Asterisk on
the back end).
However the OP was wanting it to solve a double NAT issue (when he has
control of one of
Hello,
We have a PrimuX 1S2M ISDN card.
This card has capi driver for linux-2.4.
We are tying to configure it for Asterisk but we couldn't figure out
how to do it.
Thanks.
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Hi.
I have installed asterisk 1.0.7.
I need to send dtmf with tone duration. This functionality is in the cvs
version.
Can any body send me the app_senddtmf.so binary compiled for i386 or pentium
IV to replace the 1.0.7 version.?
I want to preserve the rest of 1.0.7 version of asterisk,
On Mon, May 30, 2005 at 11:27:06AM +0200, Pepe Aracil wrote:
Hi.
I have installed asterisk 1.0.7.
I need to send dtmf with tone duration. This functionality is in the cvs
version.
Can any body send me the app_senddtmf.so binary compiled for i386 or pentium
IV to replace the 1.0.7
and , what is more interesting,
they've omitted any reference to digium resellers
and specified only distributors :(
matteo
--
Matteo Brancaleoni
System Administrator
Tel +39.02.70633354
Sip [EMAIL PROTECTED]
Iax2 [EMAIL PROTECTED]
___
On Monday 30 May 2005 03:38, Tzafrir Cohen wrote:
334,147,0 and 334,146,0 are practically the same. As for 334,0,0:
Maybethe second patter was missed? I have the same problem here.
Yeah I was thinking the same thing; Does the distinctive ring code have any
kind of filtering (say round each
Hi all,
I'm using following software and equipment and I have very strange behavior:
Asterisk CVS-NHEAD-05/30/05-16:42:41
H323 gatekeeper - GnuGK 2.2.2
IAX2 client - Firefly 1.9.8 build 3945
H323 client - SJPhone Build 1.50.271d
H323 gateway - Welltech Wellgate 3504A
When I dial from Firefly
Hi
Is there a way that you can setup 2 [EMAIL PROTECTED] boxes, to communicate
with each other.
Example:
Caller 1 [EMAIL PROTECTED] Internet
[EMAIL PROTECTED] Caller 2
Thx Q
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Hi Andy,
Thank you so much for your help. My 7910's are now working!!!:) Now I
can work with those IP phones. I am still monitoring them. I will let
you know the further status.
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Hi!
I've been trying to get ZapRAS or PPPD to work. Neither does!
All i get is LCP: timeout sending Config-Requests
But after trying, all voicelines get crazy! It sounds like robots when
somebody calls!
And since the zaptel drivers can't unload (the server hangs totaly if I
try!), I have to
Hi,Could you please post your oh323.conf file and explain which changes are required at the HiPath? ( I have the HG3550 Card ) however I have no access to the HiPath system and I need to ask someone else to perform that changes therefore my ability to debug this issue is small.Thanks a
What encryption features are available to encrypt the IAX2 traffic between
two asterisk servers. I have read that there is some encryption possible but
has anyone been able to encrypt the entire payload of IAX traffic between
two
asterisk servers.
regards,
John
Matteo Brancaleoni wrote:
and , what is more interesting,
they've omitted any reference to digium resellers
and specified only distributors :(
Yah that's kinda bad form. And seeing as I know for a fact that the
distributors will sell directly to my customers even though we're in
different
IpSec VPN ;-)
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de John Melody
Envoyé : lundi 30 mai 2005 10:37
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] IAX encrytion
What encryption features
Hi all,
I'm trying to setup an MGCP connection between my asterisk and a third party
pbx system. I have very little control over the external pbx.
The calls are failing with the following asterisk error: notice chan_mgcp.c
2347 handle_repsonse: Terminating on result 519 from aaln/[EMAIL
Hi all,
I'm trying attended
transfer on Asterisk 1.0.7 and AT-320 phone. I met a lot of problems during this
steps, while in the blind transfer all works fine.
I had this kind of
problem:
CASE 1:
A call B
B seton
hold A
B call C (that is busy for some
reason)
B try to get the first
Hi, All
I am looking for Suppliers/resellers from Dubai - UAE to buy some VoIP
products and Digium's TDM cards. could some one send me some contact
information in this regards?. mainly I want to buy Hardware SIP phones, VoIP
gateways and DTM cards (FXS).
Thank you
Kumara
After a while, version 0.21 of the Flash Operator Panel is out.
Flash Operator Panel displays information about your Asterisk PBX
activity in real time via a standard web browser with Flash plugin. It
can monitor several asterisk servers at once. It can integrate with
CRM software, by poping up a
It seems like when I use PPPD-command, or ZapRAS, Asterisk doesn't make
it a data call, but a regular voice-call.
My ISP change their behaivour depending on the incoming call-type (data
or voice).
If it's voice, they try to open up a V.90 connection. Else (data call)
it will reply with PPP
I thought I saw a Soekris embedded in the Digium booth photos, can you run
Asterisk on one of these? How? I'd be interested in it for a back pbx, given
the reliability. In fact, might want to move my home pbx to this also.
Chris Mason
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Does anyone have complete Areski Calling Card directions? I think Areski Calling Card Idiots guide is not complete :( any other guide?
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* Andrew Kohlsmith [EMAIL PROTECTED] [050529 21:07]:
On Sunday 29 May 2005 20:59, Aidan Van Dyk wrote:
1) Simply CVS head (as of some point in time) with certain features or
bug fixes backed out
2) In addition to CVS head, some important features and bug fixes.
I think it's simply
Does anyone have a Gradwell UK SIP number successfully receiving DTMF
working with their Asterisk?
If so, please could you post the relevant bits of your config files.
Thanks in advance
Tom
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Partially. I have not finished the script that will limit the calls
depending on the money available.
Darren Wiebe
[EMAIL PROTECTED]
VoIP Newbie wrote:
Does it support pre-paid billing?
On 5/30/05, Darren Wiebe [EMAIL PROTECTED] wrote:
El Flynn wrote:
Darren Wiebe wrote:
On Sun, 2005-05-29 at 14:41 -0400, C F wrote:
This is not the CLI output. Please reproduce the problem and paste the
CLI output, from both, when it's set to 10 seconds, and when It's set
to 60.
Did you remember to answer the call before passing to voicemail??
Some PSTN providers will drop
On Mon, 2005-05-30 at 13:26 +0200, Daniel Nystrm wrote:
My server is an Dell PowerEdge 1850 with an Digium TE110P card to PRI.
I'm using stable Asterisk 1.0.6 and I'm located in Sweden with TDC
Song as telco.
Try upgrading to current stable, which is either 1.0.7 or else CVS -
STABLE.
Also,
What happens if the rate changes mid call?
IE, call starts @ 18.30 and lasts till 19.15
Rate changes @1900 to off-peak.
Darren Wiebe wrote:
Partially. I have not finished the script that will limit the calls
depending on the money available.
Darren Wiebe
[EMAIL PROTECTED]
VoIP Newbie
On Sat, 2005-05-28 at 19:19 +0100, Tom Fanning wrote:
snip
The guy mentioned Java from within the browser. I believe that I am right in
saying that a Java applet should very well be able to listen for tcp
connections as well as udp datagrams. Try this primer:
Hi,
1.0.7 does not support atxfer. You nees to use CVS. I have a bunch of the ATCOM
cheapos and only CVS will take notice of 'atxfer' in features.conf.
Otherwise , consider this scenario...
Call comes in, press HOLD, dial other party to see if they wish to speak to the
caller. If so, press *
See:
http://www.kriscompanies.com/modules.php?name=Contentpa=showpagepid=3
damn cool.
-Original Message-
From: Chris Mason (Lists) [mailto:[EMAIL PROTECTED]
Sent: Monday, May 30, 2005 7:14 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users]
Chris Mason (Lists) wrote:
I thought I saw a Soekris embedded in the Digium booth photos, can you run
Asterisk on one of these? How? I'd be interested in it for a back pbx, given
the reliability. In fact, might want to move my home pbx to this also.
Chris Mason
trye ILBC codec, and make sure that the phone configuration will use
ILBC, I dont mean the sip.conf configuration, but the specific phone
client configuration that some phones allow. Alsol try to use ILBC in
sip.conf putting disallow=all, allow=ilbc
give that a try and please post here your
Chris Mason (Lists) wrote:
I thought I saw a Soekris embedded in the Digium booth photos, can you run
Asterisk on one of these? How? I'd be interested in it for a back pbx, given
the reliability. In fact, might want to move my home pbx to this also.
Chris Mason
Chris,
You sure did see a
Hi
We are in a project where we will use asterisk as a residential gateway for
IP phone service.
We are aiming to replace the primary phone line so the service must be up
as long as possible so we are looking at ways to avoid shut downs.
We are looking for a solution to allow us to
Hi there
I am moving into a production environment. I will mostly be using Meetme,
with Ztdummy for timing. I have a question on which of 2 processor setups is
favourable.
I have the choice between Dual 2.8GHz Xeon Processors and a single Pentium 4
3.06GHz Processor. These will cost me exactly
Dustin Wildes wrote:
Chris Mason (Lists) wrote:
I thought I saw a Soekris embedded in the Digium booth photos, can you
run
Asterisk on one of these? How? I'd be interested in it for a back pbx,
given
the reliability. In fact, might want to move my home pbx to this also.
Chris Mason
The Dual 2.8GHz will be much faster for running everything. If it is the
same price it should be a no brainier, take the two CPU system.
Depending on the manufacture of the system it may even take a failure of
one CPU.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Need to provide a little more info:
What's the bus speed?
What kind of motherboard would you use with each?
What kind of RAM at what speed?
What cache size are on the CPUs?
Also, what price are these as equals? I've seen two Xeon 2.8GHz 800MHz
processors for about US$450 and a single P4 at the
Maybe my point was missed.
Hardware wise - a VIA MII EDEN based board will greatly outperform a
Soekris system, which is why my embedded platform is based on the VIA
hardware instead of the Soekris, because I AND my customers did want an
all-in-one system, and small offices tend to want an
Hi,
Thanks for yuor answer.
The boot time of the phone is very very fast, 10 sec to startup and 2 or 3
second to login to asterisk. I set the NTP server to 255.255.255.255 so it
don't try to get time.
I thinked carefully to your scenario and i am going to try it, but i don't
known if it could
I am using vtun without incident:
http://vtun.sourceforge.net/
this is a bolt-on and depending on the box's specs and the number of
tunnels, it may negatively impact the server's performance. To address this,
in my application, I use a seperate box to aggregate all of the remote IAX
servers
If I'm not mistaken, the Soekris hardware does fine for a few voice
channels - but not a very high performance piece of hardware. For
example, if you wanted a full solution as a VPN, Asterisk server, media
streaming via ICEs, web server, email server, etc... it will start to
lack
On Monday 30 May 2005 11:28, Adam Goryachev wrote:
How many channels could this board deal with when purely translating
from G.729 IAX2 - G.729 SIP
That's not a codec translation; Asterisk can simply take the IAX2 audio frames
and stuff them into RTP frames without actually deconstructing
Hi,
Some time ago, there was a discussion about the inability of nortel meridian
pbx to dial
analog tones thru an meridian ATA, and the work arround was to enable 808
function that
makes the dtmf tones long for the current call.
The nortel meridian is connected via a nortel ATA to a TDM400 to a
On Monday 30 May 2005 16:19, Giordano Grandis wrote:
Hi,
Thanks for yuor answer.
The boot time of the phone is very very fast, 10 sec to startup and 2 or 3
second to login to asterisk. I set the NTP server to 255.255.255.255 so it
don't try to get time.
Well, I run a local NTP server, so
Hi Matt,
Can I know your setting in asterisk to allow
connection from SER?
Below is my configuration in sip.conf : -
; incoming calls from ser
[ser-in]
type=friend
host=1.1.1.2
And, can I have your SER configuration file ser.cfg?
Below is my ser.cfg config: -
if (uri=~1.1.1.4) {
Andrew Kohlsmith wrote:
On Monday 30 May 2005 11:28, Adam Goryachev wrote:
How many channels could this board deal with when purely translating
from G.729 IAX2 - G.729 SIP
That's not a codec translation; Asterisk can simply take the IAX2 audio frames
and stuff them into RTP frames
On Mon, 30 May 2005 08:53:12 -0700 (PDT), Miguel Ruiz Velasco Sobrino wrote
Hi,
Some time ago, there was a discussion about the inability of nortel
meridian pbx to dial analog tones thru an meridian ATA, and the work
arround was to enable 808 function that makes the dtmf tones long
for the
Dustin Wildes wrote:
Maybe my point was missed.
Hardware wise - a VIA MII EDEN based board will greatly outperform a
Soekris system, which is why my embedded platform is based on the VIA
hardware instead of the Soekris, because I AND my customers did want an
all-in-one system, and small
The procedure that will do asterisk is very nice ;) but whe it was available ?
Currently is there any way to emprove the transfer? I tryied the scenario that
u suggest me but it doesn't work :| and i don't why.
Here my sip.conf for the phone, can u say me if there is somethingh wrong ?
[2391]
On Monday 30 May 2005 07:22, Vassilis Konstantinou wrote:
Hmmmyes but last time I played with my FXO module on the TDM400 could
not detect hangup properly (that is on a London BT line). Has this been
fixed? I keep an eye on the CVS but I have not seen any fixes for that.
Maybe I missed it.
hi,
is it possible to get to this group via nntp ?
Regards,
Marcin
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Hi,
I'm developping some AGI in perl (5.8.6) on i386
using Asterisk 1.0.5.
I want to get some variables such as DIALSTATUS and ANSWEREDTIME
after a $AGI-exec(Dial, dial_string);
but here is what i get actually:
DIALSTATUS=
DIALEDTIME=ANSWER
ANSWEREDTIME=18
I searched the archives and saw that
On Monday 30 May 2005 17:22, Giordano Grandis wrote:
The procedure that will do asterisk is very nice ;) but whe it was
available ?
Asterisk's atxfer support is only in CVS.
Currently is there any way to emprove the transfer? I tryied the scenario
that u suggest me but it doesn't work :| and
Not a problem Kristian! :-)
Same here!
Comments below:
Kristian Kielhofner wrote:
Dustin Wildes wrote:
Maybe my point was missed.
Hardware wise - a VIA MII EDEN based board will greatly outperform a
Soekris system, which is why my embedded platform is based on the VIA
hardware instead
Can any one tell me what the mysql password, no it's
not password..
Try passw0rd with a disgit ZERO not o
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Sorry Everyone, My mother past away this week.
I see there might be some fixes for this. I will try them tonight.
Thanks, David
On Thu, 2005-05-26 at 11:14 -0700, trixter http://www.0xdecafbad.com
wrote:
On Thu, 2005-05-26 at 12:48 -0500, Jay Milk wrote:
Nothing wrong with putting them
Kristian Kielhofner wrote:
Dustin Wildes wrote:
Maybe my point was missed.
Hardware wise - a VIA MII EDEN based board will greatly outperform a
Soekris system, which is why my embedded platform is based on the VIA
hardware instead of the Soekris, because I AND my customers did want
an
I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for
italian IVR an HTTP pgaes.
So i can only update asterisk with CVS and try atxfer.
Thanks for all
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill
Inviato: lunedì
From what I understand if I have an asterisk pbx set
up, I can also get a vanity DID.
1) Where I get the DID from does not matter, my voip
provider can use the DID i ask them to. Is this
correct?
2) What place has good vanity DID's?
3) Do toll free DID's save me or the person calling me
Can any one tell me what the mysql password, no it's
not password..
Try passw0rd with a disgit ZERO not o
Try amp109
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link doesnt work
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 22, 2004 8:28 PM
Subject: [Asterisk-Users] News From Astricon
We've got some replies to questions
You are lucky it is still working at all. I
have seen a very high number of the phones die alltogether.
- Original Message -
From:
Jim Duda
To: asterisk-users@lists.digium.com
Sent: Sunday, May 29, 2005 7:44 AM
Subject: [Asterisk-Users] BT100 Phone
Died During
Hi,
I am using Asterisk 1.0.7 that comes with [EMAIL PROTECTED] 1.0 ISO. I have
installed chan_unicall.c and MFCR2 support with latest Steve Underwood code
unicall-0.0.2pre2 (yes this is the latest version, not 0.0.2pre19!!)
Asterisk process is keeping the cpu at 99% most of the time.
I
thx
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: 30 May 2005 11:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED]
Can any one tell me what the mysql password, no
Hi list!
What exactly is the meaning / function of the pridialplan
prilocaldialplan?
I've been trying to find out what the different possibilities for these
settings are but couldn't find a clear answer.
The possible parameters I could find are are :
It defines the number pattern that is sent to the PRI. 90% of the time, best
practice is to set this to 'unknown' and then Asterisk will dump the dialled
digits to the PRI as the user dials them, not in some predefined pattern,
UNLESS your telco expects digits to be dialled in a certain pattern.
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'
Hi,
Are you sure the process consuming your CPU is Asterisk?
Did you tried with different codecs?
Andres Maduro wrote:
Hi,
I am using Asterisk 1.0.7 that comes with [EMAIL PROTECTED] 1.0 ISO. I have
installed chan_unicall.c and MFCR2 support with latest Steve Underwood code
If I have both clients and Asterisk in the same nat, it is working fine with
internal addressing.
When using outside IP with appropriate ports open(5060,1-2000), with
following call flow,
X-Lite-asterisk -- nat-- - as5400-pstn
canreinvite=no
nat=yes
with this setting RTP's should be
Hi, my setup is like:
phones (g729/g711)--(SER)-- Asterisk --(oh323)--gateway (supports
g729g711)
problem begin when phone supports only g711 and Asterisk doesn't
negotiate this codec in full path (from phone to gateway), but tries to
do transcoding (and because I haven't g729 codec in
On Mon, 30 May 2005, Remco Barende wrote:
What exactly is the meaning / function of the pridialplan
prilocaldialplan?
Both set the two fields Type Of Number (TON) and Numbering Plan (NPI)
markers on an outgoing isdn call. These two tell a receiving isdn switch
how to interpret the
This is usually a mpeg123 problem. try removing the moh module and see if it
goes away.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Sebastian
Silva
Sent: Monday, May 30, 2005 11:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Sorry if this show up twice, yesterdays post did not show up.
We are having trouble setting up two ext's on one phone. We have it the
point where the first two lines are ext 4000 then the third line is ext
4013. We can receive calls to both ext's but we can't make out going calls
on ext 4013.
I think the first thing you have to do is to set each extension to a
different port..
Line 1 Port 5060 ... Line 2 ... 5061, etc.
The way this has worked for me is this:
Line 1 is configured with 4000... Line 2 would not be configured in
the phone.. Line 2 defaults to a dup of line 1. You would
wich version are you trying to install? and how? wich GNU/Linux
distro? did you already installed zaptel?
best regards
On 5/30/05, Ghassan Lama [EMAIL PROTECTED] wrote:
Hi;
It is my first time to use asterisk I have TDM400 wildcard and 4 FXO Modules
when I install asterisk an error
On Mon, May 30, 2005 at 06:32:34PM +0200, Marcin Kuczera wrote:
hi,
is it possible to get to this group via nntp ?
Yes.
http://gmane.org/find.php?list=asterisk
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL
I don't believe any of that works on a Polycom 500. Each Line appearance can
handle two concurrent calls, so they don't roll over properly.
The best way I have found, if you need that many extensions, is to allocate
one extension per button and roll them over in your dialplan.
exten =
Message: 19
Date: Mon, 30 May 2005 11:12:13 -0500
From: Carlos Chavez [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Meridian 808 Function
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;
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 Kewlstart signalling
May 30
On Thu, September 23, 2004 8:41, Steve Totaro said:
link doesnt work
You may want to take a look at your system settings, as I think it
unlikely that this e-mail has been in transit for approx 8 months... ;-)
--
Francesco Peeters
GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3
If you mentioned it, I missed it. I'm assuming you are trying to
use a digium TDM-fxo card?
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
Hi,
I'am having some problems with new mainboards and 3xHFC-S cards.
The the first problem was with interrupts, I mean if HFC-S card was using
interrupt i.e. 21 or higher - it didn't work.
Solved by disabling APIC.
However, still the driver behaves a little bit strange.
If card 0 1 is TE and 2
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