Hi all ,
I am used asterisk Asterisk certified/13.18-cert3 with pjsip for dialog.
I have a strange dialog with a autocom astra 5000 , i don’t understand the BYE of ip :10.0.0.180(alcatel phone).
see attached picture.
Can you help me ?
Thx.
Thy
You’d need to provide further information about which IP address maps to what. You’d also need to provide full SIP messages (such as using pjsip set logger on) with a console log. There’s just not enough information here to guess.
Below the pjsip logger on .
Thx for you help
<--- Received SIP request (891 bytes) from UDP:10.0.0.180:5060 --->
INVITE sip:4100@10.0.9.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.180;branch=z9hG4bK8f89.b0abfa4628512920eb97955110042247.0
Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK_1_948cdac_ctxe_00001981_uumid_d0d3a022
From: "Medecin de garde" <sip:3600@10.0.0.180>;tag=680024DB_nab_0047_isp_0045_cco_3902_igo_5BE7_mgt_FFFF
To: <sip:4100@10.0.9.7:5060>
Call-ID: 0201FFFF97FFDB24
CSeq: 1 INVITE
Contact: <sip:3600@10.0.0.180:5060;ctxe=00001981>
Max-Forwards: 15
P-Asserted-Identity: "Medecin de garde" <sip:3600@10.0.0.180>
Privacy: none
User-Agent: A5000 R6.2 /A700 FRA
Content-Type: application/sdp
Content-Length: 265
v=0
o=- 0 0 IN IP4 10.0.0.180
s=-
c=IN IP4 10.0.0.191
t=0 0
m=audio 16368 RTP/AVP 0 8 18 101
a=rtcp:16369
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<--- Transmitting SIP response (460 bytes) to UDP:10.0.0.180:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.180;rport=5060;received=10.0.0.180;branch=z9hG4bK8f89.b0abfa4628512920eb97955110042247.0
Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK_1_948cdac_ctxe_00001981_uumid_d0d3a022
Call-ID: 0201FFFF97FFDB24
From: "Medecin de garde" <sip:3600@10.0.0.180>;tag=680024DB_nab_0047_isp_0045_cco_3902_igo_5BE7_mgt_FFFF
To: <sip:4100@10.0.9.7>
CSeq: 1 INVITE
Server: XEVIDEO-6B 1.5.0 (20190430_15h52)
Content-Length: 0
<--- Transmitting SIP response (893 bytes) to UDP:10.0.0.180:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.180;rport=5060;received=10.0.0.180;branch=z9hG4bK8f89.b0abfa4628512920eb97955110042247.0
Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK_1_948cdac_ctxe_00001981_uumid_d0d3a022
Call-ID: 0201FFFF97FFDB24
From: "Medecin de garde" <sip:3600@10.0.0.180>;tag=680024DB_nab_0047_isp_0045_cco_3902_igo_5BE7_mgt_FFFF
To: <sip:4100@10.0.9.7>;tag=82CRJjLC5FYe5B5TDE2IYBPvfrb1epIp
CSeq: 1 INVITE
Server: XEVIDEO-6B 1.5.0 (20190430_15h52)
Contact: <sip:10.0.9.7:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Content-Type: application/sdp
Content-Length: 211
v=0
o=- 0 2 IN IP4 10.0.9.7
s=Asterisk
c=IN IP4 10.0.9.7
t=0 0
m=audio 10800 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Transmitting SIP response (927 bytes) to UDP:10.0.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.180;rport=5060;received=10.0.0.180;branch=z9hG4bK8f89.b0abfa4628512920eb97955110042247.0
Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK_1_948cdac_ctxe_00001981_uumid_d0d3a022
Call-ID: 0201FFFF97FFDB24
From: "Medecin de garde" <sip:3600@10.0.0.180>;tag=680024DB_nab_0047_isp_0045_cco_3902_igo_5BE7_mgt_FFFF
To: <sip:4100@10.0.9.7>;tag=82CRJjLC5FYe5B5TDE2IYBPvfrb1epIp
CSeq: 1 INVITE
Server: XEVIDEO-6B 1.5.0 (20190430_15h52)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:10.0.9.7:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 211
v=0
o=- 0 2 IN IP4 10.0.9.7
s=Asterisk
c=IN IP4 10.0.9.7
t=0 0
m=audio 10800 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP request (535 bytes) from UDP:10.0.0.180:5060 --->
ACK sip:10.0.9.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.180;branch=z9hG4bK8f89.abb38f82e1c56bf18c75341d268bb864.0
Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK_4_71a0a86b_ctxe_00001981_uumid_d0d3a022
From: "Medecin de garde" <sip:3600@10.0.0.180>;tag=680024DB_nab_0047_isp_0045_cco_3902_igo_5BE7_mgt_FFFF
To: <sip:4100@10.0.9.7>;tag=82CRJjLC5FYe5B5TDE2IYBPvfrb1epIp
Call-ID: 0201FFFF97FFDB24
CSeq: 1 ACK
Contact: <sip:3600@10.0.0.180:5060;ctxe=00001981>
Max-Forwards: 15
User-Agent: A5000 R6.2 /A700 FRA
Content-Length: 0
<--- Received SIP request (484 bytes) from UDP:10.0.0.180:5060 --->
BYE sip:10.0.9.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.180;branch=z9hG4bK5f89.4dc5eab5d56e4b8139486e6e35ffcb03.0
Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK_3_900b7a16_ctxe_00001981_uumid_d0d3a022
From: "Medecin de garde" <sip:3600@10.0.0.180>;tag=680024DB_nab_0047_isp_0045_cco_3902_igo_5BE7_mgt_FFFF
To: <sip:4100@10.0.9.7>;tag=82CRJjLC5FYe5B5TDE2IYBPvfrb1epIp
Call-ID: 0201FFFF97FFDB24
CSeq: 2 BYE
Max-Forwards: 15
User-Agent: A5000 R6.2 /A700 FRA
Content-Length: 0
<--- Transmitting SIP response (491 bytes) to UDP:10.0.0.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.180;rport=5060;received=10.0.0.180;branch=z9hG4bK5f89.4dc5eab5d56e4b8139486e6e35ffcb03.0
Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK_3_900b7a16_ctxe_00001981_uumid_d0d3a022
Call-ID: 0201FFFF97FFDB24
From: "Medecin de garde" <sip:3600@10.0.0.180>;tag=680024DB_nab_0047_isp_0045_cco_3902_igo_5BE7_mgt_FFFF
To: <sip:4100@10.0.9.7>;tag=82CRJjLC5FYe5B5TDE2IYBPvfrb1epIp
CSeq: 2 BYE
Server: XEVIDEO-6B 1.5.0 (20190430_15h52)
Content-Length: 0
There is nothing evident or that stands out as to why the endpoint would send a BYE. You’d need to investigate further into it specifically.
Ok ,is there any way ( config) to drop the 180 message ?
Thx
Thy
It’s a 183 session progress, and no - not really. The only option is:
;inband_progress=no ; Determines whether chan_pjsip will indicate ringing
; using inband progress (default: "no")
Which as mentioned already defaults to “no”. Disabling the message entirely would go against what users expect.
Ok jcolp 
Thx a lot
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