[Sending Out-of-Band DTMF to an IKON Gateway]

I've found a way to make the gateway behave correctly - it works now! Many thanks for your help.

[Sending Out-of-Band DTMF to an IKON Gateway]

Thanks for the reply. I can find no relevant settings on the gateway. I noticed that a=fmtp:101 0-16 is not mentioned in the gateway's reply. However, RFC 2833 says:

Since all implementations MUST be able to receive events 0 through   15, listing these events in the a=fmtp line is OPTIONAL.

Are you saying the snomone pbx insists the gateway response explicitly includes a=fmtp... ?

[Sending Out-of-Band DTMF to an IKON Gateway]

I need to send Out-of-Band DTMF to an extension, but the snomone appears to transcode the DTMF to in-band. (The "extension" is really an IKON gateway which only supports out-of-band DTMF.) Here's a part of the logfile:

 

[6] 2013/09/10 13:19:40: Call port 121: Different Codecs (local telephone-event/8000, remote PCMA/8000), callid f419f99b@pbx, falling back to transcoding

[6] 2013/09/10 13:19:40: Received DTMF *, call type attendant

[8] 2013/09/10 13:19:40: Attendant: Ignoring the DTMF * in the state connected

 

How can I make the pbx forward the DTMF as an RTP-event to the "extension"?

In case it's relevant, here's the INVITE etc.

 

2013/9/10 13:19:39 Tx: udp:172.27.66.105:5067 (959 bytes)

INVITE sip:240@172.27.66.105:5067 SIP/2.0

Via: SIP/2.0/UDP 172.27.66.106:5060;branch=z9hG4bK-abb91ea6476deec2bab243dd796c1b78;rport

From: "Svalbard MER1" <sip:401@localhost;user=phone>;tag=51546

To: "MSG Control Room" <sip:240@localhost>

Call-ID: 57a2ef9c@pbx

CSeq: 27025 INVITE

Max-Forwards: 70

Contact: <sip:240@172.27.66.106:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/5.1.0

Alert-Info: <http://127.0.0.1/Bellcore-dr3>

Content-Type: application/sdp

Content-Length: 329

 

v=0

o=- 63349 63349 IN IP4 172.27.66.106

s=-

c=IN IP4 172.27.66.106

t=0 0

m=audio 41036 RTP/AVP 8 0 9 2 3 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:9 G722/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

and here's the OK back

 

2013/9/10 13:19:39 Rx: udp:172.27.66.105:5067 (820 bytes)

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.27.66.106:5060;branch=z9hG4bK-abb91ea6476deec2bab243dd796c1b78;rport

From: "Svalbard MER1" <sip:401@localhost;user=phone>;tag=51546

To: "MSG Control Room" <sip:240@localhost>;tag=802229

Call-ID: 57a2ef9c@pbx

CSeq: 27025 INVITE

Contact: <sip:240@172.27.66.105:5067>

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

Content-Length: 149

Content-Type: application/sdp

Server: (innovaphone H323/[unknown])

Supported: replaces,100rel,sec-agree,answermode

P-Preferred-Identity: <sip:240@172.27.66.105:5067>

Remote-Party-ID: <sip:240@172.27.66.105:5067>;party=called;screen=no;privacy=off

 

v=0

o=- 49 1 IN IP4 172.27.66.105

s=-

c=IN IP4 172.27.66.105

t=0 0

a=sendrecv

m=audio 8184 RTP/AVP 8 0

a=ptime:20

a=silenceSupp:off - - - -

[Callback feature]

Thanks! It works.

[Callback feature]

I'm using snom one version 5.1.0 and need to disable the "camp on" feature. (some of my UAs have no dial pads) This option does not appear in the Domain Settings. Is it now located somewhere else or called something else?