DTMF Issue in 7936 IP Conference Phone

[Kudur MohanRaj]

Hi,

All 7936 IP conference phone the DTMF service is not working through SIP trunk but at the same time all other IP phones (7941, 7941 etc..) are working DTMF through trunk.

I have configured MTP also. But still not working for me.

My connectivity is CUCM sends the call to H.323 Gateway. Then that router which is configured with sip dial peer communicates with ISP.

I have configured MTP with G.729 codec, as well as Transcoder given both to device pool and H.323 gateway. Call gets disconnected with busy tone.

DN: 32893

phone MAC: 0004F2E0A405

I am also attaching the CUCM trace for the same.

Any help is appreciated. Thank you.

[Jonathan Schulenberg]

For starters, stop introducing unnecessary complexity (and a likely a contributing cause of this problem): convert the H.323 gateway into a SIP trunk so the router doesn't have to do protocol translation.

The 7936 doesn't support in-band RFC2833 DTMF relay which is almost certainly what your SIP carrier wants. This means that CUCM must convert the SCCP out-of-band events into whatever DTMF relay your H.323 gateway is configured for, likely H.245 alphanumeric. The gateway must then convert the H.245 event into an in-band RTP packet toward the carrier. One of the steps in this chain isn't working.

Adding an MTP into the mix won't help here because CUCM can't see a SIP call leg which wants RFC2833, all it can see is the H.323 gateway it passes the call off to. If on the other hand CUCM had a SIP trunk which was demanding RFC2833 (aka RTP payload type 101) in it's SDP offer, then it would invoke an IOS SW MTP for the 7936 to do this conversion for you.

[Kudur MohanRaj]

hi Jonathan Schulenberg,

I have created a sip trunk, means converted h323 gateway connectivity to sip trunk. Now all my other phones even not able to reach. I have taken some logs which may help in course. Please guide me fix this.

[Jonathan Schulenberg]

Please attach either a full router config (sans passwords) or at least the 'voice service voip', 'dial-peer voice', and 'sip-ua' sections. Also call out what IPv4 addresses are what: CUCM nodes, ITSP, CUBE inside/outside interfaces, etc.

[Kudur MohanRaj]

Hi Jonathan Schulenberg,

Thanks for your support, I have attached the required details.

[Ayodeji Okanlawon]

Hi,

I looked at your logs and what I see is your provider doesnt seem to like some of the codecs you are offering to them.

I suggest you disable early offer on CUCM and on the CUBE. This way your provider can send you a list of codecs they support and you can send your answer in the ACK to their 200 OK.

+++++++++++++

Received:
SIP/2.0 415 Unsupported Media Type
Supported: 100rel
Via: SIP/2.0/UDP 115.114.111.194:5060;branch=z9hG4bK44237B
To: <sip:0018668434357@202.54.112.194>;tag=3615533566-618688
From: "IT - Networks and Systems" <sip:32806@202.54.112.194>;tag=62E63018-E79
Remote-Party-Id: "IT - Networks and Systems" <sip:32806@115.114.111.194>;screen=yes;privacy=off
Call-ID: 519A46A0-158511E4-8745CE35-C8A99330@115.114.111.194
CSeq: 101 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: <sip:0018668434357@202.54.112.194:5060>
Call-Info: <sip:202.54.112.194>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Length: 0

[Kudur MohanRaj]

Hi all,

thank you all of you for your precious time to help me with this, i found Ayodeji Okanlawon suggestion helpful. 

I happened to open TAC case, where they came up with isp using g729br8, which was not supported by default in 7936 phone. Later we enabled g729br8 in transcoder which solved the issue. It is still using h323 to sip connectivity.

Regards,

Mohanraj KM