Can ChanSpy Offer Private Dialog?


I have a customer who needs to use chanspy to :

1. Interrupt a call and whisper to the local extension

2. Allow the person on the local extension to reply to the interrupter, but not have the remote party hear it.


John is on the phone with a customer. Doris receives an urgent call from John's mother. Doris uses to break into the conversation John is having with a customer. She tells John: "Your mother is on the phone. It's urgent." John replies to Doris: "OK, I'll call her in two minutes." The customer on the remote end heard none of this.


Asterisk Users 49 minutes ago 0 Answers

Call Forwarding In Asterisk


Hello Group,

I have a requirement to dialout some external number, once the call is answered the same has to be forwarded to an Internal Queue.

Please help me.

I have tried calling with two SIP end point forwarding , even that is not working,

My dial plan line is , Dial(SIP/19201/19202,300)

Asterisk Users 9 hours ago 1 Answer

Setvar Not Executed When Call Comes In Via Registry



I have a line like

register =>

in sip.conf, and a corresponding stanza (note especially the final setvar):

[trunk-sipgate] type=peer qualify=yes insecure=invite language=de dtmfmode=rfc2833 fromuser=1yyyyyyy1 defaultuser=1yyyyyyy1 secret=xxxxxxxxxxxx context=in-trunk-sipgate session-timers=accept allow=!all,alaw,ulaw,g726 setvar=FOO=BAR

If I 'sip show peer trunk-sipgate', the variable FOO is there.

I also have a stanza for my local SIP phone, e.g.

[0020fe8200de] ; abbreviated md5secret=abcdabcdabcdabcadbcdabcadbcdabcd context=in-martin setvar=DEFAULT_ORIGIN=11

When I make a call with this phone, the dialplan has access to ${DEFAULT_ORIGIN}.

However, when a call comes in through the sipgate trunk and gets routed to the in-trunk-sipgate context, the ${FOO} variable is not set and thus not available from the…

Asterisk Users 1 days ago 1 Answer

Single SIP User On Multiple Location


*Hello group! *

*Now I’m trying to solve following problem. I have a requirement that each employee should have **SIP phone at home, SIP phone in office, cell phone with same user. *

*I want all those 3 phones to be “one extension”. So, if someone calls our company number and dials my extension - I’d like 3 phones to ring at the same time.*

*e.g. Extension 555 for all the places and when anyone dial the extension 555 then it should ring at all the places simultaneously and user can pick any extension as desired.*

*Help Please.*

Asterisk Users 1 days ago 2 Answers

Problem With Cisco CUBE When Dialling Into Asterisk 13 Server



This is a problem with my Cisco CUBE (2811), so apologies for this being kind of off-topic. It is acting as a border for my Asterisk 13 server though :)

Rather than re-type the details of my problems, I have a post in the Cisco community with running-configs and various debugs attached. I'm drawing blanks as to my problem so I am reaching out wherever I can to try resolve this.

Thanks in advance,

Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)

Asterisk Users 3 days ago 0 Answers

AMI 'meetme List Concise' Hanging


I have a problem with AMI 'meetme list concise' hanging. I'm running Asterisk 11.15.1, and PHPAGI 2.20.

The AMI stuff is in the file phpagi-asmanager.php, which says it is v 1.10 2005/05/25.

Here's the relevant snippet of my PHP code:

// get list of conferences if ($debug) { echo 'getting list of conferences' . PHP_EOL; } $response = $ami->command("meetme list concise"); if ($debug) { echo 'got list of conferences' . PHP_EOL; }

I never get to 'got.'

1) Is this the current / best version of the PHP AMI interface?

2) Is this a 'known issue' with either Asterisk 11.15.1 or PHPAGI 2.20?

3) Can I set…

Asterisk Users 3 days ago 0 Answers

Escaping Parameter For ODBC Function


Hello, I just noticed a weird behavior when using ODBC functions. If the content of any of the paramter has a "=" inside, then the function is not processed correctly by asterisk.

Let's take for example the following ODBC function in func_odbc.conf

[LOG_SMS] dsn=asterisk1,asterisk2 synopsis=Log the route of a SMS writesql=INSERT IGNORE INTO sm_smslogs(sm_te_id,sm_date,sm_direction,sm_sourceraw,sm_destraw,sm_from,sm_to,sm_body,sm_fullresult,sm_response) values ('${ARG1}',NOW(),'${ARG2}','${ARG3}','${ARG4}','${ARG5}','${ARG6}','${SQL_ESC(${ARG7})}','${SQL_ESC(${ARG8})}','${SQL_ESC(${VAL1})}')

When it is called using:

[2015-08-31 16:35:16] VERBOSE[29562][C-00000001] pbx.c: Executing [103@astsms:37] Set("Message/ast_msg_queue", "ODBC_LOG_SMS(1,ONNET,<;transport=UDP>,;transport=UDP,102,103,Second test 4,)=SUCCESS") in new stack

Asterisk interprets the first "=" as assignment. In the debug log I found:

Variable: ODBC_LOG_SMS(1,ONNET,,;transport=UDP,102,103,Second test 4,)=SUCCESS

And the ODBC function is not executed.

Is there a way, beside using REPLACE,…

Asterisk Users 3 days ago 0 Answers