Route Call in Asterisk Server

0
votes

I have a scenario in which I want to route calls to sip a server from Asterisk but I want to do some thing like this

the sip server is running on : 192.168.1.10 and Asterisk server is running on : 192.168.1.4

client Diales---->sip:[email protected]>To Asterisk and then Asterisk route the calls------->sip:[email protected] --------> sip server

Can someone tell me how can I do this in Asterisk configuration routing file .

2
asterisksipvoip

2 Answers

1
votes

As os11k said, you need SIP trunk to do this. I'll add some detail information about issues we encountered when we setup SIP trunk between two cities.

Add a SIP trunk in sip.conf

  • Make sure you put sendrpid=yes in the SIP trunk configuration, or, 192.168.1.10 will not get the caller id.
  • Optionally, you may want to set context of the SIP trunk to a different context rather than default, say context=sip-server-192-168-1-10.
  • Optionally, you may disallow or allow some codecs if 192.168.1.10 only works with specific codecs.

SIP trunk sample

[general]
register => SIP_ACCOUNT:SIP_PASSWORD@TheOtherSipServer

[TheOtherSipServer]
type=peer
context=sip-server-192-168-1-10

host=192.168.1.10
defaultuser=THE_ACCOUNT_HERE
fromuser=THE_ACCOUNT_HERE
remotesecret=THE_PASSWORD_OF_ACCOUNT_HERE

; if you want to send the remote caller id to 192.168.1.10, then set sendrpid=yes .
; you also need to trust the remote caller id in 192.168.1.10 .
sendrpid=yes

; if 192.168.1.10 can dial out from here, you need to set trustrpid=yes so you can get the caller id
;trustrpid=yes

; if 192.168.1.10 is picky on codecs
;disallow=all
;allow=THE_CODEC_NAME_ALLOWED_BY_THE_OTHER_SIP_SERVER

Setup dial plan in extensions.conf

Dial is all you needed unless you need special requirement.

Dialplan sample

[globals]
SIPTrunk=SIP/TheOtherSipServer

[sip-server-192-168-1-10]
;exten=>111,1,Dial(SIP/TheOtherSipServer/111)
exten=>111,1,Dial(${SIPTrunk}/${EXTEN})