Chan_sip : to PJSIPSoftphones immediate go unreachable when on mobile if qualify is enabled

Hi , converting to pjsip, i used qualify before, but now with PJSIP, on local lan, all is fine
but on remote, with port 5060 open, and they connect, they go immediately offline after register
if i remove the qualify_frequence, then it works

seems asteriks isnt able to reach the port on phone side? why is that happening?
it worked perfect on chan_sip ?

[Mar 18 14:50:18]     -- Added contact 'sip:6000@100.70.xx.xx:39538;transport=udp' to AOR '6000' with expiration of 3600 seconds
[Mar 18 14:50:21]     -- Contact 6000/sip:6000@100.70.xx.xx:39538;transport=udp is now Unreachable.  RTT: 0.000 msec

Can anyone help me out? i’m still struggling with this…

i used before chan_sip, created a few extensions, opened port 5060 in router and the RTP ports
now with softphones i can register from public, but they are not reachable, i cant dial them, if i turn on qualify_frequancy, you see they go immediately offline after register

what is wrong?

thnx!!

[6000]
type=endpoint
context=default
disallow=all
allow=ulaw,alaw,speex,gsm,g726,g723,g722,opus
allow=h264,vp8,vp9
auth=auth6000
aors=6000
;max_video_streams=5
 
[auth6000]
type=auth
auth_type=userpass
password=1234
username=6000
 
[6000]
type=aor
qualify_frequency=60
;rewrite_contact=yes
max_contacts=5
remove_existing=yes
remove_unavailable=yes

Log

[Mar 20 08:31:38] <--- Received SIP request (618 bytes) from UDP:46.178.123.20:43145 --->
[Mar 20 08:31:38] REGISTER sip:mydomain.com:5060 SIP/2.0
[Mar 20 08:31:38] Via: SIP/2.0/UDP 99.99.56.164:38883;branch=z9hG4bK.Lsjg0dvpx;rport
[Mar 20 08:31:38] From: <sip:6000@mydomain.com>;tag=dN0aveC0u
[Mar 20 08:31:38] To: sip:6000@mydomain.com
[Mar 20 08:31:38] CSeq: 20 REGISTER
[Mar 20 08:31:38] Call-ID: tzEXn9SYw6
[Mar 20 08:31:38] Max-Forwards: 70
[Mar 20 08:31:38] Supported: replaces, outbound
[Mar 20 08:31:38] Accept: application/sdp
[Mar 20 08:31:38] Accept: text/plain
[Mar 20 08:31:38] Accept: application/vnd.gsma.rcs-ft-http+xml
[Mar 20 08:31:38] Contact: <sip:6000@99.99.56.164:38883;transport=udp>;+sip.instance="<urn:uuid:dc2169e5-5870-4374-8d64-1cdb1c9efc6b>"
[Mar 20 08:31:38] Expires: 3600
[Mar 20 08:31:38] User-Agent: LinphoneAndroid/2.1.730 (belle-sip/1.6.3)
[Mar 20 08:31:38] 
[Mar 20 08:31:38] 
[Mar 20 08:31:38] <--- Transmitting SIP response (502 bytes) to UDP:46.178.123.20:43145 --->
[Mar 20 08:31:38] SIP/2.0 401 Unauthorized
[Mar 20 08:31:38] Via: SIP/2.0/UDP 99.99.56.164:38883;rport=43145;received=46.178.123.20;branch=z9hG4bK.Lsjg0dvpx
[Mar 20 08:31:38] Call-ID: tzEXn9SYw6
[Mar 20 08:31:38] From: <sip:6000@mydomain.com>;tag=dN0aveC0u
[Mar 20 08:31:38] To: <sip:6000@mydomain.com>;tag=z9hG4bK.Lsjg0dvpx
[Mar 20 08:31:38] CSeq: 20 REGISTER
[Mar 20 08:31:38] WWW-Authenticate: Digest realm="asterisk",nonce="1647761498/8a7c4f1acb4657b53cd197a37314c12e",opaque="5f8e04cc16e742ef",algorithm=md5,qop="auth"
[Mar 20 08:31:38] Server: Asterisk PBX 18.10.1
[Mar 20 08:31:38] Content-Length:  0
[Mar 20 08:31:38] 
[Mar 20 08:31:38] 
[Mar 20 08:31:38] <--- Received SIP request (918 bytes) from UDP:46.178.123.20:43145 --->
[Mar 20 08:31:38] REGISTER sip:mydomain.com:5060 SIP/2.0
[Mar 20 08:31:38] Via: SIP/2.0/UDP 99.99.56.164:38883;branch=z9hG4bK.HYiqJhuRh;rport
[Mar 20 08:31:38] From: <sip:6000@mydomain.com>;tag=dN0aveC0u
[Mar 20 08:31:38] To: sip:6000@mydomain.com
[Mar 20 08:31:38] CSeq: 21 REGISTER
[Mar 20 08:31:38] Call-ID: tzEXn9SYw6
[Mar 20 08:31:38] Max-Forwards: 70
[Mar 20 08:31:38] Supported: replaces, outbound
[Mar 20 08:31:38] Accept: application/sdp
[Mar 20 08:31:38] Accept: text/plain
[Mar 20 08:31:38] Accept: application/vnd.gsma.rcs-ft-http+xml
[Mar 20 08:31:38] Contact: <sip:6000@99.99.56.164:38883;transport=udp>;+sip.instance="<urn:uuid:dc2169e5-5870-4374-8d64-1cdb1c9efc6b>"
[Mar 20 08:31:38] Expires: 3600
[Mar 20 08:31:38] User-Agent: LinphoneAndroid/2.1.730 (belle-sip/1.6.3)
[Mar 20 08:31:38] Authorization:  Digest realm="asterisk", nonce="1647761498/8a7c4f1acb4657b53cd197a37314c12e", algorithm=md5, opaque="5f8e04cc16e742ef", username="6000",  uri="sip:mydomain.com:5060", response="cd4e74af3842268906bbf60160c54bfa", cnonce="6yhypATueDzVp8xK", nc=00000001, qop=auth
[Mar 20 08:31:38] 
[Mar 20 08:31:38] 
[Mar 20 08:31:38] e[1;30m    -- e[0mAdded contact 'sip:6000@99.99.56.164:38883;transport=udp' to AOR '6000' with expiration of 3600 seconds
[Mar 20 08:31:38] <--- Transmitting SIP response (466 bytes) to UDP:46.178.123.20:43145 --->
[Mar 20 08:31:38] SIP/2.0 200 OK
[Mar 20 08:31:38] Via: SIP/2.0/UDP 99.99.56.164:38883;rport=43145;received=46.178.123.20;branch=z9hG4bK.HYiqJhuRh
[Mar 20 08:31:38] Call-ID: tzEXn9SYw6
[Mar 20 08:31:38] From: <sip:6000@mydomain.com>;tag=dN0aveC0u
[Mar 20 08:31:38] To: <sip:6000@mydomain.com>;tag=z9hG4bK.HYiqJhuRh
[Mar 20 08:31:38] CSeq: 21 REGISTER
[Mar 20 08:31:38] Date: Sun, 20 Mar 2022 07:31:38 GMT
[Mar 20 08:31:38] Contact: <sip:6000@99.99.56.164:38883;transport=udp>;expires=3599
[Mar 20 08:31:38] Expires: 3600
[Mar 20 08:31:38] Server: Asterisk PBX 18.10.1
[Mar 20 08:31:38] Content-Length:  0
[Mar 20 08:31:38] 
[Mar 20 08:31:38] 
[Mar 20 08:31:38] <--- Transmitting SIP request (438 bytes) to UDP:99.99.56.164:38883 --->
[Mar 20 08:31:38] OPTIONS sip:6000@99.99.56.164:38883;transport=udp SIP/2.0
[Mar 20 08:31:38] Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bKPj1b4705f7-34c3-4f04-8ed1-2a17944910dd
[Mar 20 08:31:38] From: <sip:6000@192.168.0.17>;tag=5ef82357-fc48-4c41-8e3c-b118041f16d2
[Mar 20 08:31:38] To: <sip:6000@99.99.56.164>
[Mar 20 08:31:38] Contact: <sip:6000@192.168.0.17:5060>
[Mar 20 08:31:38] Call-ID: b8004b34-266f-4297-b9a0-09d57aaa2c3e
[Mar 20 08:31:38] CSeq: 46660 OPTIONS
[Mar 20 08:31:38] Max-Forwards: 70
[Mar 20 08:31:38] User-Agent: Asterisk PBX 18.10.1
[Mar 20 08:31:38] Content-Length:  0
[Mar 20 08:31:38] 
[Mar 20 08:31:38] 
[Mar 20 08:31:39] <--- Transmitting SIP request (438 bytes) to UDP:99.99.56.164:38883 --->
[Mar 20 08:31:39] OPTIONS sip:6000@99.99.56.164:38883;transport=udp SIP/2.0
[Mar 20 08:31:39] Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bKPj1b4705f7-34c3-4f04-8ed1-2a17944910dd
[Mar 20 08:31:39] From: <sip:6000@192.168.0.17>;tag=5ef82357-fc48-4c41-8e3c-b118041f16d2
[Mar 20 08:31:39] To: <sip:6000@99.99.56.164>
[Mar 20 08:31:39] Contact: <sip:6000@192.168.0.17:5060>
[Mar 20 08:31:39] Call-ID: b8004b34-266f-4297-b9a0-09d57aaa2c3e
[Mar 20 08:31:39] CSeq: 46660 OPTIONS
[Mar 20 08:31:39] Max-Forwards: 70
[Mar 20 08:31:39] User-Agent: Asterisk PBX 18.10.1
[Mar 20 08:31:39] Content-Length:  0
[Mar 20 08:31:39] 
[Mar 20 08:31:39] 
[Mar 20 08:31:39] <--- Received SIP request (918 bytes) from UDP:46.178.123.20:43145 --->
[Mar 20 08:31:39] REGISTER sip:mydomain.com:5060 SIP/2.0
[Mar 20 08:31:39] Via: SIP/2.0/UDP 99.99.56.164:38883;branch=z9hG4bK.~b4WZGwxx;rport
[Mar 20 08:31:39] From: <sip:6000@mydomain.com>;tag=dN0aveC0u
[Mar 20 08:31:39] To: sip:6000@mydomain.com
[Mar 20 08:31:39] CSeq: 22 REGISTER
[Mar 20 08:31:39] Call-ID: tzEXn9SYw6
[Mar 20 08:31:39] Max-Forwards: 70
[Mar 20 08:31:39] Supported: replaces, outbound
[Mar 20 08:31:39] Accept: application/sdp
[Mar 20 08:31:39] Accept: text/plain
[Mar 20 08:31:39] Accept: application/vnd.gsma.rcs-ft-http+xml
[Mar 20 08:31:39] Contact: <sip:6000@99.99.56.164:38883;transport=udp>;+sip.instance="<urn:uuid:dc2169e5-5870-4374-8d64-1cdb1c9efc6b>"
[Mar 20 08:31:39] Expires: 3600
[Mar 20 08:31:39] User-Agent: LinphoneAndroid/2.1.730 (belle-sip/1.6.3)
[Mar 20 08:31:39] Authorization:  Digest realm="asterisk", nonce="1647761498/8a7c4f1acb4657b53cd197a37314c12e", algorithm=md5, opaque="5f8e04cc16e742ef", username="6000",  uri="sip:mydomain.com:5060", response="806c93c53d0b3c8a6252ce490a4bff8c", cnonce="Q9f2CLrtdoM2cmfI", nc=00000002, qop=auth
[Mar 20 08:31:39] 
[Mar 20 08:31:39] 
[Mar 20 08:31:39] <--- Transmitting SIP response (466 bytes) to UDP:46.178.123.20:43145 --->
[Mar 20 08:31:39] SIP/2.0 200 OK
[Mar 20 08:31:39] Via: SIP/2.0/UDP 99.99.56.164:38883;rport=43145;received=46.178.123.20;branch=z9hG4bK.~b4WZGwxx
[Mar 20 08:31:39] Call-ID: tzEXn9SYw6
[Mar 20 08:31:39] From: <sip:6000@mydomain.com>;tag=dN0aveC0u
[Mar 20 08:31:39] To: <sip:6000@mydomain.com>;tag=z9hG4bK.~b4WZGwxx
[Mar 20 08:31:39] CSeq: 22 REGISTER
[Mar 20 08:31:39] Date: Sun, 20 Mar 2022 07:31:39 GMT
[Mar 20 08:31:39] Contact: <sip:6000@99.99.56.164:38883;transport=udp>;expires=3599
[Mar 20 08:31:39] Expires: 3600
[Mar 20 08:31:39] Server: Asterisk PBX 18.10.1
[Mar 20 08:31:39] Content-Length:  0
[Mar 20 08:31:39] 
[Mar 20 08:31:39] 
[Mar 20 08:31:40] <--- Transmitting SIP request (438 bytes) to UDP:99.99.56.164:38883 --->
[Mar 20 08:31:40] OPTIONS sip:6000@99.99.56.164:38883;transport=udp SIP/2.0
[Mar 20 08:31:40] Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bKPj1b4705f7-34c3-4f04-8ed1-2a17944910dd
[Mar 20 08:31:40] From: <sip:6000@192.168.0.17>;tag=5ef82357-fc48-4c41-8e3c-b118041f16d2
[Mar 20 08:31:40] To: <sip:6000@99.99.56.164>
[Mar 20 08:31:40] Contact: <sip:6000@192.168.0.17:5060>
[Mar 20 08:31:40] Call-ID: b8004b34-266f-4297-b9a0-09d57aaa2c3e
[Mar 20 08:31:40] CSeq: 46660 OPTIONS
[Mar 20 08:31:40] Max-Forwards: 70
[Mar 20 08:31:40] User-Agent: Asterisk PBX 18.10.1
[Mar 20 08:31:40] Content-Length:  0
[Mar 20 08:31:40] 
[Mar 20 08:31:40] 
[Mar 20 08:31:41] e[1;30m    -- e[0mContact 6000/sip:6000@99.99.56.164:38883;transport=udp is now Unreachable.  RTT: 0.000 msec
[Mar 20 08:31:42] <--- Transmitting SIP request (438 bytes) to UDP:99.99.56.164:38883 --->
[Mar 20 08:31:42] OPTIONS sip:6000@99.99.56.164:38883;transport=udp SIP/2.0
[Mar 20 08:31:42] Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bKPj1b4705f7-34c3-4f04-8ed1-2a17944910dd
[Mar 20 08:31:42] From: <sip:6000@192.168.0.17>;tag=5ef82357-fc48-4c41-8e3c-b118041f16d2
[Mar 20 08:31:42] To: <sip:6000@99.99.56.164>
[Mar 20 08:31:42] Contact: <sip:6000@192.168.0.17:5060>
[Mar 20 08:31:42] Call-ID: b8004b34-266f-4297-b9a0-09d57aaa2c3e
[Mar 20 08:31:42] CSeq: 46660 OPTIONS
[Mar 20 08:31:42] Max-Forwards: 70
[Mar 20 08:31:42] User-Agent: Asterisk PBX 18.10.1
[Mar 20 08:31:42] Content-Length:  0
[Mar 20 08:31:42] 
[Mar 20 08:31:42] 
[Mar 20 08:31:46] <--- Transmitting SIP request (438 bytes) to UDP:99.99.56.164:38883 --->
[Mar 20 08:31:46] OPTIONS sip:6000@99.99.56.164:38883;transport=udp SIP/2.0
[Mar 20 08:31:46] Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bKPj1b4705f7-34c3-4f04-8ed1-2a17944910dd
[Mar 20 08:31:46] From: <sip:6000@192.168.0.17>;tag=5ef82357-fc48-4c41-8e3c-b118041f16d2
[Mar 20 08:31:46] To: <sip:6000@99.99.56.164>
[Mar 20 08:31:46] Contact: <sip:6000@192.168.0.17:5060>
[Mar 20 08:31:46] Call-ID: b8004b34-266f-4297-b9a0-09d57aaa2c3e
[Mar 20 08:31:46] CSeq: 46660 OPTIONS
[Mar 20 08:31:46] Max-Forwards: 70
[Mar 20 08:31:46] User-Agent: Asterisk PBX 18.10.1
[Mar 20 08:31:46] Content-Length:  0
[Mar 20 08:31:46] 
[Mar 20 08:31:46] 
[Mar 20 08:31:49] e[1;30m    -- e[0mRemote UNIX connection disconnected
[Mar 20 08:31:50] <--- Transmitting SIP request (438 bytes) to UDP:99.99.56.164:38883 --->
[Mar 20 08:31:50] OPTIONS sip:6000@99.99.56.164:38883;transport=udp SIP/2.0
[Mar 20 08:31:50] Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bKPj1b4705f7-34c3-4f04-8ed1-2a17944910dd
[Mar 20 08:31:50] From: <sip:6000@192.168.0.17>;tag=5ef82357-fc48-4c41-8e3c-b118041f16d2
[Mar 20 08:31:50] To: <sip:6000@99.99.56.164>
[Mar 20 08:31:50] Contact: <sip:6000@192.168.0.17:5060>
[Mar 20 08:31:50] Call-ID: b8004b34-266f-4297-b9a0-09d57aaa2c3e
[Mar 20 08:31:50] CSeq: 46660 OPTIONS
[Mar 20 08:31:50] Max-Forwards: 70
[Mar 20 08:31:50] User-Agent: Asterisk PBX 18.10.1
[Mar 20 08:31:50] Content-Length:  0
[Mar 20 08:31:50] 
[Mar 20 08:31:50] 
[Mar 20 08:31:54] <--- Transmitting SIP request (438 bytes) to UDP:99.99.56.164:38883 --->
[Mar 20 08:31:54] OPTIONS sip:6000@99.99.56.164:38883;transport=udp SIP/2.0
[Mar 20 08:31:54] Via: SIP/2.0/UDP 192.168.0.17:5060;rport;branch=z9hG4bKPj1b4705f7-34c3-4f04-8ed1-2a17944910dd
[Mar 20 08:31:54] From: <sip:6000@192.168.0.17>;tag=5ef82357-fc48-4c41-8e3c-b118041f16d2
[Mar 20 08:31:54] To: <sip:6000@99.99.56.164>
[Mar 20 08:31:54] Contact: <sip:6000@192.168.0.17:5060>
[Mar 20 08:31:54] Call-ID: b8004b34-266f-4297-b9a0-09d57aaa2c3e
[Mar 20 08:31:54] CSeq: 46660 OPTIONS
[Mar 20 08:31:54] Max-Forwards: 70
[Mar 20 08:31:54] User-Agent: Asterisk PBX 18.10.1
[Mar 20 08:31:54] Content-Length:  0
[Mar 20 08:31:54] 
[Mar 20 08:31:54]

You have no NAT options enabled on the endpoint, so Asterisk is trying to contact it at the IP address and port it has provided in the Contact. It said to contact it at 99.99.56.164 port 38883 - and that’s what Asterisk is doing. But it’s not reachable there.

If you want to enable NAT support in Asterisk for the endpoint, then you’d set the “rewrite_contact” option to “yes” and the “rtp_symmetric” option to “yes”.

ah, was indeed just reading about it

also this one? force_rport=yes ?

i have configured port forwarding on router from 9988 to 5060 , to make it less obvious :slight_smile:

The force_rport option is already set to yes by default.

Thnx for feedback, appreciated!!

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