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SIP Phone 6921 doesn't send DTMF to BiB recorder

Hi friends!

Our  company is developing a call recorder based on built-in-bridge  technology. This recorder uses the DTMF (RFC-2833) signals from the  telephones to activate some custom processing. We did all development using phone 7911 and CUCM 8.6 as a test base, getting awesome results.

However, we started testing other phones with BiB technology and got stuck at Phone 6921, because we couldn't find a way to make it send DTMF events ( both received and transmitted ) to the recorder.

When sniffing the network, after pressing some button on the keypad, we see this phone sending a SIP NOTIFY with DTMF data to the Call Manager, a RFC-2833 event to the far-end's RTP stream, but no RFC-2833 event to the recorder's RTP stream.

Is there some configuration we are missing out?? Is this a known bug or limitation??

Thanks in advance

2 Replies 2

Karthik Sivaram
Level 4
Level 4

Hi Sergio,

Please check this bug CSCua47105.

Hope this helps!

Thanks,

Karthik

Hey Karthik!

Phone is working with payload 101 already. We can see it sending RTPs with payload 101 to its far-end side. Only at the recorder call legs (far/near ends) they are not happening.

Here is the sequence of SIP messages within SDP negotiation, from the phone perspective, for the recorder's

near-end leg:

-> INVITE

v=0

o=CiscoSystemsCCM-SIP 4275 1 IN IP4 192.168.2.232

s=SIP Call

c=IN IP4 192.168.2.232

t=0 0

m=audio 4000 RTP/AVP 0

a=label:X-relay-nearend

a=rtpmap:0 PCMU/8000

a=ptime:20

a=inactive

<- 200 OK

v=0

o=Cisco-SIPUA 19606 0 IN IP4 192.168.2.75

s=SIP Call

t=0 0

m=audio 19134 RTP/AVP 0 101

c=IN IP4 192.168.2.75

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=inactive

-> INVITE (no SDP)

<- 200 OK

v=0

o=Cisco-SIPUA 19606 1 IN IP4 192.168.2.75

s=SIP Call

t=0 0

m=audio 19134 RTP/AVP 0 101

c=IN IP4 192.168.2.75

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

-> ACK

v=0

o=CiscoSystemsCCM-SIP 4275 2 IN IP4 192.168.2.232

s=SIP Call

c=IN IP4 192.168.2.171

t=0 0

m=audio 40042 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=recvonly

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15