How to call the mobile phone through asterisk?

Asterisk Asterisk Support
1

I want the sipphone registered on the asterisk to call the mobile phone.
I have tried these 2 methods below,but something wrong.

notice:
I have an account （A）accompanying with the password § and IPPBX server IP address( I), which can call the mobile phone by X-lite

1> I configure the asterisk through asterisk-gui.
1.1, I have set up a trunk ,of which Hostname is I (above) , username is A (above) and password is P (above).

 1.2 , then based on the trunk ,I set rule , dialplan and user. After that, by clicking the System status (gui 1.2) , I find the trunk register successfully , but unfortunately, the user on the asterisk can not call the mobile phone.
    however. through the same way , I can register the local brekeke sip server and can call the user on the brekeke sip server.

2> I directly configure the sip.conf and extensions.conf
2.1, I add a statement “register=>username:password@sip server ip address” in the session ‘general’, and modify the extension in the extencions.conf below, “exten => _XXXX,1,dial(sip/${EXTEN}@sip server ip address)”.however, when I call the mobile phone, it works beyond what I want. it can not call the mobile phone. using the same way to connect the brekeke sip server, it works fine.

now I have no idea to go ahead , and I appreciate someone can give an idea for the following step.

2

hello,happy a new day.

so sad that no one can give me a tip.
I hope that anyone can tell me a little how to do, because such function is an urgent thing.
thank you.

3

There isn’t enough information. You need to provide dialplan fragments and console traces, and may need to provide SIP traces.

Also, the reference to a mobile phone is confusing. Your problem looks like one of simply using an external SIP provider; that is something I haven’t needed to do, but if you need to do it, it would be considered a very trivial use of Asterisk.

If you are using Asterisk 1.2, you are using a version that is no longer officially supported.

4

thanks for your reply

here is my configuration

sip.conf

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
register => 2003:*****@61.235.77.119
nat=yes
pedantic=yes

[serverC]
type=peer
username=2003
secret=*****
host=61.235.77.119

[110]
type=friend
secret=123456
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=outbound

extension.conf

[default]
exten => _XXX,1,playback(auth-thankyou)
exten => _XXX,2,Dial(sip/${EXTEN})

[outbound]
exten => _0XXXXXXXXXXX,1,Playback(auth-thankyou)
exten => _0XXXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@serverC)

and the log is that
<------------->
— (14 headers 0 lines) —
Creating new subscription
Sending to 192.168.3.240 : 5060 (NAT)
Found peer '110’
Looking for 110 in outbound (domain 192.168.3.112)

<— Transmitting (NAT) to 192.168.3.240:5060 —>
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP 192.168.3.240:5060;branch=z9hG4bKa2276dcc9e;received=192.168.3.240;rport=5060
From: sip:110@192.168.3.112;tag=48789c2e
To: sip:110@192.168.3.112;tag=as03499a06
Call-ID: 6c42f27b1f3da0c40705b3e20764be08@192.168.3.240
CSeq: 819 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Creating new subscription
Sending to 192.168.3.240 : 5060 (NAT)
Found peer ‘110’

<— Transmitting (NAT) to 192.168.3.240:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.3.240:5060;branch=z9hG4bKa2276dcc9e;received=192.168.3.240;rport=5060
From: sip:110@192.168.3.112;tag=48789c2e
To: sip:110@192.168.3.112;tag=as35b01caf
Call-ID: 6c42f27b1f3da0c40705b3e20764be08@192.168.3.240
CSeq: 820 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7e5bac84"
Content-Length: 0

finally,my computer is in the LAN.

if anyone know something about solution, please tell me , thank you.

5

today such tough problem I think has been solved.
the main barrier is the Iternet sip server,not my configuration.
my configuration is right.
:laughing: