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Voice Gateway SIP Trunk configuration

George Michaell
Level 1
Level 1

Guys,

 

Need to configure a SIP trunk between Cisco Voice Gateway and Other Solution over the VOIP, so that calls can be recieved on the voice gateway and passed to IP Phone.

 

sip-ua
 registrar ipv4:(IP of Third Part Voip Solution) expires 3600 tcp
 registrar ipv4:(IP of Third Party Voip Solution2) expires 3600 tcp secondary
 sip-server ipv4:(IP of third party voip solution)

Is this the correct way to configure the trunk?

 

secondly do we need to configure POTS dial peer or voip dialpeer will do the job and what else needs to be configured for the successfull SIP trunk registeration and successfull voip call transfer to the IP phone

 

 

 

 

Regards,

AB

1 Accepted Solution

Accepted Solutions

Hi Ab,

 

For any calls from a voip provider ingressing the gateway it has to be a voip dialpeer. Pots dialpeer is configured when you have a TDM trunk like an ISDN PRI terminating on the gateway.

 

Regarding sip server command, as I mentioned earlier this is required only if you want to simplify configuration of pointing dialpeer towards a sip proxy.

 

regards,

Deepu

View solution in original post

4 Replies 4

deepaul
Level 1
Level 1

Hi AB,

 

For configuration of multiple registrars, see the following link.

http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-multi-trunks.html

 

Configuring a SIP server as a session target is useful if a Cisco SIP proxy server (SPS) is present in the network.

http://www.cisco.com/c/en/us/td/docs/ios/voice/command/reference/vr_book/vr_s11.html#wp1133185

In a typical voip call flow, you need to have an inbound voip dial peer on the gateway to accept the call from the sip provider and an outbound dial peer pointing to the cucm to send the call to the ip phone.

 

HTH

regards,

Deepu

Hi Deepu,

there are three IP addresses of the VOIP solution of the customer sending the call through the SIP trunk and the number through which the calls are sent is 5000 and we are translating it to 90040..

____________________________________

Here I am confused  whether it would be a pots dial peer  like below one or a VOIP dialpeer as below...

____________________________________________________
dial-peer voice 1 pots
 incoming called-number 5000
 direct-inward-dial
 forward-digits all

_____________________________________________________

                                      OR

______________________________________________________

dial-peer voice 1 voip
  description *** 10 Digit Calls ***
 incoming called-number 5000
  session protocol sipv2
  session target sip-server

  dtmf-relay rtp-nte
  codec g711ulaw
  no vad

________________________________________________________________

below is the dialpeer through which the call recieved will be sent towards the CVP for IVR treatment

____________________________________________________________________


dial-peer voice 90001 voip

 translation-profile incoming block
 preference 1
 destination-pattern 90040
 session protocol sipv2
 session target ipv4:172.19.34.384
 voice-class codec 1
 dtmf-relay rtp-nte h245-signal h245-alphanumeric
 no vad

 

________________________________________________________________

translating it to 90040

________________________________________________________________

num-exp 5000 90040

 

________________________________________________________________

for the trunk part... here i am confused as well whether registerar IP'S will be the ip addresses of the voip solution Gateway or what and secondly we donot have any sip proxy so do we need to configure ths SIP-Server?

________________________________________________________________
sip-ua
 registrar ipv4:192.168.777.234 expires 3600 tcp
 registrar ipv4:192.168.777.299 expires 3600 tcp secondary
 sip-server ipv4:192.168.777.297
!

________________________________________________________________

 

Any Help would be appreciated.

 

Regards,

AB

 

 

Hi Ab,

 

For any calls from a voip provider ingressing the gateway it has to be a voip dialpeer. Pots dialpeer is configured when you have a TDM trunk like an ISDN PRI terminating on the gateway.

 

Regarding sip server command, as I mentioned earlier this is required only if you want to simplify configuration of pointing dialpeer towards a sip proxy.

 

regards,

Deepu

The Cisco IOS gateway registers all its POTS dial peers to the registrar when the registrar is configured on the Gateway. The introduction of trunk registration support, the registration of a single number would represent the SIP trunk. The SIP trunk registration can then be associated with multiple dial-peers for routing outbound calls. This registration represents all of the gateway end points for routing calls from or to the endpoints. You can also opt for enterprise SIP trunking solutions from Reliance Global Call.