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please give me a example configuration for Convert SIP Call to H.323!thanks!

topology:   cisco router (gk)  <---> cisco router 2911(gw) <---->The third-party sip server <---->The third-party sip ephone

                                            (h323)                                   (sip)

 

question:  The third-party sip cann't communicate cisco ip ephone,I think that between H323 and SIP cannot normal communication.

               please give me a example configuration for  SIP and  H.323  normal communication.

  

  • IP Telephony
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1 ACCEPTED SOLUTION

Accepted Solutions

Hi nannig,

Hi nannig,

 

From what I understand you want to convert h.323 to sip in 2911GW before you send it off to the third-party sip-server and vice versa.

 

In this case the 2911GW will become a CUBE gateway and you will have to apply the below commands for that.

 

voice service voip

allow connections h323 to sip

allow connection sip to h323

allow connection h323 to h323

allow connections sip to sip

no ip address trusted authenticate

 

 

After that you will need to match calls coming from your GK on an incoming dial-peer

Dial-peer voice 100 voip
incoming called-number 9000
dtmf-relay h245-alphanumeric
codec g711alaw
!

Now you will need to send this out to the sip-server with correct codec and dtmf(check with your sip-server)

 

dial-peer voice 200 voip
destination-pattern 9000
session protocol sipv2
session target ipv4:sip-server-ip
dtmf-relay rtp-nte
codec g711alaw

 

 

This is basic setup, but in-case you decide to do transcoding don't forget to register your dsp's.

 

For futher information:

 

Have a great one.

 

1 REPLY

Hi nannig,

Hi nannig,

 

From what I understand you want to convert h.323 to sip in 2911GW before you send it off to the third-party sip-server and vice versa.

 

In this case the 2911GW will become a CUBE gateway and you will have to apply the below commands for that.

 

voice service voip

allow connections h323 to sip

allow connection sip to h323

allow connection h323 to h323

allow connections sip to sip

no ip address trusted authenticate

 

 

After that you will need to match calls coming from your GK on an incoming dial-peer

Dial-peer voice 100 voip
incoming called-number 9000
dtmf-relay h245-alphanumeric
codec g711alaw
!

Now you will need to send this out to the sip-server with correct codec and dtmf(check with your sip-server)

 

dial-peer voice 200 voip
destination-pattern 9000
session protocol sipv2
session target ipv4:sip-server-ip
dtmf-relay rtp-nte
codec g711alaw

 

 

This is basic setup, but in-case you decide to do transcoding don't forget to register your dsp's.

 

For futher information:

 

Have a great one.

 

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