[asterisk-users] asterisk sip trunk configure

[garge rama]

Hi,



I am trying to make external sip calls by using asterisk. Please provide
information regarding sip trunk configuration in conf files.



Setup is as below,

* *

*Case A:*

Register two soft phones [X-lite] with 1000 and 10001 numbers to asterisk
PBX [running in 192.168.1.11] and able to make calls in between.



Sip.conf

======

[general]

context=default

bindport=5060

bindaddr=192.168.1.11

srvlookup=yes



[1000]

type=friend

nat=yes

host=dynamic

canreinvite=no

context=default

allow=ulaw



[1001]

type=friend

nat=yes

host=dynamic

canreinvite=no

context=default

allow=ulaw



extensions.conf

============

[default]

exten => 1000,1,Dial(SIP/1000)

exten => 1001,1,Dial(SIP/1001)



*Case B:*

I have register other phone with ondo sip server running on other PC
[192.168.1.12] with number as 6001.



Now, Want to make calls between this two [asterisk PBX [1000/1001] on
192.168.1.11 and ondo [6001] on 192.168.1.12].

Please suggest how to configure sip trunk in conf files.



Thanks in advance.



Regards,

Garge.
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