SIP Trunk (Need Advice)

Asterisk Asterisk Support
1

Hello,
I am having a SIP trunk peer from the provider and has defined the following.

/etc/asterisk/sip.conf
[general]
context = incoming-BW
bindport=5060
srvlookup=yes
disallow=all
allow=ulaw

[BW-SIP-A] ; My trunk name
disallow=all
allow=ulaw
type=peer
qualify=300
progressinbound=yes
outboundproxy=IP_Provided by the SIP Trunk Provider
host=IP_Provided by the SIP Trunk Provider
dtmfmode=rfc2833
canreinvite=yes
allow=ulaw
context=incoming-BW

/etc/asterisk/extensions.conf
[incoming-BW]
exten => s,1,Playback(hello-world)
exten => s,n,Hangup

I want to play ‘Hello World’ on incoming call as Testing
My SIP DID is : +12137852497

But I don’t know why I am getting the following on my CLI

[color=#FF0000] Call from ‘BW-SIP-A’ to extension ‘+12137852497’ rejected because extension not found in context ‘incoming-BW’[/color]

I am not sure what is missing in dialplan, Do I need to create the same extensions as my DID number ?
Please advice on this.

2

The most easy way:

[incoming-BW]
exten => +12137852497,1,Goto(s,1)

exten => s,1,Playback(hello-world)
exten => s,n,Hangup

3

ah…
thank you Samael, since this is my first experience of SIP trunk.

4

I did like this
but still have the same problem, If I getting leading + sign how can I remove it. It is actually forwarded from the SIP trunk provider.

5

thanks it’s been solved after after using e64 format.