Call Forwarding In Asterisk

Home » Asterisk Users » Call Forwarding In Asterisk
Asterisk Users 4 Comments

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)

4 thoughts on - Call Forwarding In Asterisk

  • You might want to use the Originate() application instead. Check its usage by issuing the command ‘core show application originate’ on Asterisk CLI.

    2015-09-03 9:09 GMT-03:00 Kantharuban Ruban :

  • Hi,
    Thanks for your info, What is the impact of the following line in dialplan,

    Dial(SIP/19201/19202,300)

  • Hello Kantharuban,

    Friday, September 4, 2015, 8:19:28 AM, you wrote:

    It does not look like a valid format. If you are trying to dial two SIP devices (19201 and 19202) with a timeout of 300 seconds, the command would be

    Dial(SIP/19201&SIP/19202,300) and you might want to consider some of the option Dial options depending on what you do with the call after it has been answered.

    Have a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial for details of the dial command, and the options or have a look at Asterisk: The Definitive Guide which will tell you more about Originate and Local Channels, which you might also find useful.

    http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.html

    J

  • Hi ,
    I have gone through the link you have sent me , there i could find the below lines,

    *Dial() together with openining Jack ports for callee*

    *Nescesarry if you want to “capture” a record in leg B with SoundPatty
    <http://github.com/Motiejus/SoundPatty>exten =>
    _X.,n,Dial(SIP/$PROVIDER/${EXTEN},60,M(connect-jack)[macro-connect-jack]exten
    => s,1,NoOp(${CHANNEL}) ; This is leg A, skipexten =>
    s,2,Set(JACK_HOOK(manipulate,i(${CHANNEL}:input),o(${CHANNEL}:output))=on)Note:
    only for asterisk 1.6.x*

    Could you please tell me what does it do?