RFC 2833 not accepted?

[vilaro123]

My setup:

AS5350XM V12.4(1c)

My DP voice:

dial-peer voice 4000 voip

service session

destination-pattern [2-9]T

rtp payload-type nte 98

voice-class codec 55

session protocol sipv2

session target ipv4:65.xx.xx.35

dtmf-relay rtp-nte

My problem:

I can't find a way to accept the DTMF from RFC 2833 it always negociate in band.

The carrier wrote:

Here is our invite to your equipment on the DID calls?In read are the Codec Options??We are sending you a 100 telephone event which equates to the RFC 2833 for DTMF.

Header Field INVITE sip:7739021160;npdi=yes;rn=7739021160@206.xx.xx.248:5060 SIP/2.0

........ Via: SIP/2.0/UDP 67.xx.xx.186:5060;branch=z9hG4bK10f942b30b8ba7a8

........ From: <sip:9546838681@67.xx.xx.186:5060>;tag=gK067748a5

........ To: <sip:7739021160@206.xx.xx.248:5060>

........ Call-ID: 335941959_9200@67.xx.xx.186

........ CSeq: 24658 INVITE

........ Max-Forwards: 70

........ Allow: OPTIONS,INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE

........ Accept: multipart/mixed, application/sdp, application/isup, application/dtmf, application/dtmf-relay

........ Contact: <sip:9546838681@67.xx.xx.186:5060>

........ Supported: 100rel

........ Supported: timer

........ Session-Expires: 64800

........ Min-SE: 64800

........ Content-Length: 306

........ Content-Disposition: session; handling=required

........ Content-Type: application/sdp

........ v=0 o=Sonus_UAC 24259 8031 IN IP4 67.xx.xx.186 s=SIP Media Capabilities c=IN IP4 67.xx.xx.139 t=0 0 m=audio 17492 RTP/AVP 18 0 8 100 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-eve

........ /8000

........ a=fmtp:100 0-15

........ a=sendrecv

........ a=maxptime:20

And my Cisco is answering:

Here is your response in the IP messaging 183 Session in Progress to the codec negotiation. Your equipment is not responding w/the correct DTMF Option. Please see highlighted below. This shows the G.729 codec w/in-band DTMF, because there is no 100 telephone event in your response.

Server: Cisco-SIPGateway/IOS-12.x

........ CSeq: 24658 INVITE

........ Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER

........ Allow-Events: telephone-event

........ Contact: <sip:257739021160@65.xxx.xxx.90:5060>

........ Record-Route: <sip:65.xx.xx.33:5060;lr>

........ Content-Disposition: session;handling=required

........ Content-Type: application/sdp

........ Content-Length: 202

........ v=0 o=CiscoSystemsSIP-GW-UserAgent 7471 7899 IN IP4 65.xxx.xxx.90 s=SIP Call c=IN IP4 65.xxx.xxx.90 t=0 0 m=audio 16648 RTP/AVP 18 c=IN IP4 65.xxx.xxx.90 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no

Can someone tell me how to solve this?

Any help will be appreciated.

Thanks in advance

[teodorgeorgiev]

Run a debug and check that the incoming calls from your partner do match (enter and are being handled by) the reffered by you dial-peer 4000.

Might be, the incoming SIP calls do not match this dial-peer and are matched by the default dial-peer settings. The default dial-peer settings say that dtmf-relay is disabled.

This is what I can help you with.

[vilaro123]

Hi Teodor.

Thanks for your answer.

This is my dial-peer 4000:

dial-peer voice 4000 voip

service session

destination-pattern [2-9]T

rtp payload-type nte 98

voice-class codec 55

session protocol sipv2

session target ipv4:65.xxx.xxx.35

dtmf-relay rtp-nte

The voice class codec 55 puts the g729a as the preferred one.

Your answer gave me the idea where to look and found that the calls that doesn't match the dial peer 4000 and go by the default (PeerID= 0) are shown at the show call history voice command as using tx_DtmfRelay=rtp-nte

while the calls that do match the dp 4000 for an unknown reason are shown as using tx_DtmfRelay=inband-voice.

I am looking for a reason but I think it is with the supplier of the DIDs as another supplier using the same dp4000 and also G729a codec looks like using rtp-nte.

If you have any further idea please let me know.

Regards

[teodorgeorgiev]

You have to configure then a dial-peer that matches your incoming calls so they don't go

to the default dial-peer.

Use something like incoming called-number, it depends on your dialing plan.

[vilaro123]

Hi Teodor.

My problem is the other way around.

The calls that match the dial peer 4000 do not use the dtmf relay rtp-nte so they do not accept the RFC 2833. If I edit the FROM header at Sip server level in a way that it do not match the dial peer 4000 and go to the default the call is shown as accepting the RFC2833.

I don't find a cause to this. So because I need the calls using the RFC 2833 I am editing the headers and making them use the default DP. Not the solution I like but is working.

Regards

[teodorgeorgiev]

why don't you do the following: on the 4000 dial-peer do the following changes:

1. add "no vad" command

2. remove "rtp payload-type nte 98"

3. remove "service session"

if still no luck, try to put a hardcoded codec:

codec g711alaw / g711ulaw

it *will* work.