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[nachum.yaron at gmail.com]

Hello,I have installed the latest version 12 that has been released (12.1.0.rc3).


I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working.


Is it a known issue?


Below you can see the output of the asterisk monitor. 




<--- Received SIP request (1182 bytes) from UDP:10.25.153.150:5060 --->
INVITE sip:039988120@172.16.60.160:5060;user=phone SIP/2.0
Record-Route: <sip:10.25.153.150;lr;ftag=02e3a8c0-33807b-t-2>
Via: SIP/2.0/UDP 10.25.153.150:5060;branch=z9hG4bK587.67258295.0
Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3*
Max-Forwards: 68
From: "39937841 39937841" <sip:39937841;cpc=payphone@192.168.225.2:5060;user=phone>;tag=02e3a8c0-33807b-t-2
To: <sip:D39539988120@192.168.225.2:5060;user=phone>
Call-ID: 2915b6e4-02e3a8c0-0000be53@192.168.225.2 (2915b6e4-02e3a8c0-0000be53@192.168.225.2)
CSeq: 2 INVITE
Contact: <sip:10.1.1.10;line=sr-N6IAzBMsz.MwzxPfPxFsMJZfWBc7MBVuOBV-W.y6MxV*>
User-Agent: NetCentrex CCS Softswitch/7.16.0
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, INFO, PRACK, UPDATE, NOTIFY
Supported: 100rel
P-Asserted-Identity: "39937841 39937841" <sip:39937841;cpc=payphone@192.168.225.2:5060;user=phone>
Min-SE: 90
Privacy: none
Content-Type: application/sdp
Content-Length: 167


v=0
o=10.206.22.171 62708 2 IN IP4 10.206.22.171
s=SIP Call
c=IN IP4 10.206.22.171
t=0 0
a=sendrecv
m=audio 41040 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1
a=ptime:20


<--- Transmitting SIP response (602 bytes) to UDP:10.25.153.150:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.25.153.150:5060;rport;received=10.25.153.150;branch=z9hG4bK587.67258295.0
Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3*
Record-Route: <sip:10.25.153.150;lr;ftag=02e3a8c0-33807b-t-2>
Call-ID: 2915b6e4-02e3a8c0-0000be53@192.168.225.2 (2915b6e4-02e3a8c0-0000be53@192.168.225.2)
From: "39937841 39937841" <sip:39937841;cpc=payphone@192.168.225.2 (payphone@192.168.225.2);user=phone>;tag=02e3a8c0-33807b-t-2
To: <sip:D39539988120@192.168.225.2 ([email]sip%3AD39539988120@192.168.225.2[/email]);user=phone>
CSeq: 2 INVITE
Content-Length:  0




    -- Executing [039988120@from-external:1] NoOp("PJSIP/sipp-00000000", " H E L L O ! ! !") in new stack
    -- Executing [039988120@from-external:2] DumpChan("PJSIP/sipp-00000000", "") in new stack


Dumping Info For Channel: PJSIP/sipp-00000000:
================================================================================
Info:
Name=               PJSIP/sipp-00000000
Type=               PJSIP
UniqueID=           172.16.60.160-1394542052.0
LinkedID=           172.16.60.160-1394542052.0
CallerIDNum=        39937841;cpc=payphone
CallerIDName=       39937841 39937841
ConnectedLineIDNum= (N/A)
ConnectedLineIDName=(N/A)
DNIDDigits=         (N/A)
RDNIS=              (N/A)
Parkinglot=
Language=           en
State=              Ring (4)
Rings=              1
NativeFormat=       (alaw)
WriteFormat=        alaw
ReadFormat=         alaw
RawWriteFormat=     alaw
RawReadFormat=      alaw
WriteTranscode=     No
ReadTranscode=      No
1stFileDescriptor=  -1
Framesin=           0
Framesout=          0
TimetoHangup=       0
ElapsedTime=        0h0m0s
BridgeID=           (Not bridged)
Context=            from-external
Extension=          039988120
Priority=           2
CallGroup=
PickupGroup=
Application=        DumpChan
Data=               (Empty)
Blocking_in=        (Not Blocking)


Variables:
================================================================================
    -- Executing [039988120@from-external:3] Answer("PJSIP/sipp-00000000", "") in new stack
<--- Transmitting SIP response (1060 bytes) to UDP:10.25.153.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.25.153.150:5060;rport;received=10.25.153.150;branch=z9hG4bK587.67258295.0
Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3*
Record-Route: <sip:10.25.153.150;lr;ftag=02e3a8c0-33807b-t-2>
Call-ID: 2915b6e4-02e3a8c0-0000be53@192.168.225.2 (2915b6e4-02e3a8c0-0000be53@192.168.225.2)
From: "39937841 39937841" <sip:39937841;cpc=payphone@192.168.225.2 (payphone@192.168.225.2);user=phone>;tag=02e3a8c0-33807b-t-2
To: <sip:D39539988120@192.168.225.2 ([email]sip%3AD39539988120@192.168.225.2[/email]);user=phone>;tag=b23cda89-931c-4a95-85c5-0ec8b03f895c
CSeq: 2 INVITE
Contact: <sip:172.16.60.160:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   193


v=0
o=- 62708 4 IN IP4 172.16.60.160
s=Asterisk
c=IN IP4 172.16.60.160
t=0 0
m=audio 13644 RTP/AVP 8
c=IN IP4 172.16.60.160
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:150
a=sendrecv


<--- Received SIP request (703 bytes) from UDP:10.25.153.150:5060 --->
ACK sip:172.16.60.160:5060 SIP/2.0
Via: SIP/2.0/UDP 10.25.153.150:5060;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuq3w3X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJWBeIME3ugSVwWx3A3BPAMxIqg.jZWxqL3BqwMRjsW.j*
Max-Forwards: 67
From: "39937841 39937841" <sip:39937841;cpc=payphone@192.168.225.2:5060;user=phone>;tag=02e3a8c0-33807b-t-2
To: <sip:D39539988120@192.168.225.2:5060;user=phone>;tag=b23cda89-931c-4a95-85c5-0ec8b03f895c
Call-ID: 2915b6e4-02e3a8c0-0000be53@192.168.225.2 (2915b6e4-02e3a8c0-0000be53@192.168.225.2)
CSeq: 2 ACK
User-Agent: NetCentrex CCS Softswitch/7.16.0
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, INFO, PRACK, UPDATE, NOTIFY
Content-Length: 0




       > 0x99694c0 -- Probation passed - setting RTP source address to 10.206.22.171:41040
    -- Executing [039988120@from-external:4] Read("PJSIP/sipp-00000000", "dataEntry,"why-no-answer-mystery",10,,1,4") in new stack
    -- Accepting a maximum of 10 digits.
    -- <PJSIP/sipp-00000000> Playing 'why-no-answer-mystery.alaw' (language 'en')

[mjordan at digium.com]

On Tue, Mar 11, 2014 at 8:23 AM, Yaron Nachum <nachum.yaron@gmail.com> wrote:

[nachum.yaron at gmail.com]

Hi Mathew,The regular sip stack has 'auto' dtmfmode which behaved as I said - if the remote replied with telephony event it used RFC2833 otherwise it used inband.







On Tue, Mar 11, 2014 at 5:43 PM, Matthew Jordan <mjordan@digium.com (mjordan@digium.com)> wrote:

[mjordan at digium.com]

On Tue, Mar 11, 2014 at 11:23 AM, Yaron Nachum <nachum.yaron@gmail.com> wrote:

[nachum.yaron at gmail.com]

Mathew, Thanks Mathew. It's good to know the limitations
 
Is there any plan to add it?



On Tue, Mar 11, 2014 at 6:38 PM, Matthew Jordan <mjordan@digium.com (mjordan@digium.com)> wrote: