Cant transfer an extension to a SIP address with AsteriskNow

Asterisk Asterisk Support
1

I started with a fresh installation of AsteriskNow (no hardware attached, I’m only using SIP), created extension 601 for my softphone, and another extension, 602. When I call 602 from 601, I want it to automatically transfer the call to the following SIP address: "autoattendant@intertex.se".

So I added the following line to the [default] context of extensions.conf:

exten => 602,1,Dial(SIP/autoattendant@intertex.se,60,r)

When I make the call, it seems to connect properly, but I cannot hear anything.

I verified that the SIP address that I’m transferring to works properly.

Two questions:

1) Why doesn't this work?
2) Is there a better way to do this, perhaps via the GUI?

I would really, greatly appreciate any help on this.

Thanks!

2

Which way you are configured? through shell or GUI.Please check the general settings in GUI.There may be predefined key for Transfer a call.Is it happned for only one extension or for all Extensions.Please revert back

kgs_namboothiri@yahoo.co.in MCA,CIC,RHCE,CCNA,CISA

3

Well, I configured the users through the GUI, but I could not find a way to transfer the call through the GUI. So I edited that extensions.conf file for that.

I’m not sure which general settings you are referring to.

This happens for all extensions when I try to add the above Dial() string in the extensions.conf file.

Thanks so much for the help.

4

I don’t have any trunks configured because I am only using SIP. I don’t need to, do I?

5

Can I offer someone a $25 Amazon gift certificate in exchange for some help? I’m really desperate here, and I need to have this working today.

The setup is simple: I took a fresh AsteriskNow install, added two users, registered one of the users to my soft phone, and added the following extensions.conf line under [default] for the other user:

exten => 602,1,Dial(SIP/autoattendant@intertex.se,60,r)

I did nothing else.

I just want to make it so that when the 602 extension is dialed, it transfers the call to the SIP address above. But when the call connects I don’t hear any voice.

Maybe I shouldn’t be touching the extensions.conf file?
I couldn’t find a way to do it in the gui.
Maybe I need a trunk set-up, despite the fact that it’s all SIP?
Maybe the Dial() command isn’t right?

I’ve tried everything I can think of.
Please help! I’m desperate.

6

Ok, after many many hours, I finally figured this out.

The problem is that the AsteriskNow box, and the PC’s I’m using with the soft phones, are all behind a router. When you have your asterisk box behind a router, here’s what you have to do:

  • Go to your rtp.conf file and note the values for rtpstart and rtpend

  • On your router, add port forwarding for the SIP port (5060) and for the entire range of ports between rtpstart and rtpend.

  • Find your EXTERNAL IP address by clicking this link: http://checkip.dyndns.org/

  • Go to your sip.conf file and add the following under the [general] context:

    nat=yes
    externip=123.123.123.123 (replace with your external ip address)
    localnet=10.0.0.0/255.0.0.0 (replace with your local network address and mask)
    canreinvite=no

  • restart asterisk