SIP Dial Peer

[sayeedibrahim]

Hi

I have a SIP trunk configured to receive calls. When I configure DID service and send calls directly to user extensions configured on the Call Manager the calls are received successfully. When I configure the DID number to be forwarded to IVR server, I can hear welcome greeting but it does not accept the dtmf digits entered to select options provided by IVR.

I will appreciate suggestions to overcome this problem

!

voice translation-rule 3000

rule 1 /2123000/ /5911/

!

voice translation-rule 3001

rule 1 /2123001/ /1001/

!

!

voice translation-profile TP3000

translate called 3000

!

voice translation-profile TP3001

translate called 3001

!

dial-peer voice 1 voip

description ** << DP for Transfering Calls  to the DN on CCM **

destination-pattern 1...

session target ipv4:10.x.x.254

codec g711alaw

!

dial-peer voice 2 voip

description ** << DP Transferring Calls to ROUTE POINT on CCM

destination-pattern 5911

session target ipv4:10.0.8.254

dtmf-relay cisco-rtp rtp-nte digit-drop h245-signal h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 3000 voip

description ** << Incoming Call >> RoutePoint for IVR **

translation-profile incoming TP3000

incoming called-number 2123000

dial-peer voice 3001 voip

description ** << Incoming Call >> DN for User **

translation-profile incoming TP3001

incoming called-number 2123001

!

Thanks

Ibrahim

[iantra123]

Try this :

dial-peer voice 2 voip

description ** << DP Transferring  Calls to ROUTE POINT on CCM

destination-pattern 5911

session target ipv4:10.0.8.254

codec g711ulaw

no  vad

dtmf-relay rtp-nte

session protocol sipv2

!

New

[GABRIEL KNEBL TUMOLO]

Configure "dtmf-relay rtp-nte" in boths voip dial-peers, incoming and outgoing.

I tested in my lab, CM -> SIP TRUNK(NO PREFERENCE IN DTMF OPTIONS) -> GW -> POTS PSTN. If I remove "dtmf-relay rtp-nte" from voip incoming dial-peer, any IVR do not receives the DTMF. When I add It, every digit is sent to PSTN.

This command fixs your issue!

Gabriel Tumolo

[lavanyaps]

Hi

Just try this change highlighted in bold.

dial-peer voice 2 voip

description

destination-pattern 2978

session protocol sipv2

session target ipv4:10.0.8.254

incoming called-number .T

voice-class codec 1

no vad

voice class codec 1

codec preference 1 g729r8

codec preference 2 g729br8

codec preference 14 g723r53

codec preference 15 g723ar53

codec preference 16 g711alaw

codec preference 17 g723r63

[sayeedibrahim]

I have already tried the suggested configurations but still no luck.

I guess i am gonna dig more into SIP now.

Thanks

Ibrahim

[sayeedibrahim]

Hi

My problem was related to DTMF. Communication and connectivity part was ok as i could place and receive calls with my existing setup

DTMF issue got fixed after forcing rtp-nte under the outgoing and incoming dial peers using the command "voice-class sip dtmf-relay force rtp-nte"

R

[Steven Holl]

Why do you not have any DTMF relay specified on the CM peer?

Try:

dial-peer voice 1 voip

dtmf-relay rtp-nte

If that doesn't fix it, get the following:

debug voip ccapi inout

debug ccsip all

debug voip rtp sess name

Collect as follows:

Router(config)# service sequence
Router(config)# service timestamps debug datetime msec
Router(config)# logging buffered 10000000 7
Router(config)# no logging con
Router(config)# no logging mon
Router(config)# voice iec syslog

Router# term len 0

Router# sh logg

Place a test call, and hit some DTMF.

[dconstantino]

If your SIP trunk is to and from a SP I assuming you have added the following from your config which is H323 to SIP.

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
supplementary-service h450.12

If you are just SIP

voice service voip
sip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0

[Steven Holl]

dconstantino wrote:
If your SIP trunk is to and from a SP I assuming you have added the following from your config which is H323 to SIP.
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 supplementary-service h450.12
If you are just SIP
voice service voip
 sip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0

Those changes aren't going to effect DTMF.  I don't recommend you bind SIP on a SIP-to-SIP CUBE, since it removes the listener from the external interface.  90% of installs shouldn't need a SIP bind on SIP to SIP since we'll source from the interfaces closest to the destination of the SIP packet.

To the original poster:

I didn't even realize inititally that this was H323-SIP CUBE.

Your original config doesn't have a SIP peer, and several of your H323 peers don't have H245-alpha configured on it for DTMF.

You need to read this document first:

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml

That's important, because DTMF caps are going to depend on the inbound/outbound dial-peers which are matched.  You need to ensure that on the inbound dial-peer match for the call that you match the dial-peer for that protocol, and for the outbound dial-peer match, that you match the dial-peer for that protocol.  Use 'debug voip ccapi inout' to verify, or the pids in 'sh call active voice br' once the call is up.

To keep things simple, just have 'dtmf-relay h245-alpha' on your H323 peers, and 'dtmf-relay rtp-nte' on your SIP peers.  Then make your 'incoming called-number' statements on at least one SIP and one H323 peer specific enough that you match the right one on the *inbound* dial-peer match for a call in each direction.

If all else, fails, post a CCAPI debug output with a recent config and we'll straighten ya out.

[Tommer Catlin]

Have you tried dtmf-relay rtp-nte sip-notify on your dial peer?

[Steven Holl]

tcatlinins wrote:
Have you tried dtmf-relay rtp-nte sip-notify on your dial peer?

That isn't going to correct the misconfiugration that the OP has with no dtmf-relay configured on the h323 peers.

[SaibhuwanSudina70631]

Hi Sayeeibrahim,

I want to receive the call from few grandstream phone, which are registered on a remote IP-PBX, through a SIP Trunk.

The grand stream phones are able to call through a customer care application, while i am unable to place a call through Cisoc ip phones, which are registered on CUCM.

so as i am unable to process the call through SIP Trunk-route group-route list-route pattern, the vendor of IP-PBX asked me to configure peer to process the calling from cucm to ip-pbx.

note : remote ip-pbx was integrated through SIP trunk.

note : given details to configure peer. are "peer no" , "password" , "IP" , "SIP port"

please suggest how to configure that.

Thanks & Regards

Sai Bhuwan

[SaibhuwanSudina70631]

I want to receive the call from few grandstream phone, which are registered on a remote IP-PBX, through a SIP Trunk.

The grand stream phones are able to call through a customer care application, while i am unable to place a call through Cisoc ip phones, which are registered on CUCM.

so as i am unable to process the call through SIP Trunk-route group-route list-route pattern, the vendor of IP-PBX asked me to configure peer to process the calling from cucm to ip-pbx.

note : remote ip-pbx was integrated through SIP trunk.

note : given details to configure peer. are "peer no" , "password" , "IP" , "SIP port"

please suggest how to configure that.

Thanks & Regards

Sai Bhuwan