[asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833
Yaron Nachum
nachum.yaron at gmail.com
Tue Mar 11 08:23:04 CDT 2014
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 at 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 at 192.168.225.2:5060
;user=phone>;tag=02e3a8c0-33807b-t-2
To: <sip:D39539988120 at 192.168.225.2:5060;user=phone>
Call-ID: 2915b6e4-02e3a8c0-0000be53 at 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 at 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 at 192.168.225.2
From: "39937841 39937841" <sip:39937841;cpc=payphone at 192.168.225.2
;user=phone>;tag=02e3a8c0-33807b-t-2
To: <sip:D39539988120 at 192.168.225.2;user=phone>
CSeq: 2 INVITE
Content-Length: 0
-- Executing [039988120 at from-external:1] NoOp("PJSIP/sipp-00000000", "
H E L L O ! ! !") in new stack
-- Executing [039988120 at 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 at 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 at 192.168.225.2
From: "39937841 39937841" <sip:39937841;cpc=payphone at 192.168.225.2
;user=phone>;tag=02e3a8c0-33807b-t-2
To: <sip:D39539988120 at 192.168.225.2
;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 at 192.168.225.2:5060
;user=phone>;tag=02e3a8c0-33807b-t-2
To: <sip:D39539988120 at 192.168.225.2:5060
;user=phone>;tag=b23cda89-931c-4a95-85c5-0ec8b03f895c
Call-ID: 2915b6e4-02e3a8c0-0000be53 at 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 at 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')
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