hi,
for your help:)
In fact, I have installed MYSQL successfully, for I have tested it and can
use it for store cdr date.
do it need some configuration for connect it?
config.c:920 find_engine: Realtime
mapping for 'sippeers' found to engine 'mysql', but the engine is not
available
On Sun, May 28, 2006 at 11:41:00PM -0400, Steve Totaro wrote:
Henry J. Cobb wrote:
to increase the security for remote extensions I would like to limit a
sip-peer to a specific MAC address. Is it possible to hook into the
authentication mechanism in asterisk and allow/deny incoming
On 5/27/06, Shenen Shenen [EMAIL PROTECTED] wrote:
Hi!I've installed [EMAIL PROTECTED] and I have a ISDN card,(Cologne Chip Design GmbH ISDN network controller [HFC-PCI](rev 0.2)This is how I installed bristuff:how to install hfc cardafter unload asterisk and amportal whit
amportal stoptype
Shenen Shenen wrote:
On 5/27/06, *Shenen Shenen* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi!I've installed [EMAIL PROTECTED] and I have a ISDN card,(Cologne
Chip Design GmbH ISDN network
controller [HFC-PCI](rev 0.2)
This is how I installed bristuff:
how to
I doubt this would be a codec issue . can you tell us what is the codec used on the sip and iax --Giridhar BandiOn 5/29/06, MC
[EMAIL PROTECTED] wrote:Got 1 issue I can't seem to knock out of this particular box.
The IVR works fine on the zap channels and the incoming SIP routes. Butcoming in via
Coming in on the IAX route is G729.
On the SIP lines it is alaw.
Giridhar Reddy Bandi wrote:
I doubt this would be a codec issue . can you tell us what is the
codec used on the sip and iax
--Giridhar Bandi
On 5/29/06, *MC* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
Got 1
Hi,
I have a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a
TDM2400P with echo canceller. I installed the card but no echo
cancellation is being made...seems like the echo canceller module does
not work, infact the software cancellation is working.
My zapata.conf has
have you edit res_mysql.conf file ?
On 5/29/06, 吴应芳 [EMAIL PROTECTED] wrote:
hi,
for your help:)
In fact, I have installed MYSQL successfully, for I have tested it and can use
it for store cdr date.
do it need some configuration for connect it?
config.c:920 find_engine: Realtime
mapping
this problem has been resolved. Its actually a dtmf problem.
From: Akpome Akpoguma [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] # key
Date: Wed, 24 May 2006
everything is ok!
thanks!!
:)
have you edit res_mysql.conf file ?
On 5/29/06, 吴应芳 [EMAIL PROTECTED] wrote:
hi,
for your help:)
In fact, I have installed MYSQL successfully, for I have tested it and can
use it for store cdr date.
do it need some configuration for connect it?
The following is my AGI script done in perl
#!/usr/bin/perl
use strict;
use DBI;
$|=1;
my %AGI;
while(STDIN) {
chop;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
my $ext = $AGI{extension};
if
Hello,
I use and sale (as Distributor) Micronet and Aliwei gateways.
Fine and stable, without echo.
Each port is seen as a separate SIP account.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Nikolay
Pavlov
RA == Rich Adamson [EMAIL PROTECTED] writes:
RA setting. Essentially, the switch port and the attached device
RA auto negotiates at the same time, and one device sees what it
RA thinks is half duplex when the other device is in the middle of
RA its negotiation process. In most cases, statically
On Fri, 26 May 2006, Guido Hecken wrote:
We had the same problems with some cheap LevelOne Switches.
The Snoms rebooted during a call, calls dropped etc.
Replacing the switches was the solution.
A switch should NEVER cause ANY device to lockup, ever. Period.
If a phone locks up /
Hi,
I need some basic help to get going. I have done the following in
extensions.conf on * machine #1 to use mwanalyze for incoming calls:
--snip--
; Mwanalyze
exten = 31,1,Answer
exten = 31,2,Mwanalyze(8|8000|10|328)
exten = 31,3,NoOp(${mwa_ampitude})
exten = 31,4,NoOp(${mwa_ripple})
exten =
Guido Hecken wrote:
I looked long and hard at the LAN and it was basically narrowed down to
the
switches. In this smaller install, several cheapo Dlink ($30) switches
de-aggregate a Cisco Catalyst switch. What I noticed was that any phone
plugged direcly into the Catalyst did *not*
In your extension.conf, in the misdn context you defined in
/etc/asterisk/misdn.conf you have to add something linke the following line
[from-pstn]
exten = 0108680550,1,Dial(SIP/201)
If you don't want to have to write a string for each called extension,
you can put something like. Obviously
Hey all,
I am not sure if this is the correct place to do this, however, I have
been working on a New Zealand style voice prompt set.
(This has also been announced on New Zealand Asterisk users list)
Finally these have got to a point where I consider them to be stable
and able to be used in a
Hi, Ciao
I have a bunch ( 30) 320's connected to HP switches, fw version 6.0.4
They work great but occasionally they where signalling the warning network
cable disconnected.
Monday, May 29, 2006, 10:38:09 AM, you wrote:
TC Guido Hecken wrote:
I looked long and hard at the LAN and it was
Hi
Guys
This has been
discussed a little in the list before so my apologies for sendig it again but I
have done what others have done in the list but to no avail.
I have configured
Asterisk to send the callerID of extension phones as "firstname lastname" and
that seems to work well and
Hi, folks.
Did someone use successfully Asterisk with Asotel Dynamix DW04/S
terminal? Is there any problems with configuration?
--
=
= Best regards, Nikolay Pavlov. =
what does pedantic=no|yes means?-- Regards,Mark Quitoriano, CCNAFan the flame...http://www.spreadfirefox.com/?q=user/registerr=19441
___
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To UNSUBSCRIBE or update options
Is there any reason why I cant see the environment dump display on asterisk
console when call agi-test.agi from my dialplan?
reponses would be appreciated
_
Express yourself instantly with MSN Messenger! Download today it's FREE!
did you check your verbose level for your console?On 5/29/06, Akpome Akpoguma [EMAIL PROTECTED] wrote:
Is there any reason why I cant see the environment dump display on asteriskconsole when call
agi-test.agi from my dialplan?reponses would be
Thanks i guess that must be the problem.
From: Marco Mouta [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re:
SIP uses port 5060 by default. Chances are your SIP phones are set to
use port 5060 by default. Some phones have a tick box that says Use
Random Port or you can specify a port. Start with port 5060 and move
up so phone one would be 5060 phone two 5061 and so on. The problem is
most likely
Hi,
i want to define some call groups like:
extensions.conf
[globals]
GROUP1=IAX2/idefixSIP/200
[capi-in]
exten = 55,1,Dial(${GROUP1})
exten = 55,2,Hangup
But Dial will not dial the defined numbers.
exten = 55,1,Dial(IAX2/idefixSIP/200) works fine.
What is here wrong?
Or is there a better way
trixter aka Bret McDanel wrote:
On Sun, 2006-05-28 at 23:41 -0400, Steve Totaro wrote:
Henry J. Cobb wrote:
to increase the security for remote extensions I would like to limit a
sip-peer to a specific MAC address. Is it possible to hook into the
authentication mechanism in asterisk
I trust you have G729 licenses? Try GSM instead as a test. If it works
then the problem must be with the G729.
Thanks,
Steve Totaro
MC wrote:
Coming in on the IAX route is G729.
On the SIP lines it is alaw.
Giridhar Reddy Bandi wrote:
I doubt this would be a codec issue . can you tell us
I like the TenorAX but that might be overkill for your situation (24
ports) but the amount of features and settings is almost too much.
Thanks,
Steve Totaro
[EMAIL PROTECTED] wrote:
Hello,
I use and sale (as Distributor) Micronet and Aliwei gateways.
Fine and stable, without echo.
Each port
please check if you are able to hear the sounds using alaw on IAX i had some problem listening to the sounds using G729 on sip client --Giridhar BandiOn 5/29/06,
MC [EMAIL PROTECTED] wrote:Coming in on the IAX route is G729.
On the SIP lines it is alaw.Giridhar Reddy Bandi wrote: I doubt this
Hi All,
First off all, this is my first mail to this mailing-list, so if I am
doing something wrong please tell me. And apologies for my english in
advance, it's not my native language.
Anyway, I have few machines running Asterisk 1.2.7.1. All machines but
one are Gentoo (other one is Debian).
Hi all,
I hava SER with many clients (sipura SPA2100). One of these is an
Asterisk which have others clients (sipuraSPA2100).
I also have a Cisco GW which give me access to the PSTN.
I make calls to all IP phones in my network, but I can't call PSTN
numbers. After I dial, I hear 2 ringbacks but
I'm also not an expert, but could it as any relationship with your Telephony card drivers??Which Telephony boards do u use?On 5/29/06, Attilla de Groot
[EMAIL PROTECTED] wrote:Hi All,
First off all, this is my first mail to this mailing-list, so if I amdoing something wrong please tell me. And
Hardware platform and specs? Call volume? Any messages in your logs?
I had this problem on an Itanium2 box, went away when I downgraded to
a Xeon.
Attilla de Groot wrote:
Hi All,
First off all, this is my first mail to this mailing-list, so if I am
doing something wrong please tell me.
Marco Mouta wrote:
I'm also not an expert, but could it as any relationship with your
Telephony card drivers??
Which Telephony boards do u use?
None. :)
I only use Asterisk as an VoIP pbx. Only the zaptel drivers installed
for ztdummy as a timer interface.
Greetings,
Attilla
Hi, I am new here on
thislist, and have a problem of which I hope that somebody here can help
me with it. I have a Voipbuster account, with which I would like to make
phone calls via my Asterisk PBX. If I let X-Lite register directly at
voipbuster.com, everything is OK, but if I let
Steve Totaro wrote:
Hardware platform and specs? Call volume? Any messages in your
logs? I had this problem on an Itanium2 box, went away when I
downgraded to a Xeon.
Hi Steve,
At this moment I don't have real acurate statistics. But think of 40
calls a day. So nothing fancy I think. And
Benny Amorsen wrote:
RA == Rich Adamson [EMAIL PROTECTED] writes:
RA setting. Essentially, the switch port and the attached device
RA auto negotiates at the same time, and one device sees what it
RA thinks is half duplex when the other device is in the middle of
RA its negotiation process. In
Remko Muis wrote:
Hi,
I am new here on this list, and have a problem of which I hope that
somebody here can help me with it.
I have a Voipbuster account, with which I would like to make phone
calls via my Asterisk PBX. If I let X-Lite register directly at
voipbuster.com, everything is OK,
Maybe a silly question but can you ping sip.voipbuster.com from your
asterisk box?
Second question and probably the answer, what is your dial statement in
extensions.conf?
Contact:sip:[EMAIL PROTECTED] EXTERN IP]
One way to test is to create a dial statement like this exten =
exten = _4XX,1,Dial(SIP/[EMAIL PROTECTED])
it works to me (my provider sends me the last 3 digits)
I hava SER with many clients (sipura SPA2100). One of these is an
Asterisk which have others clients (sipuraSPA2100).
I also have a Cisco GW which give me access to the PSTN.
I make calls to
RA == Rich Adamson [EMAIL PROTECTED] writes:
RA That's a total crock. There isn't any such thing as other side
RA doesn't answer for speed duplex negotiation.
Of course there is. Each side advertises which speed and duplex
settings it supports, and so they pick a setting which both support.
On
May be updatedb or some other such heavy application, which runs at
night is causing heavy load on the system and spoils the working of
asterisk.
See if this phenomenon happens at the same time of the day everyday. Also, see what processes run at *that time*.
Cheers,
Vij
On 5/29/06, Attilla de
Hi Steve Attilla,
Thanks for the quick replies!!
Attilla: your suggestion sounds promising, since I know my system clock is
not too accurate. But that is the reason I use the network time protocol
daemon. Time and date settings are now correct.
Steve: your question about pinging the
On Mon, May 29, 2006 16:20, Remko Muis said:
Hi Steve Attilla,
Thanks for the quick replies!!
Attilla: your suggestion sounds promising, since I know my system clock is
not too accurate. But that is the reason I use the network time protocol
daemon. Time and date settings are now correct.
If the domain resolves you are probably OK, they just dont reply to pings.
Type asterisk -r then type sip debug and even set verbose 15 and
try to dial. Post the relevant console output. Also, disable iptables
for testing, just to eliminate that as an issue.
Thanks,
Steve
Remko Muis
Peter,the configurations that I have seen do not auto-answer on CID. You need to set the Alertinfo field in the sip header in order to make this work. The polycoms do have an ability to customize the ring based on the caller which is set in the telephone's inernal directory. You may be able to
Hi Francesco,
No, I'm using the DNS servers of my ISP (Wanadoo). Here is the output of
traceroute:
[EMAIL PROTECTED] asterisk]# traceroute sip.voipbuster.com
traceroute: Warning: sip.voipbuster.com has multiple addresses; using
194.120.0.203
traceroute to sip.voipbuster.com (194.120.0.203),
Steve,
I will try that, but now I am at my office. Can I dial some number from the
command line ;-) ?
Thanks,
Remko
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday,
On Thu, May 25, 2006 at 04:42:57PM -0600, Colin Anderson wrote:
CA I looked long and hard at the LAN and it was basically narrowed down to the
CA switches. In this smaller install, several cheapo Dlink ($30) switches
What switches you mean? How they named?
--
JID: [EMAIL PROTECTED]
ICQ:
Well I just set the port to 5061, and no other devices on this end have
that port. I still have the same problems though. The strange thing is
that I have better luck calling the asterisk box itself rather than an
outside line, but even that is intermittent. Actually what I have found
is
If you call Digium they will help you get the card configured properly. You
get installation support with any of their hardware products.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Giorgio Incantalupo
Sent: Monday, May 29, 2006 12:33 AM
To:
No. If you can ssh into the box you could tunnel VNC to a windows box
and try from a softphone there. Thats how I do it.
Remko Muis wrote:
Steve,
I will try that, but now I am at my office. Can I dial some number
from the command line ;-) ?
Thanks,
Remko
- Original Message -
Make sure you have qualify=yes for each phone. Type sip show peers in
the asterisk CLI and post the output when and when you are not able to
make calls. Make sure that the new port settings are reflected in asterisk.
Miles Scruggs wrote:
Well I just set the port to 5061, and no other devices
Hi,
I'd like to use the TDM2400P echo cancellation hardware module but I do
not know how to set the correct parameter inside zapata.conf. Where can
I find an example? Must I use opermode when loading zaptel module since
I live in Italy?
TIA
Giorgio Incantalupo
Kevin P. Fleming wrote:
Vij wrote:
May be updatedb or some other such heavy application, which runs at
night is causing heavy load on the system and spoils the working of
asterisk.
See if this phenomenon happens at the same time of the day everyday.
Also, see what processes run at *that time*.
Cheers,
Vij
Hi
yup everything is there:
Name/username HostDyn Nat ACL Port Status
pap2-2/pap2-2 123.123.123.123D N 5062 OK (93 ms)
pap2-1/pap2-1 123.123.123.123D N 5061 OK (39 ms)
I'm really confused why it has N for NAT when the
How do you enable call waiting on sip phones? Ive looked and googled and
can only find call waiting pstn phones butnot for sip. Is their a way of
setting this up within the dailplan?
___
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It doesn't work for me :-(
How do you have the peer configuration in asterisk, to connect ot SER?
Sebastian
On 5/29/06, Woodoo People .pGa! [EMAIL PROTECTED] wrote:
exten = _4XX,1,Dial(SIP/[EMAIL PROTECTED])
it works to me (my provider sends me the last 3 digits)
I hava SER with many
How do you enable call waiting on sip phones? Ive looked and googled and
can only find call waiting pstn phones butnot for sip. Is their a way of
setting this up within the dailplan?
Which SIP phone ?
If you're using [EMAIL PROTECTED], you have to dial *70
hth
Hi,
We've got troubles trying to connect a Brother 8360P
to a TDM400 FXS board.Design is :PSTN -- Junghanns QuadBRI - Digium TDM400 --- Brother faxAs you can guess, QuadBRI and TDM400 are plugged into 1.0.10 Asterisk server.
- from Asterisk console,we can observe calls
Until now, I had never heard about VNC-tunnels. I will install TightVNC as
soon as I am home. Thanks for the hint! The next thing to do then is posting
the output of sip debug while dialling some number. I fear, however, that it
will be terribly long, because of the frequent registration
We are using the cards successfully with Zap 1.2.4 and Zap 1.2.5, but
I've never tried it with 1.2.1. The echo keywords are sensitive as to
where they are placed in the file. If you are still getting echo after
turning that on, I would start tweaking the rxgain and txgain.
The settings
Is there any chance you're connecting to a remote share using CIFS?
What does slabtop look like on your machines?
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On 29-May-06, at 8:35 AM, Attilla
How is it implemented within the dialplan and can call waiting be
implemented for softphones? Is their a way to do this. In my sip.conf file
for one of my configured softphones ive used the limit-call parameter to
limit calls to one across this channel or softphone. Would this invalidate
Moises Silva wrote:
my mistake.
the line in the unicall.conf should be something like
loglevel=4
Hi Moises,
I search for a error message and found this thread in the list:
http://lists.digium.com/pipermail/asterisk-users/2006-April/148978.html
I'm having the same error here.
Do you have
How is it implemented within the dialplan and can call waiting be
implemented for softphones? Is their a way to do this. In my sip.conf file
for one of my configured softphones ive used the limit-call parameter to
limit calls to one across this channel or softphone. Would this invalidate
Anthony Rodgers wrote:
Is there any chance you're connecting to a remote share using CIFS?
What does slabtop look like on your machines?
I would like to answer both question, but I don't know what CIFS of
slabtop is. But I'm sure you can tell me.
Greetings,
Attilla
Attilla de Groot wrote:
Vij wrote:
May be updatedb or some other such heavy application, which runs at
night is causing heavy load on the system and spoils the working of
asterisk.
See if this phenomenon happens at the same time of the day everyday.
Also, see what processes run at *that
N means NAT. No N no NAT.
Can you call now with audio in both directions? Can you set the phones
to register every two minutes (expiration)? Is the output from sip show
peers still the same before and after the audio working? Does sip debug
give any info? What type of router?
More
Remko Muis wrote:
Steve,
I will try that, but now I am at my office. Can I dial some number
from the command line ;-) ?
Thanks,
Remko
Not from the command line, but you *can* from the manager API...
(not that it matters now, as I'm sure you're home now, just like me G)
Have a look at
Attilla de Groot wrote:
Anthony Rodgers wrote:
Is there any chance you're connecting to a remote share using CIFS?
What does slabtop look like on your machines?
I would like to answer both question, but I don't know what CIFS of
slabtop is. But I'm sure you can tell me.
Hello Giorgio,
I am a TDM2400 happy user. :-)
Could you show your zaptel.conf zapata.conf config files ?
Think to tell us how many modules you have and where they are plugged on the
TDM2400P.
Are the leds on the echocan modules running as a LasVegas casino (scrolling
in a circular pattern) ?
If
Time Bandit wrote:
How is it implemented within the dialplan and can call waiting be
implemented for softphones? Is their a way to do this. In my sip.conf
file for one of my configured softphones ive used the limit-call
I gathered that but it has its uses. Could you then give us soem tips
I gathered that but it has its uses. Could you then give us soem tips on
how to get this working. Call forwarding is a done deal but i cant seem to
find any info on call waiting anywhere? Help needed. Customer fustrated.
Are you using [EMAIL PROTECTED] ?
If not, are you using AMP (now
Is it possible that voipbuster refuses to connect to asterisk?, perhaps
asterisk agent is blacklistet by them
Josep
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Remko Muis
Enviado el: lunes, 29 de mayo de 2006 16:51
Para: Asterisk Users Mailing List -
When i made internal call into my LAN using x-lite sip phone client I
retrive in askterisk CLI :
---
ERROR
--
Verbosity is at least 6
-- Remote UNIX connection
-- Executing Dial(SIP/201-979d, SIP/201|60|t) in new stack
-- Called 201
May 29 18:09:28 WARNING[6082]:
Carlos Chavez wrote:
Now that Nufone is dead, what are other providers of 800 numbers that
work with Asterisk?
Nufone is NOT dead. It is working and I just added more funds into my
account.
You may also consider Asterlink. I'm a new client there, their support
is a little slow, sometimes
Justin Newman wrote:
Did echo disappear?
no - it has always been there.
Jeremy
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To UNSUBSCRIBE or update options visit:
At 10:59 AM 5/29/2006, you wrote:
I gathered that but it has its uses. Could you then give us soem tips on
how to get this working. Call forwarding is a done deal but i cant seem to
find any info on call waiting anywhere? Help needed. Customer fustrated.
I'm likely going to make a fool of
We need to conference together a call center agent, a customer and a third
party IVR and send DTMF tones from the agent to the IVR.
MeetMe has been eating our DTMFs so we'd like to try app_conference.
Has anybody setup such a configuration in app_conference and how did you
configure it?
-HJC
Hola Omar:
solo cambia tu extension.conf
[entrada]
exten = s,1,Wait,11
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Dial(SIP/200,60,Ttr)
exten = s,5,Dial(SIP/201,60,Ttr)
exten = s,6,Dial(SIP/202,60,Ttr)
exten = s,7,Dial(SIP/203,60,Ttr)
Saludos.
- Original Message -
From: Omar
Hmm all your questions are covered in this email, but I'll summarize it
again in this reply:
Server: 1.2.7.1 direct connection to the Internet
config settings:
[pap2]
type=friend
secret=something
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=private
callgroup=6
pickupgroup=6
On 5/29/06, Juan Miguel Yamakawa [EMAIL PROTECTED] wrote:
Hola Omar:
solo cambia tu extension.conf
[entrada]
exten = s,1,Wait,11
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Dial(SIP/200,60,Ttr)
exten = s,5,Dial(SIP/201,60,Ttr)
exten = s,6,Dial(SIP/202,60,Ttr)
exten =
Miles Scruggs wrote:
Hmm all your questions are covered in this email, but I'll summarize it
again in this reply:
Server: 1.2.7.1 direct connection to the Internet
config settings:
[pap2]
type=friend
secret=something
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=private
Derek Whitten wrote:
Miles Scruggs wrote:
Hmm all your questions are covered in this email, but I'll summarize it
again in this reply:
Server: 1.2.7.1 direct connection to the Internet
config settings:
[pap2]
type=friend
secret=something
qualify=yes
nat=yes
host=dynamic
canreinvite=no
Josep Aguilar wrote:
Is it possible that voipbuster refuses to connect to asterisk?, perhaps
asterisk agent is blacklistet by them
Josep
Unlikely, as mine connects just fine...
--
Francesco
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do you have any problems with audio connection
mine connection is just fine but audio from voipbuster.com is poor
(breaking)
with any other SIP client audio is OK
fpeeters pravi:
Josep Aguilar wrote:
Is it possible that voipbuster refuses to connect to asterisk?, perhaps
asterisk agent is
Miles Scruggs wrote:
Derek Whitten wrote:
Miles Scruggs wrote:
Hmm all your questions are covered in this email, but I'll summarize it
again in this reply:
Server: 1.2.7.1 direct connection to the Internet
config settings:
[pap2]
type=friend
secret=something
qualify=yes
nat=yes
Matic wrote:
do you have any problems with audio connection
mine connection is just fine but audio from voipbuster.com is poor
(breaking)
with any other SIP client audio is OK
fpeeters pravi:
Josep Aguilar wrote:
Is it possible that voipbuster refuses to connect to asterisk?, perhaps
On Fri, 2006-05-26 at 08:26 +0200, Pieter Claassen wrote:
Can anybody recommend a reseller in Europe (Netherlands) for modules for the
X100P (FXO and FXS modules)?
Cost, support are important.
Also, what is a reasonable price for an X100P in Europe? Is there a
difference
in price
Steve Totaro wrote:
Miles Scruggs wrote:
Derek Whitten wrote:
Miles Scruggs wrote:
Hmm all your questions are covered in this email, but I'll
summarize it
again in this reply:
Server: 1.2.7.1 direct connection to the Internet
config settings:
[pap2]
type=friend
secret=something
Well, being unable to compile mpg123 under x86_64 i installed lame and
transformed the mp3--wav--raw.
and using files as the format player.
Are there any good scripts to stress test MoH?
I want to test this machine for 1000 calls on hold.
http://www.asteriskguru.com/tutorials/astertest.html
Not sure what else to tell you. If the eyebeams work fine then the
problem must be your in your linksys PAP2-NA.
Well I'm sure it does, but what I can't figure out is why it would
work intermittently. What is interesting is the eyebeams register on
random ports such as: 24130, 8332, or
Time Bandit wrote:
I gathered that but it has its uses. Could you then give us soem tips on
how to get this working. Call forwarding is a done deal but i cant seem
to
find any info on call waiting anywhere? Help needed. Customer
fustrated.
Are you using [EMAIL PROTECTED] ?
If not,
No definitely not. These tones are generated by the phone, not Asterisk.
H
On 5/28/06, T.S [EMAIL PROTECTED] wrote:
Sure that's not the message waiting stuttering indicator?
Terrelle
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hugolivude
Sent:
Hi,
I'd like to use the convenience of apt packaging, but debian sarge's
default asterisk is something like 1.0.7.
Are there any apt repositories which provide newer versions of the software?
Thanks!
--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP Telecom
Jean-Michel Hiver wrote:
I'd like to use the convenience of apt packaging, but debian sarge's
default asterisk is something like 1.0.7.
Are there any apt repositories which provide newer versions of the
software?
sure: http://pkg-voip.buildserver.net/debian
=Stefan
--
reuter network
Stefan Reuter a écrit :
Jean-Michel Hiver wrote:
I'd like to use the convenience of apt packaging, but debian sarge's
default asterisk is something like 1.0.7.
Are there any apt repositories which provide newer versions of the
software?
sure: http://pkg-voip.buildserver.net/debian
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