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Snom and PJSIP duplicated SIP Invite packet

 I have changed over from Chan_sip to PJSIP and now I experience a problem with my Snom phones. They are sending a double INVITE packet to my Asterisk 16 PBX server and this creating to outgoing channels.


Any suggestion as to the cause of the problem?



<--- Received SIP request (1083 bytes) from UDP:169.255.228.18:2052 --->
INVITE sip:0720163530@pbx-new.desktop.ddns.desktop-ns.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.4.203:2052;branch=z9hG4bK-n25hnb0ohy9f;rport
From: "203" <sip:203@pbx-new.desktop.ddns.desktop-ns.co.za>;tag=7f9xnn7gjo
To: <sip:0720163530@pbx-new.desktop.ddns.desktop-ns.co.za;user=phone>
Call-ID: 3934363730383532313433343331-elxtne2oes9f
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom300/8.7.5.35
Contact: <sip:203@192.168.4.203:2052;line=203ey6y9>;reg-id=1
X-Serialnumber: 0004132FE96D
P-Key-Flags: keys="3"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 996421369 996421369 IN IP4 192.168.4.203
s=call
c=IN IP4 192.168.4.203
t=0 0
m=audio 49580 RTP/AVP 0 8 18 101
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
a=sendrecv

<--- Received SIP request (1083 bytes) from UDP:169.255.228.18:2052 --->
INVITE sip:0720163530@pbx-new.desktop.ddns.desktop-ns.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.4.203:2052;branch=z9hG4bK-n25hnb0ohy9f;rport
From: "203" <sip:203@pbx-new.desktop.ddns.desktop-ns.co.za>;tag=7f9xnn7gjo
To: <sip:0720163530@pbx-new.desktop.ddns.desktop-ns.co.za;user=phone>
Call-ID: 3934363730383532313433343331-elxtne2oes9f
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom300/8.7.5.35
Contact: <sip:203@192.168.4.203:2052;line=203ey6y9>;reg-id=1
X-Serialnumber: 0004132FE96D
P-Key-Flags: keys="3"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 996421369 996421369 IN IP4 192.168.4.203
s=call
c=IN IP4 192.168.4.203
t=0 0
m=audio 49580 RTP/AVP 0 8 18 101
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
a=sendrecv


<--- Transmitting SIP response (564 bytes) to UDP:169.255.228.18:2052 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.4.203:2052;rport=2052;received=169.255.228.18;branch=z9hG4bK-n25hnb0ohy9f
Call-ID: 3934363730383532313433343331-elxtne2oes9f
From: "203" <sip:203@pbx-new.desktop.ddns.desktop-ns.co.za>;tag=7f9xnn7gjo
To: <sip:0720163530@pbx-new.desktop.ddns.desktop-ns.co.za;user=phone>;tag=z9hG4bK-n25hnb0ohy9f
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1603310760/91cf0c30c01fbe779d05e816afe69f5a",opaque="7404df5e5be2aec5",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.14.0
Content-Length:  0

Regards

1 Comment

Hello, 


I saw that you have opened a ticket for this wrong behaviour, we will get back to you there soon.


best regards

snom support

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