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Michael Konietzny
2006-Oct-08 02:02 UTC
[asterisk-users] Transfer app and DTMF via SIP info

Hello asterisk-users,

I'm currently investigating a problem related to the Transfer app and
DTMF tones via SipInfo.
My setup depends on:

Asterisk  1.2.10
Zaptel 1.2.8
libpri 1.2.3
Elmeg IP 290 (snom190)
Wildcard TE400 (E1)

The following dialplan is given:

exten => 555, 1, Transfer(554);

exten => 554, 1,Dial (SIP/tel3, 10, tT);
exten => 554, 2,Dial (Zap/g1/017123123123, 10, tT);
exten => 554, 3,Hangup();

If I dial 555 on my SIP phone it transfers to 554 and connecting me to
that zap channel.
Arriving there I'm not able to type ANY DTMF tones.

If the Transfer is skipped the DTMF tones are available. I've included
the SIP debugs to help you track the problem.

Greetings and many thanks in advance,

Michael Konietzny

-------------- next part --------------
    -- Executing Transfer("SIP/tel2-b721ef28", "554") in new
stack
Reliably Transmitting (no NAT) to 192.168.97.21:2054:
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP
192.168.97.21:2054;branch=z9hG4bK-gu4c0f6c0cim;rport;received=192.168.97.21
From: "tel2" <sip:tel2@192.168.97.11>;tag=r7pzlq4bdy
To: <sip:555@192.168.97.11;user=phone>;tag=as21b6ba81
Call-ID: 3c26743ab71b-6y8or3m5c9u7@snom190
CSeq: 2 INVITE
User-Agent: MMS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: Transfer <sip:554@192.168.97.11>
Content-Length: 0

...

    -- Called tel3
    -- SIP/tel3-082c99c8 is ringing
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
192.168.97.21:2054;branch=z9hG4bK-zqrwcwsdcly4;rport;received=192.168.97.21
From: "tel2" <sip:tel2@192.168.97.11>;tag=lt4rnm3do0
To: sip:554@192.168.97.11;tag=as20294491
Call-ID: 3c26743ae09c-uwtyxtc15w1w@snom190
CSeq: 2 INVITE
User-Agent: MMS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:554@192.168.97.11>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 31556 31556 IN IP4 192.168.97.11
s=session
c=IN IP4 192.168.97.11
t=0 0
m=audio 18426 RTP/AVP 8 3 0
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 telephone-event/8000
a=fmtp:0 0-16
a=silenceSupp:off - - - -

Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-zqrwcwsdcly4;rport
From: "tel2" <sip:tel2@192.168.97.11>;tag=lt4rnm3do0
To: sip:554@192.168.97.11
Call-ID: 3c26743ae09c-uwtyxtc15w1w@snom190
CSeq: 2 CANCEL
Max-Forwards: 70
Contact: <sip:tel2@192.168.97.21:2054;line=pisnle1m>
Content-Length: 0

... 

    -- Called g1/017123123123
We're at 192.168.97.11 port 18426
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 192.168.97.21:2054:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
192.168.97.21:2054;branch=z9hG4bK-zqrwcwsdcly4;rport;received=192.168.97.21
From: "tel2" <sip:tel2@192.168.97.11>;tag=lt4rnm3do0
To: sip:554@192.168.97.11;tag=as20294491
Call-ID: 3c26743ae09c-uwtyxtc15w1w@snom190
CSeq: 2 INVITE
User-Agent: MMS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:554@192.168.97.11>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 31556 31556 IN IP4 192.168.97.11
s=session
c=IN IP4 192.168.97.11
t=0 0
m=audio 18426 RTP/AVP 8 3 0
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 telephone-event/8000
a=fmtp:0 0-16
a=silenceSupp:off - - - -

....

    -- Hungup 'Zap/1-1'
-------------- next part --------------
INVITE sip:554@192.168.97.11;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-7zpvewacvy6j;rport
From: "tel2" <sip:tel2@192.168.97.11>;tag=dtndk3lw7m
To: <sip:554@192.168.97.11;user=phone>
Call-ID: 3c267453e7ef-ou4n1yu4s21k@snom190
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:tel2@192.168.97.21:2054;line=pisnle1m>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60x
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1728010931 1728010931 IN IP4 192.168.97.21
s=call
c=IN IP4 192.168.97.21
t=0 0
m=audio 62868 RTP/AVP 8 0 3 9 18 4
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:3 gsm/8000
a=rtpmap:9 g722/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=ptime:20
a=sendrecv

... 

INVITE sip:554@192.168.97.11;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-qgjwhifwkg39;rport
From: "tel2" <sip:tel2@192.168.97.11>;tag=dtndk3lw7m
To: <sip:554@192.168.97.11;user=phone>
Call-ID: 3c267453e7ef-ou4n1yu4s21k@snom190
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:tel2@192.168.97.21:2054;line=pisnle1m>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60x
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Proxy-Authorization: Digest
username="tel2",realm="asterisk",nonce="61364bf6",uri="sip:554@192.168.97.11;user=phone",response="5140f1d5f042256b8daf901b18c603af",algorithm=md5
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1728010931 1728010931 IN IP4 192.168.97.21
s=call
c=IN IP4 192.168.97.21
t=0 0
m=audio 62868 RTP/AVP 8 0 3 9 18 4
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:3 gsm/8000
a=rtpmap:9 g722/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=ptime:20
a=sendrecv

    -- Called tel3
wum97011*CLI>
<-- SIP read from 192.168.97.21:2054:
SUBSCRIBE sip:554@192.168.97.11;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-xpi4mpqz7oo7;rport
From: <sip:tel2@192.168.97.11>;tag=0vtl8gobz9
To: <sip:554@192.168.97.11;user=phone>
Call-ID: 3c267453ea60-6rhkgam1ezu2@snom190
CSeq: 2 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:tel2@192.168.97.21:2054;line=pisnle1m>
Event: dialog;purpose=call-completion
Accept: application/dialog-info+xml
Authorization: Digest
username="tel2",realm="asterisk",nonce="24fec071",uri="sip:554@192.168.97.11;user=phone",response="00113b1d54ec1b40f61e2ebf054d1bc4",algorithm=md5
Expires: 3600
Content-Length: 0


    -- Called g1/017123123123
We're at 192.168.97.11 port 16630
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Transmitting (no NAT) to 192.168.97.21:2054:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
192.168.97.21:2054;branch=z9hG4bK-qgjwhifwkg39;rport;received=192.168.97.21
From: "tel2" <sip:tel2@192.168.97.11>;tag=dtndk3lw7m
To: <sip:554@192.168.97.11;user=phone>;tag=as75e14252
Call-ID: 3c267453e7ef-ou4n1yu4s21k@snom190
CSeq: 2 INVITE
User-Agent: MMS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:554@192.168.97.11>
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 31556 31556 IN IP4 192.168.97.11
s=session
c=IN IP4 192.168.97.11
t=0 0
m=audio 16630 RTP/AVP 8 0 3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
asterisk users - Oct 2006 - Transfer app and DTMF via SIP info

[asterisk-users] Transfer app and DTMF via SIP info
