how to configure sip trunk between cme and issabel

how to configure sip trunk between cme and issabel or how to configure sip trunk between cme and elastix.

Some time we require sip communication between Cisco CME and Asterisk servers to send and receive calls to and from CME to Asterisk .

First we will configure Issabel side with following steps. we will go through following steps .

1- Configure Trunk

2- Configure Outbound Routes

3-Inbound Routes

1- Configure Trunk

1- Login to your web interface of Issabel

Go to PBX > PBX Configuration > Trunks

Add SIP Trunk as following change the setting according to your requirement.

Now on Outgoing setting > PEER Detalis

type=friend ————– to send and receive calls
context=from-internal ——- this is like Partition in Cisco CME .
host=172.16.10.254 ———- Change with CME Router IP Address
disallow=all ——– this will disallow all codec
allow=ulaw&alaw&729 —– this will allow G711a and G711u and G729 Codec
nat=no ——– NAT will not be used
maxexpirey=3600
canreinvite=yes
qualify=no
insecure=port,invite

Now submit the changes and apply the changes

2- Configure Outbound Routes

Go to PBX > PBX Configuration > Outbound Routes > Add Route

Give the Route Name

Add the dial patterns according to your requirement

Choose the Trunk Sequence for Matched Routes as you created in the Trunk configuration

As I have created cmeout

Submit Changes and Apply changes

3-Inbound Routes

Your Default Incomming Route is OK .

or If you are using DID numbers then create inbound routes according to your DID.

Now We will Configure Cisco CME Side

After Basic configuration of CME we will configure the below sections

voice service voip
 ip address trusted list                   -----this to allow sip communication from asterisk 
  ipv4 172.16.10.120                       -----  this is your asterisk server 
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 no supplementary-service h450.2
 no supplementary-service h450.3
 no supplementary-service h450.7
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 sip
  bind control source-interface FastEthernet0/0   ---- this is imp
  bind media source-interface FastEthernet0/0     ---- this is imp
  registrar server




dial-peer voice 5500 voip
 description sip trunk to asterisk out
 destination-pattern 55...                             ---- atserisk ext numbers
 session protocol sipv2
 session target ipv4:172.16.10.120                     ---- asterisk server IP    
 dtmf-relay rtp-nte
 codec g711alaw
!
dial-peer voice 400 voip
 description sip trunk to asterisk incomming
 destination-pattern 4...                             ---- CME Extensions numbers    
 session protocol sipv2
 session target ipv4:172.16.10.120
 incoming called-number 4...
 dtmf-relay rtp-nte
 codec g711alaw

Download CME Full config for reference

CMEFullconfig

Now you can test calls to and from Asterisk to CME and CME to Asterisk.

for debug in Asterisk

asterisk -vvvvvvvvvvvvvr

sip set debug on / sip set debug off

on Cisco CME

debug ccsip all

This is How to configure SIP trunk between Cisco CME and Asterisk

reference taken from

Gkhan Tips