hi,
I have the following setup:
PSTN - cisco-as5300 - asterisk - sip-phones
the as5300 sends a remote-party-id-header with privacy information
depending on what is coming from the PSTN.
for example the as5300 is sending as the remote-party-id header:
sip:[EMAIL
Hi All,
I'm wondering if anyone can share any info on why I frequently get peer
timeouts like below, and receive 489 messages from another A*k server on
the same LAN.
For the peers, we've one L2 switch. ICMP is 1ms. The CPU of the main
A*k server is usually 2%. So I can't see why we'd get
Hello everybody, when I run make config I have this error:
install: cannot stat `init.asterisk': No such file or directory
make: *** [config] Error 1
I don't understand.
For what is make config? to put on /etc/init.d/?
Thanks for all
___
--Bandwidth
Hi,
I'm looking fore a way to play a dial tone before our IVR platform
answered the phone line.
I want to use for the following reason:
When a caller calls our Voice Platform, the call will direct dial out to
a number.
I want to dial out before the inbound call is answered.
But now
Then i think u should use Atis's idea of using transfer_context
variable...you should set it inside your dialplan and it should be the
first thing you do in your dialplan.
Are you sure there is no leak in your dialplan, because asterisk cant
transfer your caller to an extension it cant find.
For what is make config? to put on /etc/init.d/?
in adition make config for start asterisk during startup...
but for debian you still need to run the command bellow after make config
update-rc.d asterisk defaults
thanks
atik
___
--Bandwidth and
Hi, if you are using asterisk 1.2 and OS is debian then modify in the
Makefile under asterisk folder after you unzip ..
original line
..
config:
if [ -d
Hello, I have some problems with mISDN.
I can't send or receive call from the Billion ISDN card
Mi configuration files are thoose:
extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten = 101,1,Dial(SIP/101,30,Ttm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,Ttm)
exten =
180 w/ SDP is valid, although not ideal. 183 w/ SDP is a better choice for
early-media. The SIP specifications do not dictate what a UAC should do when
it receives 180 w/ SDP. It depends on the policy implemented in the UAC.
As far as Asterisk is concerned, it could treat 180 w/ SDP same as 183
I use ENUM lookup in my dialplan before sending calls through my PSTN trunk.
One problem arises... When ENUMLOOKUP finds an SIP contact for that e164
number, Asterisk dials that contact, but when the remote server that
should answer the call is down, or the IP link is down for some reason,
the
Two points,
first (I believe from many previous threads, and viewing source code
) you must answer a call to place audio on the channel.
second, depending on the type of access being used by the originator of
the call, the carrier will not pass audio on the channel back to the
originator
Have you tested the actual throughput on the link? What's it max out...
What kind of latency are you seeing as it gets loaded.
Can you do a local call to your own internal network (softphone or
hardphone) as the system gets loaded and play the same file. Do you
have quality issues. This will
I want to execute the function Chanspy, does any one know how can I execute
a function throw the console or throw AMI, AGI...
I´m making a dial plan throw AMI, does any one know how to execute CHANSPY
throw AMI?.
Please Help.
Thanks.
___
--Bandwidth
I want to execute the function Chanspy, does any one know how can I execute
a function throw the console or throw AMI, AGI...
I´m making a dial plan throw AMI, does any one know how to execute CHANSPY
throw AMI?.
Please Help.
Thanks.
___
--Bandwidth
Enable verbose logging for the asterisk log
Set verbose level to 4
Review the log file for anything that looks like a phantom call.
There should be enough information to get some idea of why this is
happening.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
With what command can I execute chanspy throw Asterisk Console.
THANKS.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On 6/19/07, Carlos Garcia Mujica [EMAIL PROTECTED] wrote:
With what command can I execute chanspy throw Asterisk Console.
THANKS.
You are less likely to be answered if you spam the list.
___
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hi
i'd like to write a simply application in php with phpAgi that:
- connect to Asterisk
- call an external number using a Zap channel
- play a message
here is some code:
?php
$asm = new AGI_AsteriskManager();
if ($asm-connect()) {
$asm-Originate(Zap/g1/1,number,default,1);
/*
play
An alternative to this method might be to create a call file and place
it in the spool. Have it either dial and connect the caller to an
extension that plays that sound, or just execute that sound itself.
Right now, we use this functionality for a server scanner. When it
detects a particular port
Thanks man...
So far everything worked as expected...
How can I make internal calls stay within the PBX. For example, when one
SIP-Friend tries to call another SIP-Friend without sending the call out
on Trunk and receive it back. Same like dialing from one extension
number to another
I've done this many times, also used the .call files. If you don't need
your application to initiate the call the .call files are the better way
to go, otherwise it's a bit too much file management overhead.
Here's some working code on our end. In this case the Channel is
actually a context
Vadim Berezniker wrote:
Enable verbose logging for the asterisk log
Set verbose level to 4
Review the log file for anything that looks like a phantom call.
There should be enough information to get some idea of why this is
happening.
-Original Message-
From: [EMAIL PROTECTED]
sorry, can you post me an example of a call file?
thanks
On 6/19/07, Rob Schall [EMAIL PROTECTED] wrote:
An alternative to this method might be to create a call file and place
it in the spool. Have it either dial and connect the caller to an
extension that plays that sound, or just execute
Channel: Zap/g2/5052
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: internal
Extension: 8855
Priority: 1
Application: Festival
Data: This, is, a, message, from, UCS, Call, One, Server Status. The,
server, %s, is, currently, not, responding, on, ports, 80.
Once you have this in a file with a
On Tue, 2007-06-19 at 09:36 -0400, Nitesh Divecha wrote:
Thanks man...
So far everything worked as expected...
How can I make internal calls stay within the PBX. For example, when
one
SIP-Friend tries to call another SIP-Friend without sending the call
out
on Trunk and receive it
David Boyd wrote:
Two points,
first (I believe from many previous threads, and viewing source code
) you must answer a call to place audio on the channel.
second, depending on the type of access being used by the originator of
the call, the carrier will not pass audio on the channel
Since its all part of a program I would do it using the AGI like
Christopher was talking about. However, I think this would be a 2
program issue. First, you would have a program that would check a
database or whatever to see who is late and make the call to the
supervisior. That call I would
Finally, this is what I was looking for... to generate a call.
I have been working on my Time Clock application, where an employee will
call into the system using his cellphone to clock in and clock out his
hours. And it works perfect...
Now I was looking for an option where or if an employee
many thanks to all.
I am interested to originate the call using phpAGI with this code.
?php
require('PHPAGI/phpagi-asmanager.php');
$callid = 'Somebody';
$asm = new AGI_AsteriskManager();
if($asm-connect())
{
$call = $asm-send_request('Originate',
Yes Lee, he could, however he doesn't want to answer the call until the
call has been completed on the outbound leg.
Dave
On Tue, 2007-06-19 at 10:26 -0400, Lee Jenkins wrote:
David Boyd wrote:
Two points,
first (I believe from many previous threads, and viewing source code
) you
I can't make a misdn show channels on the CLI.
It looks like the mISND isn`t registered.
thanks for all
2007/6/19, Josu Lazkano [EMAIL PROTECTED]:
Hello, I have some problems with mISDN.
I can't send or receive call from the Billion ISDN card
Mi configuration files are thoose:
That should be pretty easy to do with a .call file. The context that
you drop your called party off to will play the sounds and do the
transfer. So really you need to concentrate on creating that context,
the .call files are very easy to generate.
Nitesh Divecha wrote:
Finally, this is
At 10:37 AM 6/18/2007, you wrote:
Can anyone recommend any wholesale SIP termination providers that
will automatically charge a credit card? Most seem to want upfront
payment and a credit balance but that sucks when you have to watch
it like a hawk to make sure the balance never hits zero.
This should help:
http://www.voip-info.org/wiki/index.php?page=Asterisk+channels
Channel can be anything that is valid on your system. I use Local
because it allows me to better control the outbound call.
nik600 wrote:
many thanks to all.
I am interested to originate the call using phpAGI
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Monday, June 18, 2007 5:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 180 Ringing with SDP
On Mon, 18 Jun 2007,
On Jun 16, 2007, at 4:37 PM, Tzafrir Cohen wrote:
On Sat, Jun 16, 2007 at 08:55:24PM +0100, Senad Jordanovic wrote:
Brett Crapser wrote:
On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote:
Paul Hales wrote:
GUI bad! CLI good!
PaulH
Really...?
So explain why every major PBX
On Tue, 19 Jun 2007, Douglas Garstang wrote:
Tell that to level 3. :)
Is Level3, or, more precisely, the implementors of the SIP stacks for
the vendors of Level3's equipment, of a different persuasion?
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel:
In his defense, when I first posted to the list, I wish I had found some
instructions somewhere that said messages might take at least a half
hour to reach the list, so _don't_ double post! Hopefully this upgrade
to the list software will reduce some of the strain on its server.
William Moore
Tom Rymes wrote:
On Jun 16, 2007, at 4:37 PM, Tzafrir Cohen wrote:
On Sat, Jun 16, 2007 at 08:55:24PM +0100, Senad Jordanovic wrote:
Brett Crapser wrote:
On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote:
Paul Hales wrote:
GUI bad! CLI good!
PaulH
Really...?
So explain why
At 22:29 6/18/2007, Deepak Naidu wrote:
Hi,
I have Asterisk-1.2.18 install with FreePBX more than 75
extnsion, daily I come accross an issue try resolving them its
either user learning curve or my ignorance.
But, I dont know what to say regarding this issue.
I have my Dial Plan for
Thanks man,
Is there any other way without dialing 9... it will be kinda pain for a
customer to dial 9 every time and plus they need to know also...
Is there any intelligent way to identify? if its a local SIP then don't
route to Trunk else route to Trunk.
Cheers,
Nitesh
Guillermo Salas M.
Is there any info on how to create .call files with some examples? And
where to place this file? And how to initiate it..?
Thanks man...
Cheers,
Nitesh
Christopher Robinson wrote:
That should be pretty easy to do with a .call file. The context that
you drop your called party off to will
Hi All,
Is there a way to have A*k record calls on-the-fly, at the users
request? i.e. a possible scenario:
Party A calls Party B
During the call, Party A wants to start recording the call, so presses
*, A*k announces recording.. and starting MixMonitor to a file.
Once the call is finished,
I made some progress on this issue...
It seems that I now see logs of DTMF for IAX/SIP outbound calls, but not
for internal SIP calls (aka meetme).
Not sure why.
A.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian
Marsh
Sent: 05 June 2007 18:40
A cursory interpretation of the RFC suggests that 180
Ringing is a
message designed solely to convey ringback, and that it is
the payload
of the 183 response that may be used to convey additional details
about the nature of the call's progress. Therefore, a 180
would be an
On 6/17/07, Nick Seraphin [EMAIL PROTECTED] wrote:
Yes... 1.5 cents per dip... you prepay the fees... and they deduct from
the prepaid amount. You can start with $5.00 which seems like a low-risk
to check it out at least.
The CLEC I use is more expensive that that for CNAM, and they want
Adrian Marsh wrote:
Hi All,
Is there a way to have A*k record calls on-the-fly, at the users
request? i.e. a possible scenario:
Party A calls Party B
During the call, Party A wants to start recording the call, so presses
*, A*k announces recording.. and starting MixMonitor to a file.
If you do not dial 9 then there will be a conflict between internal
extensions and external phone numbers.
How would Asterisk determine if you are dialing extension 458 or
458-1234? It cannot. Asterisk would have to wait for a timeout when
dialing the extensions. If you force users to
Nitesh Divecha wrote:
Is there any info on how to create .call files with some examples? And
where to place this file? And how to initiate it..?
Thanks man...
Cheers,
Nitesh
Christopher Robinson wrote:
That should be pretty easy to do with a .call file. The context that
you
What is the point of line lights on the phone?
The lights are so you would know when the KSU is out of lines.
With Asterisk if the system is setup right it should never run out of
lines to use.
Best regards,
Al Bochter
Bochter Services
Given that Asterisk is modeled on, in the telephone industry, an
obsolete PBX design, without many of the modern day hybrid features, and
only recently has any effort been made to provide buttons and lights for
lines ( Is that yet working in 1.4??) one would have to do some very
careful number
If you want to look up phone numbers try and its FREE
http://www.asteriskextras.com/index.php?option=com_contenttask=viewid=21Itemid=2
Best regards,
Al Bochter
Bochter Services
--
Did you check your US Greenbacks for GOLD Today?
Not so.
The point of BUTTONS and LIGHTS is for users. Remember them?
Press a button to answer a call under a flashing light.
Press a button to grab a call on hold under a light flashing at a
different rate
Press a button to place an external call.
Too many more reasons to enumerate.
Also,
When making an outbound call, the outbound peer return a 301 forwarded with
URI to other domain, but asterisk think it's a local domain and
try to look it up from extension.conf.
How to configure so that a 301 forwarded with URI from other domain thinks
it's outgoing to another proxy? thanks!
In the features.conf file, under featuremap, add automon = *1
Then in extensions.conf...
[general]
DYNAMIC_FEATURES=automon ; Auto Monitor Calls by pressing *1
now if you press *1 while on a call, it will begin recording. Press *1
again and it will complete the recording.
Rob
Drew Gibson
I'm 192.168.1.250
vopilot74*CLI -- Got SIP response 301 Forwarded back from
192.168.1.120
-- Got SIP response 301 Forwarded back from 192.168.1.120
vopilot74*CLI -- Now forwarding SIP/192.168.1.150 to
'Local/[EMAIL PROTECTED]' (thanks to SIP/sip_proxy-out-081c0d10)
[Jun 19 11:47:51]
Lucian,
I am not sure that Asterisk has that capability, since it's not itself a
proxy or a router.
One thing you might try is putting the URI ([EMAIL PROTECTED]) straight
into the dial plan and seeing what happens. SIP URIs can be alphanumeric.
Otherwise, not sure that you can handle this
We have an Asterisk box setup and are ready to start offering VOIP to our
Wireless and DSL customers. Who do you guys recommend for DID's, 911, Long
Distance, etc.? We are looking for a solution to use where we can provide
the Asterisk box and they provide everything else.
Thanks,
We use broadwing and paetec for most of our pri stuff. Paetec is a bit
better with their call detail, but both seem to provide steady service.
It depends on your location, pricing, etc though.
Rob
Duracom Lists wrote:
We have an Asterisk box setup and are ready to start offering VOIP to
our
In this scenario, how to make asterisk send the invite to
SIP/[EMAIL PROTECTED]:5064
instead of
Local/[EMAIL PROTECTED]
Thanks
-- Forwarded message --
From: Lucian Romi [EMAIL PROTECTED]
Date: Jun 19, 2007 11:52 AM
Subject: Fwd: Urgent. When the peer returned a 301 forwarded,
I have this in my dialplan...
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,Answer
exten = 5000,n,Wait(1)
exten = 5000,n,NoOp(${CALLERID(num)})
exten = 5000,n,Playback(tt-monkeys)
which, when I dial 5000, executes this...
== Parsing
Thanks Alex.
I suspect in this scenario, asterisk will treat everything as local dial
plan. I tried to modify the domain settings in sip.conf, but I haven't
figure out how to make it recognized this as a outgoing URI yet.
If I configure the local extension dialplan forward to this URI, it
On 6/19/07, Lucian Romi [EMAIL PROTECTED] wrote:
In this scenario, how to make asterisk send the invite to
SIP/[EMAIL PROTECTED]:5064
instead of
Local/[EMAIL PROTECTED]
Thanks
Asterisk isn't a SIP proxy. As such, you need to use some workarounds
to make what you want to do work. One way
Thanks everyone for the input...
In real world we can not ask the customers to dial 9, if they want to
call another SIP user... and trust me its confusing for a customer
also... meaning when to dial 9 and when to not...
We have a custom proprietary system which does this part very well...
Yes. That maybe true.
Can Asterisk do sip REFER blind transfer.
My configure looks like this
in sip.conf
[sip_proxy-out]
type=peer ; we only want to call out, not be called
host=192.168.1.180
port 5064
in extension.conf
[default]
exten = 519,1,Transfer(SIP/sip_proxy-out)
It
Thanks.
I tried to get ideas from this setup and I can only get 302 Move temporary
for Transfer.
But I expect SIP REFER, asterisk can or cann't do it? I'll really appreciate
if anybody can tell me this.
On 6/18/07, Alex Balashov [EMAIL PROTECTED] wrote:
Lucian,
Perhaps this can be of
In a2billing just change the 9 to what you need it is right in the conf
file.
Best regards,
Al Bochter
Bochter Services
--
Need to call me use our web phone at the link below
http://www.bochterservices.com/voip/iaxphone.php?cn=250
pleease post your context exactly for the exten 5000 as u have it in live
system.
On 6/19/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I have this in my dialplan…
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,Answer
exten = 5000,n,Wait(1)
exten =
I figure it out.
This is because sip_channel is in dialing process. Oh man, how can you do
blind transfer when a call is not establish yet.
so I added
exten = 511,1,Playback(demo-abouttotry)
; Let them know what's going onexten = 500,n,Transfer
exten = 511,n,Dial(SIP/sip_proxy-out)
now it
What does the output of 'show dialplan start' look like?
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Douglas Garstang
Sent: Tuesday, June 19, 2007 3:20 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Ex-Girlfriend
So, I am not sure whether its a zaptel issue. It have TE212P card which has
echo based hardware cancellor.
--
Deepak
Deepak Naidu [EMAIL PROTECTED] wrote: Hi,
I have Asterisk-1.2.18 install with FreePBX more than 75 extnsion,
daily I come accross an issue try resolving them its
Has anyone succesfully tried using ChanSpy on SIP channels with the latest
Asterisk 1.4? I tried ChanSpy(SIP/5060) to monitor SIP extension 5060 and the
console displays, Monitoring Sip/5060, but I don't hear anything. I am able to
monitor Zap
1.2.10
On 6/19/07, Doug [EMAIL PROTECTED] wrote:
At 02:08 6/17/2007, Rilawich Ango wrote:
HI all,
Recently, I got the following message from CLI and finally the
asterisk will hang. Anyone can tell me how to fix the problem or why
it will happen.
Thanks.
Version?
Also:
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