[asterisk-dev] 1.6.2.0-beta4 - SIP TCP or TLS - Ringing/OK ignored

[Stefan Tichy]

Asterisk 1.6.2.0-beta4 has udp and tcp enabled for SIP calls.
If the phone snom360-SIP 7.3.7 uses udp everything seems to work,
but if I change this to tcp incoming calls do fail. No problem with
outgoing calls or registration.

Asterisk does send INVITE, ignores 180 Ringing and 200 OK but
observes BYE at the end of the dialog.

I don't see that there is anything wrong with the first two responses.

Same problem if tls is used instead of tcp.


Thanks in advance

-- 
Stefan Tichy  ( asterisk2 at pi4tel dot de )
-------------- next part --------------

INVITE sip:snom2 at 192.168.17.22:2097;transport=tcp;line=vzbq7tqh SIP/2.0
Via: SIP/2.0/TCP 192.168.17.5:5060;branch=z9hG4bK0b40f361;rport
Max-Forwards: 70
From: "Stefan Tichy" <sip:24 at 192.168.17.5>;tag=as6ab21f9b
To: <sip:snom2 at 192.168.17.22:2097;transport=tcp;line=vzbq7tqh>
Contact: <sip:24 at 192.168.17.5;transport=TCP>
Call-ID: 35dd42a56b205c6f072f8eaa2fbd1bf0 at 192.168.17.5
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.0-beta4
Date: Thu, 20 Aug 2009 16:40:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 812536728 812536728 IN IP4 192.168.17.5
s=Asterisk PBX 1.6.2.0-beta4
c=IN IP4 192.168.17.5
t=0 0
m=audio 10204 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Aug 20 18:40:02] VERBOSE[15681] app_dial.c:     -- Called snom2
[Aug 20 18:40:02] VERBOSE[15656] chan_sip.c: 
<--- SIP read from TCP:192.168.17.22:2097 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.17.5:5060;branch=z9hG4bK0b40f361;rport=5060
From: "Stefan Tichy" <sip:24 at 192.168.17.5>;tag=as6ab21f9b
To: <sip:snom2 at 192.168.17.22:2097;transport=tcp;line=vzbq7tqh>;tag=53aoekj9df
Call-ID: 35dd42a56b205c6f072f8eaa2fbd1bf0 at 192.168.17.5
CSeq: 102 INVITE
Contact: <sip:snom2 at 192.168.17.22:2097;transport=tcp;line=vzbq7tqh>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

<------------->
[Aug 20 18:40:02] VERBOSE[15656] chan_sip.c: --- (10 headers 0 lines) ---
[Aug 20 18:40:13] VERBOSE[15656] chan_sip.c: 
<--- SIP read from TCP:192.168.17.22:2097 --->
SIP/2.0 200 Ok
Via: SIP/2.0/TCP 192.168.17.5:5060;branch=z9hG4bK0b40f361;rport=5060
From: "Stefan Tichy" <sip:24 at 192.168.17.5>;tag=as6ab21f9b
To: <sip:snom2 at 192.168.17.22:2097;transport=tcp;line=vzbq7tqh>;tag=53aoekj9df
Call-ID: 35dd42a56b205c6f072f8eaa2fbd1bf0 at 192.168.17.5
CSeq: 102 INVITE
Contact: <sip:snom2 at 192.168.17.22:2097;transport=tcp;line=vzbq7tqh>;reg-id=1
User-Agent: snom360/7.3.7
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 245

v=0
o=root 1689322625 1689322626 IN IP4 192.168.17.22
s=call
c=IN IP4 192.168.17.22
t=0 0
m=audio 62094 RTP/AVP 8 3 101
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
[Aug 20 18:40:13] VERBOSE[15656] chan_sip.c: --- (13 headers 12 lines) ---
[Aug 20 18:40:17] VERBOSE[15656] chan_sip.c: 
<--- SIP read from TCP:192.168.17.22:2097 --->
BYE sip:24 at 192.168.17.5;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 192.168.17.22:2097;branch=z9hG4bK-b17fj9n3itmx;rport
From: <sip:snom2 at 192.168.17.22:2097;transport=tcp;line=vzbq7tqh>;tag=53aoekj9df
To: "Stefan Tichy" <sip:24 at 192.168.17.5>;tag=as6ab21f9b
Call-ID: 35dd42a56b205c6f072f8eaa2fbd1bf0 at 192.168.17.5
CSeq: 1 BYE
Max-Forwards: 70
Contact: <sip:snom2 at 192.168.17.22:2097;transport=TCP;line=vzbq7tqh>;reg-id=1
User-Agent: snom360/7.3.7
RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=197,Tx_Pkts=197,Remote_Tx_Pkts=0
Content-Length: 0

<------------->
[Aug 20 18:40:17] VERBOSE[15656] chan_sip.c: --- (12 headers 0 lines) ---
[Aug 20 18:40:17] VERBOSE[15656] chan_sip.c: Sending to 192.168.17.22 : 2097 (no NAT)
[Aug 20 18:40:17] VERBOSE[15656] chan_sip.c: 
<--- Transmitting (no NAT) to 192.168.17.22:2097 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.17.22:2097;branch=z9hG4bK-b17fj9n3itmx;received=192.168.17.22;rport=2097
From: <sip:snom2 at 192.168.17.22:2097;transport=tcp;line=vzbq7tqh>;tag=53aoekj9df
To: "Stefan Tichy" <sip:24 at 192.168.17.5>;tag=as6ab21f9b
Call-ID: 35dd42a56b205c6f072f8eaa2fbd1bf0 at 192.168.17.5
CSeq: 1 BYE
Server: Asterisk PBX 1.6.2.0-beta4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0