Hi Jonas,
I get this error when I incorrectly set my PBX gateway AND I have a sip
peer trying to register outside (i.e.: a sip provider).
Are you sure about your sip.conf?
Giorgio Incantalupo
Jonas Kellens wrote:
Hello,
my Asterisk CLI is flooded with the following message :
[Jun 25
For codecs use CHANNEL function, but you will only get CallLegA codecs.
Without hacking Asterisk, you will not be able to get CallLegB codecs.
Patch for Asterisk 1.4.33.1 attached to get such info.
Retrieve such info with variables:
RTPAUDIOQOS
BRTPAUDIOQOS
And even more:
LEG1DATA
LEG2DATA
Hello list,
this is the setup :
analogue phone -- Grandstream GXW4008 -- Linksys WAG160N --
Asterisk-server (public)
and
Zoiper softphone -- Linksys WAG160N -- Asterisk-server (public)
When calling with an analogue phone + Grandstream GXW and also when
calling with the Zoiper softphone, we
Hi bruce,
SIPDefault.conf
#Image Version
image_version:P0S3-08-8-00
#Proxy server address
# Emergency Proxy info
proxy_emergency: 192.168.20.4
proxy_emergency_port: 5060
# Backup Proxy info
proxy_backup: 192.168.20.4
proxy_backup_port: 5060
# NAT/Firewall Traversal
nat_enable: 0
Jonas Kellens wrote:
Hello list,
this is the setup :
analogue phone -- Grandstream GXW4008 -- Linksys WAG160N --
Asterisk-server (public)
and
Zoiper softphone -- Linksys WAG160N -- Asterisk-server (public)
When calling with an analogue phone + Grandstream GXW and also when
Hello,
I also thought about echo because the Zoiper softphone is used with a
headset. But that didn't explain why the echo also appeared on the
analogue phone + gateway.
I have the same Grandstream GXW 4008 gateway with 5 analoge phones
attached in another environment and there, there are
Routers wont cause echo. In order for them to do so they would have to
store the outbound voice traffic, delay it and then mix it into the
inbound voice.
Telephones inherently cause echo. For domestic calls the audio path is
normally so short that any echo arrives back so quick the human ear
On 30 June 2010 10:28, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello,
I also thought about echo because the Zoiper softphone is used with a
headset. But that didn't explain why the echo also appeared on the analogue
phone + gateway.
It will present it self on the analogue phone when
Hello,
I stated in my first post that both ends hear an echo when one speaks to
the other...
The only place where echo cancellation is being applied is in the
Asterisk server. I have the following in sip.conf :
;-- JITTER BUFFER CONFIGURATION
Thats the jitter buffer. It has no effect on echo.
So you get echo when calling from the softphone to the analogue phone?
What about when one of those calls somewhere else?
What if they call a regular telephone number?
How do you connect in order to send calls to normal phone numbers?
Jonas
Hello,
I did not say that the analogue phone calls the Zoiper softphone or vica
versa.
Calls are made to from the Zoiper to an external number like a cellphone.
Calls are also made from the analogue phone to external numbers like an
international number in Holland...
Jonas.
On 30 June
Hi,
I had a breif telco outage with one of my sip providers.
Is there a way to add failed calls to the cdr aswell as the connected ones?
I was also thinking about having an automated process that monitored congested
calls vs Succesful ones on a carrier and weight the dial plan using
Kenny Watson wrote:
Hi,
I had a breif telco outage with one of my sip providers.
Is there a way to add failed calls to the cdr aswell as the connected ones?
I was also thinking about having an automated process that monitored
congested calls vs Succesful ones on a carrier and
Samantha,
Are you using some type of GUI ? If you send all the traffic to a specific
context in there you can set a default route to one peer and then set
exceptions for the others. For example
[from-pri]
Exten = _X.,1,Dial(SIP/${ext...@peer1)
Exten = _X61280X,1,Dial(SIP/${ext...@peer2)
Anahi,
What kind of line do you have ? POTS, PRI, SIP ? It seems like the DTMF is not
coming in correctly or you have some bad settings on your end.
- Original Message -
From: Anahi Ludueña
To: asterisk-users@lists.digium.com
Sent: Wednesday, June 30, 2010 01:17
Subject:
Micholas,
1) Do you have net=yes in sip.conf ?
2) How often are you registering with the Asterisk server ? You may want to run
ngrep (http://ngrep.sourceforge.net/) against the remote IP and see what
happens. Chances are your router is blocking it.
For ngrep you want to run something like
Hi Gareth,
The problem I have had in the past with providers is either that the registrar
is still up and its further down the line in the provider that the call is
being congestied, so the qualify doesnt work!
or that the providers registrar has issues but the rest of their services is up
so
Using standard AGI will add a fair bit of load and most of that will be
due to loading the perl or php interpreter every time it is called. Your
call volume is relativly high so I agree that whatever solution you go
for you want to make it as streamlined as possible.
Therefore I would advise
Hi Gareth thanks again for the responses!
I defiantly think I would have to run the agi on a separate server, I'll maybe
setup this in a lab. As I say the built in CDR is fine if it could include
failed calls!
I was planning to use a ratio of good/bad calls from a provider to determine
the
I'm not entirely sure I see where he implied it was. His answer refers to the
question, I want to know what is the best OS for installing Asterisk...?
I like both CentOS and Ubuntu. The next edition of the O'Reilly Asterisk book
will cover installing Asterisk on both OS's.
Leif.
Tiago Geada
Hi, do you mean what kind of extension I have? it is SIP, but from it,
everything works well...
In the SIP extension, the DTMF mode is rfc2833.
Thanks,
From: asteriskus...@dovid.net
To: asterisk-users@lists.digium.com
Date: Wed, 30 Jun 2010 13:54:50 +0300
Subject: Re: [asterisk-users] Dial
Hi, Have you tried sending the dtmf inband? I've had more success interoping
betwen different vendors with inband DTMF.
Thanks
Kenny Watson
Kenny Watson
From: Anahi Ludueña a_ludu...@hotmail.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, 30 June, 2010 12:50:23
Hi, yes, I've just tried to use the dtmf mode inband, but it doesn't work with
landline phones or cell phones...
Thanks,
Anahi Ludueña
Date: Wed, 30 Jun 2010 12:56:59 +0100
From: kwat...@geniusgroupltd.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dial options
Hi!
For codecs use CHANNEL function, but you will only get CallLegA
codecs. Without hacking Asterisk, you will not be able to get CallLegB
codecs. Patch for Asterisk 1.4.33.1 attached to get such info.
Thank you! In the meanwhile I found that with the help of the M option to
Dial (macro
What is the minimal module set required to run SIP with database CDR logging.
I compiled Asterisk from source and I obviously compiled more stuff
than I needed for VoIP and CDR logging to postgres.
Sometimes there is a long gap between Asterisk starting and devices
being able to register. sip
this can be cause if you are using an ADSL link with your remote phones .. or
maybe some 3G networks can cause that delay in the first response as the ACK
message will be late to arrive and if the delay was too high .. the call will
drop.one more thing if your remote phones are (Queue
On 06/30/2010 12:20 PM, Gareth Blades wrote:
So you get echo when calling from the softphone to the analogue phone?
From softphone to analogue phone is echo.
What if they call a regular telephone number?
Calling to a cellphone number or a fixed number on another Telco-network
: echo
Jonas Kellens wrote:
On 06/30/2010 12:20 PM, Gareth Blades wrote:
So you get echo when calling from the softphone to the analogue phone?
From softphone to analogue phone is echo.
What if they call a regular telephone number?
Calling to a cellphone number or a fixed number on
Hi!
Sometimes there is a long gap between Asterisk starting and devices
being able to register.
First you should check your DNS setup - it has been discussed many a
times on this list.
Philipp
--
_
-- Bandwidth and
Internet Telephony Service Provider = SIP provider. The company that
connects the Asterisk-server via a SIP trunk with the other networks
like GSM, analogue carriers...
Jonas.
By ITSP do you mean a SIP provider?
--
_
On 30 Jun 2010, at 13:48, Gareth Blades wrote:
By ITSP do you mean a SIP provider?
ITSP: Internet Telephony Service Provider
S
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
Hi!
The network setup is :
analogue+GXW / softphone -- Linksys WAG160N -- Asterisk server -- ITSP
-- other networks
Do it step-by-step: Take the Asterisk server out of the equation, i.e.
call the destination directly with your softphone or the Grandstream ATA
and see if that removes the
Thanks a lot.
-Bruce
On Wed, Jun 30, 2010 at 4:55 AM, Emanuele Carbone carbe...@gmail.comwrote:
Hi bruce,
SIPDefault.conf
#Image Version
image_version:P0S3-08-8-00
#Proxy server address
# Emergency Proxy info
proxy_emergency: 192.168.20.4
proxy_emergency_port: 5060
# Backup Proxy
The DNS setup itself is fine. The sip module just seems to take too
much time to load. My modules.conf uses autoload=yes and it seems that
many unwanted modules are loaded before sip itself starts.
On 30 June 2010 13:52, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
Jonas Kellens wrote:
Internet Telephony Service Provider = SIP provider. The company that
connects the Asterisk-server via a SIP trunk with the other networks
like GSM, analogue carriers...
Jonas.
By ITSP do you mean a SIP provider?
Thats where I believe the problem lies. You are
Gareth,
multiple users/SIP-accounts use this asterisk server from many
locations. Like I said: in another location with a similar setup, there
are no echo-complaints on received or made calls.
If you say that it has nothing to do with the Cisco-router, I don't
really know what to go looking
Hello list,
I notice on the wiki that it is possible to execute a macro or a gosub
within the queue-command in asterisk 1.6.x
1. Does this mean the macro/gosub is executed everytime a queued call is
answered by a queue member ?
2. I'm using asterisk 1.4.30. Is there a backport or other way
Try the SIP phone. If it is better then you might try looking to see if
there are any echo cancelation settings on the softphone or analogue
adapter you can change. Try turning echo cancelation off aswell since if
there are two running they can interfere with each other and make the
situation
Hi,
I have installed Asterisk 1.6. I have to configure Asterisk as a Video
Conferancing purpose. What package I need to configure and what steps I
need to follow to configure in dialplan to authenticate user.
Regards,
Hiren Mistry
--
Will turning off the jitter buffer affect the quality of the other calls ??
jbenable = no
I must say I'm not really into these jitter-settings in asterisk. I made
jbenable=yes as it can do no harm...
Jonas.
On 06/30/2010 04:24 PM, Gareth Blades wrote:
Try the SIP phone. If it is better
Yes it gets called when the call is connected to a queue member.
In version 1.4.x you can execute an AGI instead of a sub or macro.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 30, 2010, at 7:20 AM, Jonas Kellens wrote:
Hello list,
I notice on the wiki
The harm in any of these settings is environmentally controlled. What
does no harm in one setup can be a deal breaker on a smaller machine or
slightly different technology. How harmful or harmless jbenable is depends
on your hardware and what your other settings are.
_
From:
This gives you some flexibility and change-proofing that a back-port will
not. Since gosub is a depreciation candidate, you can use the AGI to
either run the macro or do the macro functionality internally. I'm a HUGE
fan of AGI, but keeping things in the dialplan is a better option when you
can.
Yes if you have a link where there is a lot of jitter it may affect the
call quality. I would try turning it off to see if it cures the problem
and if it does then you can restore the setting and implement a workaround.
Jonas Kellens wrote:
Will turning off the jitter buffer affect the quality
On Wed, Jun 30, 2010 at 8:40 AM, bruce bruce bruceb...@gmail.com wrote:
Thanks a lot.
-Bruce
On Wed, Jun 30, 2010 at 4:55 AM, Emanuele Carbone carbe...@gmail.comwrote:
Hi bruce,
SIPDefault.conf
I think you need one of the newer XML config files for the 7965. I have an
example that
Hi Paul,
On Sat, Jun 26, 2010 at 1:33 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Sat, Jun 26, 2010 at 7:33 AM, Deepesh D deep.d2...@gmail.com wrote:
Is it possible to do this action on hook flash?
Currently no. You would need to add logic to the channel driver. Or
use DTMF to
Hi,
Thanks, I thought I could find out about that without installing 1.6, but in
the end I did install it on a test server and it answered a few questions.
One thing though: I can park calls, in separate private lots, but I can
never pick them up again. I have context = some_context defined
On Wed, Jun 30, 2010 at 8:50 AM, Frank Church voi...@googlemail.com wrote:
The DNS setup itself is fine. The sip module just seems to take too
much time to load. My modules.conf uses autoload=yes and it seems that
many unwanted modules are loaded before sip itself starts.
You can stop
Hi All,
I installed a2billing with asterisk FreePBX . I can able to login and make
a call with FreePBX but
when i am using the users which is created in a2billing the call was not
established . I know somewhere i missed
the configuration please any one help me to resolve this issue . Thanks
Actually, I should simply have tried. I did need to set
CHANNEL(parkinglot). I may have some more questions, but at least it's
working right now, and use my own custom extension to pickup the calls. So
basically I don't need to (or even can!) include the parking context, I
need to setup the
Hi people,
we have some extensions which are included in the IVRs and/or queues.
Everything works fine, but the calls done from these extensions are hang up
after 30 o 35 seconds. If they are not included in the IVR or queues, the calls
are performed well.
Do you know if there is something
Taking my first steps into AGI then :
[r...@asterisk agi-bin]# cat sample.agi
#!/usr/bin/php -q
?php
$MYSQLSERVER2=localhost;
$MYSQLUSER2=user;
$MYSQLPASSWD2=passwd;
set_time_limit(30);
require('phpagi/phpagi.php');
$agi = new AGI();
$db=mysql_connect($MYSQLSERVER2, $MYSQLUSER2,
Sounds like you are getting a dial without bridge asterisk dials x and
make the connection, but because the bridge doesnt happen for what ever
reason, the call disconnects like no one ever answered.
_
From: asterisk-users-boun...@lists.digium.com
1. (personal preference) I wouldn't use PHP
2. that out of the way, I comment out the AGI stuff and run my AGI's
from bash to make sure the non AGI stuff is happy.
3. the AGI seems to be ok here, I'd make sure my SQL stuff is good.
_
From:
Here is a simple AGI using cagi that creates a user event when a call is
connected with a queue member:
#include stdio.h
#include stdarg.h
#include cagi.h
int main (int argc, char *argv[]) {
AGI_TOOLS agi;
AGI_CMD_RESULT res;
intrtn;
char
Here is my only question left about parkinglots in 1.6. How does the
parkinghints=yes parameter work?
I've tried using core show hints , but there are never any hints. Even
when a call is actually parked in the correct parking lot.
Any tips?
Mike
From:
Danny,
1. I only know php, I'm no programmer
3. the query works in normal PHP.
Can I debug to know what's going wrong ?
Jonas.
On 06/30/2010 05:42 PM, Danny Nicholas wrote:
1. (personal preference) I wouldn't use PHP
2. that out of the way, I comment out the AGI stuff and run my
In 1.4 you set up the lots you want to monitor as hints; not sure how this
works in 1.6.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, June 30, 2010 11:24 AM
To: 'Asterisk Users Mailing List -
I cut and pasted the PHP from your OP and ran it from a shell. When Table
AstDB in Database Asterisk contains context foobar, here is the output
$php jonas.php
VERBOSE query is: SELECT vmcontext FROM AstDB WHERE ID='40' 3
VERBOSE VMCONTEXT is: Array 3
I know, I've done this with 1.4 manually with hint extensions. But in 1.6
there is a parameter called parkinghints=yes that is supposed to set them up
automatically. It certainly doesn't seem to be doing anything for me.
Thanks,
Mike
From:
Thank you for your help. It works now. So these were my first steps into
AGI...
Jonas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
Warren Selby wrote:
On Wed, Jun 30, 2010 at 8:50 AM, Frank Church voi...@googlemail.com
mailto:voi...@googlemail.com wrote:
The DNS setup itself is fine. The sip module just seems to take too
much time to load. My modules.conf uses autoload=yes and it seems that
many unwanted
Good afternoon list.
I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi. But
after running the script, it just returns me 0 (true). Thus:
-- SIP/213-0019AGI Script check.agi completed, returning 0
I tried putting the lines return false; or return 1; but did not
Add void exit (1); to the end of your php script (where you have return 1).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo Lang
Sent: Wednesday, June 30, 2010 1:40 PM
To: Asterisk Users Mailing List -
Thanks Danny, but I don't know what I should do to fix it...
Could you help me?
Anahi Ludueña
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Wed, 30 Jun 2010 10:33:31 -0500
Subject: Re: [asterisk-users] Problem with extensions in IVR and queues
It did not work. Returned the broken pipe error. Obs I using phpagi.
Thanks,
Rodrigo Lang.
2010/6/30 Danny Nicholas da...@debsinc.com
Add void exit (1); to the end of your php script (where you have return
1).
--
*From:*
Can you post the dialplan section and CLI output from one of these calls?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Wednesday, June 30, 2010 2:05 PM
To: asterisk-users@lists.digium.com
Subject: Re:
Can you post the script?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo Lang
Sent: Wednesday, June 30, 2010 2:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Return
Hi Danny. I solve the problem. I put exit (return); where return is equal
to ${AGISTATUS} text. Example:
exit(SUCCESS);
exit(FAILURE);
exit(HANGUP);
This application sets the following channel variable upon completion:
AGISTATUS The status of the attempt to the run the AGI script
:1] GotoIf(SIP/9050-001185aa,
1?check) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [...@macro-record-enable:4] AGI(SIP/9050-001185aa,
recordingcheck|20100630-154030|1277926830.37214) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- AGI
/9050-001185aa,
record-enable|4010|Group) in new stack
-- Executing [...@macro-record-enable:1] GotoIf(SIP/9050-001185aa,
1?check) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [...@macro-record-enable:4] AGI(SIP/9050-001185aa,
recordingcheck|20100630-154030|1277926830.37214
On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote:
Thank you Andrew,
I will check it out. We are currently running 1.4.
-Matt
On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote:
Remote Party ID in trunk, it works There are hacks for other
We are doing hardware tests with recent dahdi-2.3.0.1 and both
asterisk-1.4.33.1 and asterisk-1.6.2.8. Recently, we have noticed that
whenever an ISDN port is in RED alarm (unsynchronized), we get a stream
of warnings in /var/log/asterisk/full that look like this:
[Jun 30 17:38:41]
Hello
I'm taking a look at how to write scripts to be called from the
dialplan, and saw pbx_lua mentioned.
I'd like to know more about this feature, such as what the difference
is with just calling the Lua interpreter through AGI (same difference
as between php-cgi and mod_php?), whether it's
On Sat, 26 Jun 2010 17:53:27 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
Dial an extension that answers and stores to voicemail, say blah blah into
it for one minute and check the resulting file size. divide it by 60 and
you'll get a good estimate of the number of bytes per
On Sun, 13 Jun 2010, Tilghman Lesher wrote:
I would generally suggest something a little more deterministic (where
101 is your extension):
$ echo '101This is a salt' | sha1sum
22c3c098bfc2289396af84ecfb1ab77419a6537e
Aside from being 8 characters longer, why do you prefer sha1sum to
On Thu, 1 Jul 2010, Gilles wrote:
I'm taking a look at how to write scripts to be called from the
dialplan, and saw pbx_lua mentioned.
I'd like to know more about this feature, such as what the difference is
with just calling the Lua interpreter through AGI (same difference as
between
On Wed, Jun 30, 2010 at 6:10 PM, CunningPike cunningp...@gmail.com wrote:
On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote:
Thank you Andrew,
I will check it out. We are currently running 1.4.
-Matt
On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com
Dear all,
I want to retrieve the value from Contact header and from From header
which is 0345001280 from the following two lines:
Contact: sip:0345001...@123.50.217.143 sip%3a0345001...@123.50.217.143
From: 99
sip:0345001...@113.34.235.106sip%3a0345001...@113.34.235.106
;tag=as191896a1
You might take a look at the SIPHEADER function which can return specific SIP
headers.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 30, 2010, at 7:36 PM, kamrun nahar bina wrote:
Dear all,
I want to retrieve the value from Contact header and from From
I am sure this is simple, but have been struggling. I want to create a
call file that dials out a particular Dahdi channel to enable call
forwarding on a POTS line. I have this in extensions.conf:
[custom-callfwd]
exten = s,1,Answer
exten = s,n,Dial(DAHDI/4-1/*717157750)
exten =
On Thu, 1 Jul 2010, Jeff LaCoursiere wrote:
I am sure this is simple, but have been struggling. I want to create a
call file that dials out a particular Dahdi channel to enable call
forwarding on a POTS line. I have this in extensions.conf:
[custom-callfwd]
exten = s,1,Answer
exten =
On Wednesday 30 June 2010 18:38:51 Steve Edwards wrote:
On Sun, 13 Jun 2010, Tilghman Lesher wrote:
I would generally suggest something a little more deterministic (where
101 is your extension):
$ echo '101This is a salt' | sha1sum
22c3c098bfc2289396af84ecfb1ab77419a6537e
Aside from
On Wednesday 30 June 2010 13:39:57 Rodrigo Lang wrote:
Good afternoon list.
I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi.
But after running the script, it just returns me 0 (true). Thus:
-- SIP/213-0019AGI Script check.agi completed, returning 0
I
Dear Jim Dickenson.
Ｔｈａｎｋｓ for you mail. I have got the solution.
Thanks
Nahar
On Thu, Jul 1, 2010 at 11:45 AM, Jim Dickenson dicken...@cfmc.com wrote:
You might take a look at the SIPHEADER function which can return specific
SIP headers.
--
Jim Dickenson
mailto:dicken...@cfmc.com
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