Non-codec capabilities (dtmf) (telephone-events)

Asterisk Asterisk Support
1

Hi:
Im a problem with a SIP message "183 Session Progress", and i cant hear the called party ringing or callback ringing.

I was looking asterisk logs, and think that my provider sendme a RTP stream in “telephone-events” DTMF codification, but my asterisk does not recognize this.

Thanks for all!!! :smiley:

This is the log:

<--- SIP read from UDP:190.xxx.xxx.xxx:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK76e580c1;received=201.xxx.xxx.xxx
From: "Phone1" <sip:234243234@201.xxx.xxx.xxx>;tag=as68297917
To: <sip:wefwefwefwe@190.xxx.xxx.xxx>;tag=as67776a09
Call-ID: 7291dda57ce1458821ff8f032566e596@200.69.11.244:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.7.0(1.4.42)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:234243234@190.xxx.xxx.xxx>
Content-Type: application/sdp
Content-Length: 301
v=0
o=root 1862 1862 IN IP4 46.37.1.36
s=session
c=IN IP4 190.xxx.xxx.xxx
t=0 0
m=audio 10596 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: --- (12 headers 15 lines) ---
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7291dda57ce145882$
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing session-level SDP o=root 1862 1862 IN IP4 190.xxx.xxx.xxx... UNSUPPORTED.
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing session-level SDP s=session... UNSUPPORTED.
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing session-level SDP c=IN IP4 190.xxx.xxx.xxx... OK.
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: Found RTP audio format 0
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Setting payload 0 based on m type on 0xb543c024
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: Found RTP audio format 8
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Setting payload 8 based on m type on 0xb543c024
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: Found RTP audio format 3
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Setting payload 3 based on m type on 0xb543c024
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: Found RTP audio format 18
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Setting payload 18 based on m type on 0xb543c024
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Setting payload 18 based on m type on 0xb543c024
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: Found RTP audio format 101
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Setting payload 101 based on m type on 0xb543c024
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: Found audio description format PCMU for ID 0
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: Found audio description format PCMA for ID 8
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: Found audio description format GSM for ID 3
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: Found audio description format G729 for ID 18
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: Found audio description format telephone-event for ID 101
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED.
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Incorporating payload 0 on 0xb543c024
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Incorporating payload 3 on 0xb543c024
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Incorporating payload 8 on 0xb543c024
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Incorporating payload 18 on 0xb543c024
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Incorporating payload 101 on 0xb543c024
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10e (gsm|ulaw|alaw$
[Feb 11 09:21:35][b] VERBOSE[7959] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon$[/b]
[Feb 11 09:21:35] DEBUG[7959] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9781ee0'
[Feb 11 09:21:35] VERBOSE[7959] chan_sip.c: Peer audio RTP is at port 190.xxx.xxx.xxx:10596
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Copying payload 0 from 0xb543c024 to 0x978208c
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Copying payload 3 from 0xb543c024 to 0x978208c
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Copying payload 8 from 0xb543c024 to 0x978208c
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Copying payload 18 from 0xb543c024 to 0x978208c
[Feb 11 09:21:35] DEBUG[7959] rtp_engine.c: Copying payload 101 from 0xb543c024 to 0x978208c
[Feb 11 09:21:35] DEBUG[7959] chan_sip.c: We're settling with these formats: 0x10e (gsm|ulaw|alaw|g729)
[Feb 11 09:21:35] VERBOSE[18032] app_dial.c:     -- SIP/EasyPhonia-00000021 is making progress passing it to SIP/CABTEST1-00$
[Feb 11 09:21:35] DEBUG[18032] rtp_engine.c: Setting early bridge SDP of 'SIP/CABTEST1-00000020' with that of 'SIP/EasyPhoni$
[Feb 11 09:21:35] DEBUG[18032] chan_sip.c: Setting framing from config on incoming call
[Feb 11 09:21:35] DEBUG[18032] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True
[Feb 11 09:21:35] DEBUG[18032] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Feb 11 09:21:35] DEBUG[18032] chan_sip.c: -- Done with adding codecs to SDP
...
2

You have had a successful audio and RFC 2833 negotiation.

Your problem is that, since about 1.6.2, Asterisk requires that Progress() be explicity called before early media is passed. This is presumably because sending early media to the PSTN is likely to start charging for an outgoing call, or be ignored.

3

David, Thanks four you response.
I add the Progress application in the dial plan, as given below

exten => 336, 1, Progress()
exten => 336, n, Dial(SIP/Peerxxxx/511xxxxxxxx)

However, the fail is following… :frowning: