Hi,
the problem was the telco line was not a pure telephone line but a mixed
one (phone + data).
Changing the line solved everything!
Giorgio
P.S.: reverse parameters are not not necessary anymore!
[EMAIL PROTECTED] wrote:
Hi Giorgio,
Which telco are you using? I never used a TDM 400 card
On Tue, Jan 30, 2007 at 09:24:48AM +0100, Giorgio Incantalupo wrote:
Hi,
the problem was the telco line was not a pure telephone line but a mixed
one (phone + data).
Changing the line solved everything!
Data? On an analog line? Do you mean some DSL?
--
Tzafrir Cohen
On 2007-01-28 08:37:43 -0800, Eric Germann [EMAIL PROTECTED] said:
We LOVE Teliax. We're on a Time Warner business class fiber connection and
avg 25ms latency from Ohio to Denver CO.
With that connection I would love Teliax also.
Marty
___
On Mon, Jan 29, 2007 at 12:05:28PM +0100, Benny Amorsen wrote:
PC When I set for Extern1/2 canreinvite=yes it works, but
PC Intern-2-Extern doesn't work because Asteisk gives out the
PC private IP-Adresses of Int1/2
Asterisk can't give out a public IP-address for Int1/2. Where
would it get
Hello all,
Witch snmp system do you use to collect info about their asterisk boxes, for
example, uptime, downtime, max load, HD, free memory, asterisk status,
,etc?
I have made a look to Cacti and MRTG, but I am not sure they will monitor
asterisk.
Witch is best snmp system to monitor
Brad Templeton schrieb:
On Mon, Jan 29, 2007 at 12:05:28PM +0100, Benny Amorsen wrote:
PC When I set for Extern1/2 canreinvite=yes it works, but
PC Intern-2-Extern doesn't work because Asteisk gives out the
PC private IP-Adresses of Int1/2
Asterisk can't give out a public IP-address for
HI, today I download Web-MeetMe-3.0.0 for asterisk 1.4.0 but when I call the
extension which invoke cbmysql, a warning appears:
WARNING[20225] pbx.c: No application 'CBMysql' for extension (default,
1995, 3)
I check the application, it didn't registered
CLI core show
Hello, we have this problem with Trixbox 1.23
I have created an outgoing route where the 1st line
has Voipjet and the 2nd an 3rd have voipcheap accounts.
The problem is that at certain moments, when we call all
the calls go through the voipcheap SIP accounts SIP, whose
quality are not only not
Hi guys.
I'm working on a VOIP service provider.
We have two customers running asterisk. Customer A and B.
When A call to B everything is ok.
When B call to A the call ring but sip messages didn't arrive on
asterisk A. In my softswitch i see the invite sip message sended to A.
When every other
That's what I get:
The requested URL / was not found on this server
:)
l.
On Mon, 29 Jan 2007 23:16:47 +0100, Facundo Ameal [EMAIL PROTECTED] wrote:
Hello everyone! In my humble try of creating a Blog, I've made this:
http://fameal.blogdns.org.
By now, it's hosted in my own server but
Kind of - you could link that to the Local/xxx channel called for agents,
or you could fork the dialplan and on one branch send the user to the
queue and on the other one run the AGI.
l.
On Mon, 29 Jan 2007 15:55:21 +0100, nik600 [EMAIL PROTECTED] wrote:
Hi everyone
dou you know if is
Did you try Druid from Voiceroute? it's a commercial product, but we find
it pretty handy.
l.
On Mon, 29 Jan 2007 04:09:38 +0100, Santiago del Castillo
[EMAIL PROTECTED] wrote:
Hi, I'm looking a queue manager compatible with queues.conf. It should
allow me to change agents from one
Environment:
Asterisk 1.2.14, FreePBX 2.2.0, Aastra 480i IP telephones firmware
version 1.4.1.1077.
Problem:
Intercom feature: the dialed phone does not play the splash tone when
auto-answering an intercom call. Otherwise, intercom works perfectly.
Questions:
What is the extensions.conf syntax
Hi,
I just want out find out how to do bill recon's when you send calls to a provider. They send me
their CDR's, and when I compare it to my * CDR's, some calls are 1 second off, either way.
How in general is it done by others?
--
thanks,
Yusuf
___
nothing
On 1/28/07, Paul Hales [EMAIL PROTECTED] wrote:
What appears on the Asterisk console?
PaulH
On Sun, 2007-01-28 at 20:06 -0500, James Caffrey wrote:
Hello everyone. I am having trouble receiving via my Linksys SPA-3102.
I have not problem dialing out. It is like asterisk never
If you have r option in the Dial command remove it.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken Morley
Sent: Tuesday, January 30, 2007 9:06 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] No intercom splash tone?
I have a customer that wants to hire someone to install and
configure SugarCRM for them on their Fonality Astrerisk
system.You can do it remotely.They will pay by the hour and
figure it could take up to a month.
Ted Gibson
Director of Engineering
TGI
On 1/30/07, Lenz [EMAIL PROTECTED] wrote:
Kind of - you could link that to the Local/xxx channel called for agents,
or you could fork the dialplan and on one branch send the user to the
queue and on the other one run the AGI.
l.
sorry, i've not completely understand.
This is the scenario that
So I have my extensions.conf
(http://www.infiltrated.net/exten.stupidity.conf) shortened
in case someone wants to look. Has someone encountered the following?
I've racked my
brain on this for too long...
I have two contexts, day and night...
Caller (Daytime) -- Dials an extension -- Caller
Greetings!
I've searched far and wide for an answer and have gotten no where, so I
was hoping one of you guys might have the answer;
Is it possible to dynamically add a context to the dialplan?
You can add extensions via the CLI, however if the context doesn't exist
I get an error message
j wrote:
Greetings!
I've searched far and wide for an answer and have gotten no where, so I
was hoping one of you guys might have the answer;
Is it possible to dynamically add a context to the dialplan?
You can add extensions via the CLI, however if the context doesn't exist
I get an error
You seem to have to many s,3's
[main-night-aa]
exten = s,1,Answer
exten = s,2,Background(/etc/asterisk/night)
exten = s,3,Directory(foobarcorp,internal,l)
exten = s,3,Wait(3)
exten = s,4,Voicemail([EMAIL PROTECTED])
exten = s,5,Hangup
exten = 1,1,Directory(foobarcorp,internal,l)
exten =
Yes, there are duplicate 3 lines, which can cause havoc. For this reason it is
recommend you use 'n' in your contexts. Such as:
[main-night-aa]
exten = s,1,Answer
exten = s,n,Background(/etc/asterisk/night)
exten = s,n,Directory(foobarcorp,internal,l)
'n' allows for the addition and deletions
Hi,
I'm configuring a ISDN trunk using 2 old Diva PCI 2.01 ISDN cards. I followed
the indications in http://www.voip-info.org/wiki-Asterisk+ISDN4Linux, and I can
call and receive call perfectly.
But I have the following problem: After some seconds (between 0 and 10)
Asterisk starts to write
It seems to work OK for me.
I have heard there are problems if you have dual nics on the same subnet, but
that makes sense.
I have one for internal, one for external and set up DNS to have my server name
on our public DNS.
My laptop users no longer have to VPN in to use their softphones and I
On Tue, 2007-01-30 at 18:04 +0200, yusuf wrote:
j wrote:
Greetings!
I've searched far and wide for an answer and have gotten no where, so I
was hoping one of you guys might have the answer;
Is it possible to dynamically add a context to the dialplan?
You can add extensions via the
john beaman wrote:
Yes, there are duplicate 3 lines, which can cause havoc. For this reason it is
recommend you use 'n' in your contexts. Such as:
You seem to have to many s,3's
Thanks but no dice. I removed them and re-tried, still the same error.
Asterisk shows the phone is
Hello!
I've upgraded from 1.2.9 to 1.2.14 recently but experience an
unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was
playing continuous on sequential extensions after a
timeout, it is restarted for every extension in 1.2.14:
;music starts
exten = 902,1,Dial(SIP/[EMAIL
I am experiencing the same problem. Fresh install.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ma Zhiyong
Sent: Tuesday, January 30, 2007 6:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] web-meetme
J. Oquendo wrote:
john beaman wrote:
Yes, there are duplicate 3 lines, which can cause havoc. For this
reason it is recommend you use 'n' in your contexts. Such as:
You seem to have to many s,3's
Thanks but no dice. I removed them and re-tried, still the same error.
Asterisk shows
john beaman wrote:
Yes, there are duplicate 3 lines, which can cause havoc. For this reason it is
recommend you use 'n' in your contexts. Such as:
More debug logs... When I dial the main-night-aa number this is what
Asterisk sees via debug... The call DOES GO THROUGH just fine, the
Ma write:
HI, today I download Web-MeetMe-3.0.0 for asterisk
1.4.0 but when I call the extension which invoke
cbmysql, a warning appears:
What version of Asterisk? I ask because I have had
Reports of problems against svn trunk and svn branches
after 1.4.0 was released.
WARNING[20225]
Three sites are experiencing ~10sec period of one-way audio. This
happens several minutes into the call, and it is very intermittent
(infrequent). It does not happen on inter-office calls but only on
calls to/from the PSTN.
Occasionally, a spurt of white noise precedes the drop-out much
Hello everyone:
using chan_capi 0.7 and asterisk 1.4
Quick question:
How can I detect the number of voice channels (B channels) in use at a
given time. I'd like to call Busy if two B channels are used on my BRI
to give the calling customer a Busy signal.
Long question:
On my single-line BRI
Here are my logs:
Downloaded the no trunk version from www.asterisk.org this morning and
installed on Fedora Core 5.
Asterisk 1.4.0 built by root @ blah on a i686 running Linux on 2007-01-30
14:56:42 UTC
confserver1*CLI module unload app_cbmysql.so
Unable to unload resource app_cbmysql.so
Is anyone having problems with Cisco's 2960/3560 LAN switch? Problems
causing retries exceeded in Asterisk?
Thanks
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Eric ManxPower Wieling wrote:
Make sure you have /etc/asterisk/indications.conf
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I don't know what's happened, but now is fixed. Sorry.
On 1/30/07, Lenz [EMAIL PROTECTED] wrote:
That's what I get:
The requested URL / was not found on this server
:)
l.
On Mon, 29 Jan 2007 23:16:47 +0100, Facundo Ameal [EMAIL PROTECTED] wrote:
Hello everyone! In my humble try of
Benko wrote:
Hello!
I've upgraded from 1.2.9 to 1.2.14 recently but experience an
unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was
playing continuous on sequential extensions after a
timeout, it is restarted for every extension in 1.2.14:
;music starts
exten =
A question about Queues and Dial Plans
We are trying to set up a customer service queue. I've set up the queue
and agents who will participate. However, there's still one area I'm not
sure how to make it work. After 60 seconds, I need it to decide that no
one is available, and forward it to
On Tue, 2007-01-30 at 10:11 -0500, J. Oquendo wrote:
So I have my extensions.conf
(http://www.infiltrated.net/exten.stupidity.conf) shortened
in case someone wants to look. Has someone encountered the following?
I've racked my
brain on this for too long...
I have two contexts, day and
On Tue, 2007-01-30 at 16:23 +0200, yusuf wrote:
Hi,
I just want out find out how to do bill recon's when you send calls to a
provider. They send me
their CDR's, and when I compare it to my * CDR's, some calls are 1 second
off, either way.
How in general is it done by others?
Yusuf,
Ma wrote:
WARNING[20225] pbx.c: No application 'CBMysql'
for extension (default, 1995, 3)
I check the application, it didn't registered
CLI core show application CBMySQL
Your application(s) is (are) not registered
But I can see it use show module
I made a small mess of
More info: I've noticed that Asterisk CPU utilization has spiked to
100% for a period of 10-20 seconds.
Michael Welter wrote:
The commonality between all sites is Asterisk/Zaptel 1.4.0. The TDM04B
site started reporting this problem after the upgrade to 1.4.0.
I just encountered this last night when trying out demo apps. When a caller dialed, Asterisk consistently picked up the ring and kicked off the dial plan. However, neither side received audio - caller not hearing announcement, Asterisk not receiving DTMF. Asterisk restart "cured" this problem and
Hi people,
Necessary to record agents, and that format of the archive is as below:
queue-agent-exten-callerid-timestamp.wav
Somebody can help me?
Thanks,
Rafael
Rafael Augusto
Gerente de Suporte
Central de Relacionamento GoVoIP
http://www.junghanns.net/en/ISDNguard_produkt.html
I am no longer with the company that got frelled by ATT and/or
asterisk. However this unit would have definitely helped out. Just
disconnect when heartbeat not found.
On 1/29/07, Shane Spencer [EMAIL PROTECTED] wrote:
It was either down or
switch is layer two device and transparent to communication asterisk to
phone
Michael Welter wrote:
Is anyone having problems with Cisco's 2960/3560 LAN switch? Problems
causing retries exceeded in Asterisk?
Thanks
___
--Bandwidth and Colocation
Rob Schall wrote:
A question about Queues and Dial Plans
We are trying to set up a customer service queue. I've set up the queue
and agents who will participate. However, there's still one area I'm not
sure how to make it work. After 60 seconds, I need it to decide that no
one is available,
We have a PRI from Telepacific. Asterisk 1.2 and a Sangoma A101 T1 card.
Outgoing calls to certain toll-fee (8XX) numbers fail -- we hear ringing but
the calls are never answered. All other calls, and most toll-free numbers
are not affected. The numbers that are affected are all travel
Here is what he was getting at:
1.2.x and ztdummy and meetme all work fine.
However
Compile Zaptel 1.4.0, install, reboot
Zttool shows ztdummy as the timing device. Lsmod shows it loaded.
If you then compile Asterisk 1.4.0 it fails to compile app_meetme
My quick and dirty solution just
Hi,
I am planning couple small business installations and wader what should I
use for 2 to 6 lines a gateway or pci card?
Any comments greatly appreciated on pros and cons and brands.
Thanks,
robert
___
--Bandwidth and Colocation provided by
We have run into this problem before, especially with Airline carrier TFN's.
The deal is THEY DO NOT SEND ANSWER, we have looked at this on a ss7 level,
and most if not all major air carriers do this. I believe their try to lower
their costs.
The fix, answer the call, first thing.
IPKall
Probably a layer 1 or 2 issue.
Look at the stats for the port on the Cisco switch, there may be clues
there.
Check the speed and negotiation settings for the port and NIC, lock it
to 100MB.
Change the patch cables (DO NOT use hand-made cables)
Change the NIC in the Asterisk server
regards,
That solved the problem thank you.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin
Sent: Tuesday, January 30, 2007 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] web-meetme cbmysql not
On Tue, Jan 30, 2007 at 12:00:17PM +0100, Patrick Cervicek wrote:
Brad Templeton schrieb:
On Mon, Jan 29, 2007 at 12:05:28PM +0100, Benny Amorsen wrote:
PC When I set for Extern1/2 canreinvite=yes it works, but
PC Intern-2-Extern doesn't work because Asteisk gives out the
PC private
I don't think it really matters. I'd go with which ever is cheaper.
On 1/30/07, Robert Augustyn [EMAIL PROTECTED] wrote:
Hi,
I am planning couple small business installations and wader what should I
use for 2 to 6 lines a gateway or pci card?
Any comments greatly appreciated on pros and cons
In order to do this, I have to add a couple quick extensions to the
dial plan dynamically, so I have to be able to add my own context.
from API use Command to run the CLI command add extension
-
The fish are biting.
Get more visitors on your site using Yahoo!
On Tue, Jan 30, 2007 at 12:59:15PM -0800, chester c young wrote:
In order to do this, I have to add a couple quick extensions to the
dial plan dynamically, so I have to be able to add my own context.
from API use Command to run the CLI command add extension
But you can only add to an
PC == Patrick Cervicek [EMAIL PROTECTED] writes:
PC But then all RTP Traffic of my internal phones will go over
PC Asterisk. I want RTP to go Peer-to-Peer. == Intern-2-Intern
PC and Extern-to-Extern should go P2P and Intern-2-Extern should
PC go over Asterisk, see picture
I understand what you
When I have HylaFAX answer a call redirected to the fax extension in
Asterisk when it detects CNG, Asterisk hangs up:
Jan 30 14:32:59 VERBOSE[1098]: -- IAX2/ttyIAX0552/12 is ringing
Jan 30 14:32:59 VERBOSE[1098]: -- IAX2/ttyIAX0552/12 answered Zap/23-1
Jan 30 14:32:59 DEBUG[1098]: Ooh,
Hi, Giorgio
I'm from lima Peru, I have the same problem, and the solution was the
signaling in zapata.conf
I use :
fxsls
for
fxsks,
Regards
On 1/30/07, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
the problem was the telco line was not a pure telephone line but a mixed
one (phone +
On Tue, 2007-01-30 at 23:11 +0200, Tzafrir Cohen wrote:
On Tue, Jan 30, 2007 at 12:59:15PM -0800, chester c young wrote:
In order to do this, I have to add a couple quick extensions to the
dial plan dynamically, so I have to be able to add my own context.
from API use Command to run the
Realtime.. Realtime.. Realtime..
On 1/30/07, j [EMAIL PROTECTED] wrote:
On Tue, 2007-01-30 at 23:11 +0200, Tzafrir Cohen wrote:
On Tue, Jan 30, 2007 at 12:59:15PM -0800, chester c young wrote:
In order to do this, I have to add a couple quick extensions to the
dial plan dynamically, so I
Reload.. Reload.. Reload..
On 1/30/07, Shane Spencer [EMAIL PROTECTED] wrote:
Realtime.. Realtime.. Realtime..
On 1/30/07, j [EMAIL PROTECTED] wrote:
On Tue, 2007-01-30 at 23:11 +0200, Tzafrir Cohen wrote:
On Tue, Jan 30, 2007 at 12:59:15PM -0800, chester c young wrote:
In order to do
J. Oquendo wrote:
Eric ManxPower Wieling wrote:
Make sure you have /etc/asterisk/indications.conf
Thx that did it ...
You and everyone else seemed to be stuck on contexts. Ringback issues
have nothing to do with contexts.
BEFORE a call is answered, Asterisk just sends a message back
Just a general question on dialplan programming... I've implemented a
fairly full-featured system using dialplan code only. I've not used any
AGI for it, yet it ticks all the boxes I want it to tick (diverts,
follow-me, voicemail, dnd, outdialing restrictions, simple auto-attendant,
and
From:Gordon Henderson [EMAIL PROTECTED]Just a general question on dialplan programming... I've implemented a fairly full-featured system using dialplan code only. I've not used any AGI for it, yet it ticks all the boxes I want it to tick (diverts, follow-me, voicemail, dnd, outdialing
On Tue, Jan 30, 2007 at 10:23:09PM +0100, Benny Amorsen wrote:
PC == Patrick Cervicek [EMAIL PROTECTED] writes:
PC But then all RTP Traffic of my internal phones will go over
PC Asterisk. I want RTP to go Peer-to-Peer. == Intern-2-Intern
PC and Extern-to-Extern should go P2P and
On 1/30/07, Benko [EMAIL PROTECTED] wrote:
Hello!
I've upgraded from 1.2.9 to 1.2.14 recently but experience an
unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was
playing continuous on sequential extensions after a
timeout, it is restarted for every extension in 1.2.14:
From: Yuan LIU [EMAIL PROTECTED]
But I'm curious as to the approach others use. Is doing dialplan
coding in an AGI more efficient, or do people just do it that way
because it's easier than learning dialplan code? Or are there some
things that people think they can't do any other way?
So I'm just
Is there any chance you could contact me or give me a website to
monitor the current status of implementing multiple parking lots.
Multiple parking lots in 1.4 is something we were hoping for and would
love to see happen.
I'd be happy to help with testing/debuging. Please feel free to
contact me.
Colin Anderson wrote:
When I have HylaFAX answer a call redirected to the fax extension in
Asterisk when it detects CNG, Asterisk hangs up:
Jan 30 14:32:59 VERBOSE[1098]: -- IAX2/ttyIAX0552/12 is ringing
Jan 30 14:32:59 VERBOSE[1098]: -- IAX2/ttyIAX0552/12 answered Zap/23-1
Jan 30
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