Can ChanSpy Offer Private Dialog?

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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.

Narrative:

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.

The…

Asterisk Users 49 minutes ago 0 Answers

Call Forwarding In Asterisk

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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

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Hi,

I have a line like

register => 1yyyyyyy1:xxxxxxxxxxxx@sipconnect.sipgate.de/incoming

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

[trunk-sipgate] type=peer qualify=yes insecure=invite language=de dtmfmode=rfc2833 host=sipconnect.sipgate.de fromdomain=sipconnect.sipgate.de 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

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*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

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Hello,

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.

https://supportforums.cisco.com/discussion/12589596/cisco-ube-hangs-calls-immediately-after-being-answered

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) www.OntheNet.com.au

Asterisk Users 3 days ago 0 Answers

AMI 'meetme List Concise' Hanging

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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

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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,< sip:102-DEVEL@devel.mirtapbx.com;transport=UDP>,sip:103@devel.mirtapbx.com;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,,sip:103@devel.mirtapbx.com;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