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Configuring extensions.conf and sip.conf

January 3, 2016, 10:29 pm
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No phones and server are in the same network...
I corrected those errors but it doesn't work!
Can you post me an example of a sip.conf file and an extensions.conf file?
I think that the error is in the extensions..

Thank you

Ale

Statistics : Posted by alemono95 • on Mon Jan 04, 2016 12:29 am • Replies 4 • Views 121
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Configuring extensions.conf and sip.conf

January 3, 2016, 10:29 pm
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Hi to everyone!
I'm trying to set up asterisk to make inbound and outbound calls, without good results..
Here's my sip.conf
Code: [40]
type = friend
host = dynamic
port = 5060
context = out
secret = password
nat = yes
canreinvite = no
dtmfmode = rfc2833
mailbox = 40
deny = 0.0.0.0/0.0.0.0
permit = 192.168.1.0/255.255.255.0

[41]
type = friend
host = dynamic
port = 5060
context = out
secret = password
nat = yes
canreinvite = no
dtmfmode = rfc2833
mailbox = 41
deny = 0.0.0.0/0.0.0.0
permit = 192.168.1.0/255.255.255.0

and here my extensions.conf
Code:     [general]
    static = yes
    writeprotect = no
    autofallthrough = no

    [default]
    ; Segreteria Telefonica
    exten => 777,1,VoiceMailMain(${EXTEN@default})
    exten => 777,2,Hangup()

    ; Definizione interni
    exten => 40,1,Macro(intcall,40);
    exten => 41,1,Macro(intcall,41);

   [from-pstn]
exten=> s,1,Answer()
   exten=> s,2,Dial(SIP/${ARG1})

    [macro-intcall]
    exten => s,1,Dial(SIP/${ARG1},30)
    exten => s,2,VoiceMail(${ARG1})
    exten => s,3,Hangup()

*Note that there isn't an outbound rule, because i'm setting up the inbound rules; if you now how to setu up it help me Image *

Can you help me with this issues?
I'm new with asterisk!
Can you post the right code? I would like only to make and receive calls..

Thank you so much!
Ale

Statistics : Posted by alemono95 • on Mon Jan 04, 2016 12:29 am • Replies 4 • Views 121
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Asterisk API - call forward, speed dial, etc

January 4, 2016, 8:46 pm
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voipmuch wrote:Hi ambiorixg12,

Thanks for the reply.

We are not looking for any channel related function... I have found those details.

We are looking to get/set an user/extension call forward details.
We are looking to get/set an user/extension speed dial details.
We are looking to get/set an user/extension voicemail password.
etc..

Basically all user/extension based functions.


I'll cut out the back-and-forth.

No, Asterisk's REST API does not provide calls that allow manipulating call forward, speed dial, or voicemail password.

Cheers

Statistics : Posted by voipmuch • on Mon Jan 04, 2016 10:46 pm • Replies 4 • Views 202
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Asterisk API - call forward, speed dial, etc

January 4, 2016, 8:46 pm
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RealTime with a web interface you can allow customers limited ability to edit their own dialplan without needing a reload.

RealTime support is currently available for the following families:
sippeers
sipusers
iaxpeers
iaxusers
voicemail
musiconhold
queues and queue_members (used together for the Queue application).
extensions NOTE: The family name for RealTime extensions can be whatever you want. Please read Asterisk RealTime Extensions for more info.

Take a look at http://www.voip-info.org/wiki/view/Asterisk+RealTime

Statistics : Posted by voipmuch • on Mon Jan 04, 2016 10:46 pm • Replies 4 • Views 202
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Asterisk API - call forward, speed dial, etc

January 4, 2016, 8:46 pm
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Hi ambiorixg12,

Thanks for the reply.

We are not looking for any channel related function... I have found those details.

We are looking to get/set an user/extension call forward details.
We are looking to get/set an user/extension speed dial details.
We are looking to get/set an user/extension voicemail password.
etc..

Basically all user/extension based functions.

Statistics : Posted by voipmuch • on Mon Jan 04, 2016 10:46 pm • Replies 4 • Views 202
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Asterisk API - call forward, speed dial, etc

January 4, 2016, 8:46 pm
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I dont know exactly what you are trying to do but :

Quote:AGI is analogous to CGI in Apache. AGI provides an interface between the Asterisk dialplan and an external program that wants to manipulate a channel in the dialplan. In general, the interface is synchronous - actions taken on a channel from an AGI block and do not return until the action is completed.


Quote:AMI provides a mechanism to control where channels execute in the dialplan. Unlike AGI, AMI is an asynchronous, event driven interface. For the most part, AMI does not provide mechanisms to control channel execution - rather, it provides information about the state of the channels and controls about where the channels are executing.


If you need something more powerful take look to ARI https://wiki.asterisk.org/wiki/pages/vi ... d=29395573

Statistics : Posted by voipmuch • on Mon Jan 04, 2016 10:46 pm • Replies 4 • Views 202
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Asterisk API - call forward, speed dial, etc

January 4, 2016, 8:46 pm
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Hello,

I have searched everywhere and have not been able to find any details...

Please tell me there is an Asterisk Rest API that will allow us to get/set things like Call Forward (busy, no answer, unconditional), speed dial, voicemail password, etc...

Statistics : Posted by voipmuch • on Mon Jan 04, 2016 10:46 pm • Replies 4 • Views 202
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[BUG ? almost SOLVED] DTLS failure occurred on RTP instance

December 29, 2015, 1:12 pm
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After some input from Asterisk Jira to point me to the res_rtp_asterisk.c file, I found the following :

On my server http://forums.asterisk.org/configure detects and sets "#define HAVE_OPENSSL_ECDH_AUTO 1"

So I tried and activate the alternative code,by commenting the other code that would have been used if this was not set.
To my great joy everything worked straight away.

Currently awaiting action to be taken to fix the problem on a permanent base.
This is either a bug in CentOS or a bug in the http://forums.asterisk.org/configure detection system.

Statistics : Posted by pay123 • on Tue Dec 29, 2015 3:12 pm • Replies 1 • Views 237
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[BUG ? almost SOLVED] DTLS failure occurred on RTP instance

December 29, 2015, 1:12 pm
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Using Chrome Version 47.0.2526.106 (64-bit) (needs full https: connections since 15/dec/2015) on a CentOS 7.2 KDE desktop, when dialing a WebRTC peer (using SIPml5 API) I get the following errors in Asterisk 13.6 :

res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0x7fe8c8024178' due to reason 'missing tmp ecdh key', terminating
res_rtp_asterisk.c: RTP Read error: Unspecified. Hanging up.

An earlier bug report listed this as a problem on FireFox : https://issues.asterisk.org/jira/browse/ASTERISK-25265
It is said to be fixed in 13.6

Certificates used are purchased and fully functional.
Using the following on the server :
CentOS7.22015-11
Asterisk13.62015-10
jansson2.72014-10-02
PJSIP (pjproject)2.4.52015-08-12
sipML52.0.22015-12

05/Jan/2016 Due to no feedback from the forum, this has been reported as a bug https://issues.asterisk.org/jira/browse/ASTERISK-25659

Statistics : Posted by pay123 • on Tue Dec 29, 2015 3:12 pm • Replies 1 • Views 237
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Transfer a Call But continue to listen the caller

December 10, 2015, 12:36 pm
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Hello,
I need to know if it is possible to transfer a call but continue to listening the caller.

Here is the idea:
1) A call arrives at the extension.
2) Extension pick-up the call.
3) The call is transfered for another extensions. (with attended transfer)
4) The caller continue to listen the attended while he informs the person that the can is being transfered.

Is this possible and how could this be achived?

Thank you and regards.

Statistics : Posted by stuartalt • on Thu Dec 10, 2015 2:36 pm • Replies 3 • Views 1163
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Asterisk AMI PlayDTMF to instantiate disconnect key combinat

January 8, 2016, 10:59 am
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You do it before you start the outging leg. Incoming dials Local/ Local/ dials outgoing.

Statistics : Posted by jakehallas • on Fri Jan 08, 2016 12:59 pm • Replies 1 • Views 32
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Asterisk AMI PlayDTMF to instantiate disconnect key combinat

January 8, 2016, 10:59 am
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I'm currently having a play about with AMI in Asterisk 11. I seem to of hit a snag where utilising the 'PlayDTMF' action will go out on the channel rather than be seen as incoming DTMF.

This is a bit of an issue, I've provided my features.conf file config below:

Code: Builtin Feature           Default Current
---------------           ------- -------
Pickup                    *8      *8
Blind Transfer            #       #
Attended Transfer                 *2
One Touch Monitor
Disconnect Call           *       *99
Park Call                         #72
One Touch MixMonitor


I'm wanting to execute the *99 key combination over AMI to allow the call to disconnect. My AMI Action looks like this (developed in nodejs):

Code: ami.action({
    'action': 'playdtmf',
    'channel': e.channel, //this is the callers channel
    'digit': '*99'
}, function (err, res) {
    console.log(err);
    console.log(res);
});


I receive a JSON response which tells me everything has been queued successfully. I can also hear the DTMF tones come through the SIP Phone

Code: { response: 'Success',
  actionid: 'xxx',
  message: 'PlayDTMF successfully queued' }


I have read references to interposing a Local channel on the first leg. How would I go about interposing a local channel while I have an ongoing call?

Thanks very much in advance!

Statistics : Posted by jakehallas • on Fri Jan 08, 2016 12:59 pm • Replies 1 • Views 32
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asterisk odbc realtime sip registration

January 7, 2016, 8:29 am
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can you show an little example ?

Statistics : Posted by system32 • on Thu Jan 07, 2016 10:29 am • Replies 4 • Views 188
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asterisk odbc realtime sip registration

January 7, 2016, 8:29 am
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Friend is equivalent to setting both users and peers.

You must use device names that are distinct and translate the extension to the device name. You could use <tenant><extension> but that isn't particularly secure.

Statistics : Posted by system32 • on Thu Jan 07, 2016 10:29 am • Replies 4 • Views 188
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asterisk odbc realtime sip registration

January 7, 2016, 8:29 am
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firstly thanks for answer.

with odbc there are two parameteres sipusers and sippeers => sip.table

but how i can do this as mult-tenant.. with php ? or with database? another table configuration as view ?

and all these parameters (odbc module) are staticly and its problem.

who can tell me anything about this?

Statistics : Posted by system32 • on Thu Jan 07, 2016 10:29 am • Replies 4 • Views 188
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asterisk odbc realtime sip registration

January 7, 2016, 8:29 am
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Not possible.

You are probably confusing device and extension names. Whilst that can simplify dialplans in single tenant environments, it does work for multi-tenant environments, and its is bad security practice in all environments. This is true even with pure text file configuration. type=peer is also usually better for security.

Statistics : Posted by system32 • on Thu Jan 07, 2016 10:29 am • Replies 4 • Views 188
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asterisk odbc realtime sip registration

January 7, 2016, 8:29 am
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hello all

i need a little help

there is a asterisk box with full realtime odbc configuration with mysql.

i need create sip users with same sip names...

example :

[100]
type=friend
host=dynamic
secret=test1
context=test1
....

[100]
type=friend
host=dynamic
secret=test2
context=test2
...

all this configs are in sip mysql table .

thanks

Statistics : Posted by system32 • on Thu Jan 07, 2016 10:29 am • Replies 4 • Views 188
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Disable DTMF logging to specific call

January 8, 2016, 12:34 pm
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No.

Statistics : Posted by stuartalt • on Fri Jan 08, 2016 2:34 pm • Replies 1 • Views 8
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Disable DTMF logging to specific call

January 8, 2016, 12:34 pm
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Hello,
I know I can disable/enable full DTMF log via logger.conf., but I need to disable the DTMF logging for a specific call.
Is this possible?

This is because I'm capturing credit card information and this information cannot be written anywhere.

Any help would be appreciate.

Thanks in advance.

Statistics : Posted by stuartalt • on Fri Jan 08, 2016 2:34 pm • Replies 1 • Views 8
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dahdi install under centos 7

December 25, 2015, 5:00 pm
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Actually some man files were indeed missing. That's what these errors were about:
Code: /usr/bin/install: cannot stat ‘http://forums.asterisk.org/dahdi_registration.8’: No such file or directory
/usr/bin/install: cannot stat ‘http://forums.asterisk.org/xpp_sync.8’: No such file or directory
/usr/bin/install: cannot stat ‘http://forums.asterisk.org/lsdahdi.8’: No such file or directory
/usr/bin/install: cannot stat ‘http://forums.asterisk.org/xpp_blink.8’: No such file or directory
/usr/bin/install: cannot stat ‘http://forums.asterisk.org/dahdi_genconf.8’: No such file or directory
/usr/bin/install: cannot stat ‘http://forums.asterisk.org/dahdi_hardware.8’: No such file or directory
/usr/bin/install: cannot stat ‘http://forums.asterisk.org/twinstar.8’: No such file or directory


This is what I did to workaround this error (assuming your install dir is /usr/src/dahdi-*):
Code: cd /usr/src/dahdi-linux-complete-2.11.0+2.11.0/tools/xpp/
touch http://forums.asterisk.org/dahdi_registration.8 http://forums.asterisk.org/xpp_sync.8 http://forums.asterisk.org/lsdahdi.8 http://forums.asterisk.org/xpp_blink.8 http://forums.asterisk.org/dahdi_genconf.8 http://forums.asterisk.org/dahdi_hardware.8 http://forums.asterisk.org/twinstar.8 astribank_tool.8 astribank_hexload.8 astribank_allow.8 astribank_is_starting.8


Doing this, I was able to get through the installation (make, install & config), BUT I still don't have the init script installed. Image

Statistics : Posted by meral • on Fri Dec 25, 2015 7:00 pm • Replies 14 • Views 2194
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