Carrier Based Routing Enhancements with SIP

[s_malinskiy]

Hello,

I use carrier based routing for separate SIP providers, but i have fax issue.  How force ISR to use dtmf-relay sip-notify rtp-nte for one voice source-group and sip notify method for other one voice source-group?

Thank you.

[yuanwu]

Hello,

From your description, you mentioned voice source-group, fax, and dtmf, if understood correctly, for the different dtmf method for the different voice source-group, if you have the significance of called party number or calling party number, you can use different incoming dial-peer matching for the different dtmf method or use the translation-profile to change calling or called party number for the significance and changed it back at the outgoing dial-peer again.

BTW, If you can give an detail example,it will be easy to find the solution for it.

Rgds/Randy

[s_malinskiy]

Hello,

Thank you for you advice!

trunk group  westcall_pstn
carrier-id west_call_pstn
max-calls any 60
!
!
trunk group  westcall_pstn_conf
carrier-id west_call_pstn_conf
max-calls any 60

!
voice source-group west_call_pstn
carrier-id target west_call_pstn
translation-profile incoming Westcall_in
incoming called-number 7495748861[0,9]
!
voice source-group west_call_pstn_conf
carrier-id target west_call_pstn_conf
translation-profile incoming Westcall_in
incoming called-number 74957488616
!

ial-peer voice 1001 voip
description SIP test account - incoming
modem passthrough nse codec g711alaw
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
incoming called-number 7495748861.$
dtmf-relay sip-notify rtp-nte
  fax protocol t38 nse ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw

dial-peer voice 1002 voip
description SIP test account - incoming
modem passthrough nse codec g711alaw
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
incoming called-number 7495748861[9,0]$
dtmf-relay sip-notify rtp-nte
  fax protocol t38 nse ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
!

!
dial-peer voice 7 pots
trunkgroup westcall_pstn_conf
huntstop
carrier-id target west_call_pstn_conf
forward-digits all
no sip-register
!
dial-peer voice 8 pots
trunkgroup westcall_pstn
huntstop
carrier-id target west_call_pstn
forward-digits all
no sip-register
!

7495748861[0,9] - successfully matched with 1002  dp

74957488616 - successfully matched with 1001 dp

How we can match incoming dial-peers on terminating gateway based with carrier_id?

Thank you.

[Muthurani Lavanya Paneerselvam]

Hi

Find the answer for your query,

How we can match incoming dial-peers on terminating gateway based with carrier_id?

Two parameters are used to terminate the call in the gateway based on carried-id. One is Carrier ID Source and Trunk group lable target.

Carrier ID Source : Will be the name of the label defined in the Incoming call leg.

Trunk group label Target : Will be the name of the label defined in the Dialpeer POTS to match the call coming with the Carried ID Source and Terminate the call in the mentioned Diapeer.

# show run
voice source-group Incoming-call-term
carrier-id source Inc-west-call
trunk-group-label target Inc-west-call-pstn conf

dial-peer voice 1 voip
description Inc-call-routing
voice-class codec 1
incoming called-number .T
session target ipv4:192.168.5.10

!
dial-peer voice 2 pots
trunkgroup westcall-pstn-conf
tone ringback alert-no-PI
translation-profile outgoing typenational
trunk-group-label target Inc-west-call-pstn conf
no digit-strip
direct-inward-dial
forward-digits all

Incoming call terminated in the gateway with the carrier id source "Inc-west-call".  On the voice source-group we define the trunk-group-lable target as "Inc-west-call-pstn conf". This lable is then matched in the dial-peer voice 2 pots and then the call will be routed to corresponding trunkgroup(dialpeer 2 pots)  configured with the E1/T1 interface.

Thanks. Pl rate if it is helpful.