user agent become unreachable on asterisk cli
muhamm...@admaxim.com
James Criscuolo
Justin Graham
On 30 May 2020, at 2:32 am, James Criscuolo <ja...@onsip.com> wrote:
Logs with trace sip enabled in a gist please.
On Friday, May 29, 2020 at 10:42:52 AM UTC-4, muhamm...@admaxim.com wrote:user-agent register on asterisk 16.10.0 but become unreachable, tried with sipml5 works fine there but in case sip.js or ctxsip it becomes unreachable
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Justin Graham
On 30 May 2020, at 2:32 am, James Criscuolo <ja...@onsip.com> wrote:
Logs with trace sip enabled in a gist please.
On Friday, May 29, 2020 at 10:42:52 AM UTC-4, muhamm...@admaxim.com wrote:user-agent register on asterisk 16.10.0 but become unreachable, tried with sipml5 works fine there but in case sip.js or ctxsip it becomes unreachable
--
Muhammad Zain
debugger logs :-
-- Registered SIP '1061' at 119.160.118.159:51097
[May 30 20:00:09] ERROR[3232]: chan_sip.c:4344 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
[May 30 20:00:13] NOTICE[1111]: chan_sip.c:30545 sip_poke_noanswer: Peer '1061' is now UNREACHABLE! Last qualify: 0
ip-192-168-2-179*CLI> SIP set debug peer 1061
SIP Debugging Enabled for IP: 119.160.118.159
Reliably Transmitting (NAT) to 119.160.118.159:51097:
OPTIONS sip:qtel...@58dc6ttjhp38.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.2.179:5060;branch=z9hG4bK11719d37;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@192.168.2.179>;tag=as16ede78d
To: <sip:qtel...@58dc6ttjhp38.invalid;transport=ws>
Contact: <sip:aste...@192.168.2.179:5060;transport=ws>
Call-ID: 0ea798a904d5d54c...@192.168.2.179:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.10.0
Date: Sat, 30 May 2020 15:00:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[May 30 20:00:23] ERROR[1111]: chan_sip.c:4344 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
Really destroying SIP dialog '0ea798a904d5d54c...@192.168.2.179:5060' Method: OPTIONS
<--- SIP read from WS:119.160.118.159:51097 --->
INVITE sip:10...@52.15.202.126 SIP/2.0
Via: SIP/2.0/WSS 58dc6ttjhp38.invalid;branch=z9hG4bK6232492
Max-Forwards: 70
To: <sip:10...@52.15.202.126>
From: "1061" <sip:10...@52.15.202.126>;tag=3d6qd8aknj
Call-ID: 6eoain49tac83mlu25nu
CSeq: 6557 INVITE
Contact: <sip:qtel...@58dc6ttjhp38.invalid;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.7.8
Content-Type: application/sdp
Content-Length: 2283
v=0
o=- 1630447477073986558 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS ggDdjlX6em2z7nNjbQKyafYeg47bH73uGdKb
m=audio 29667 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 119.160.118.159
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2250720258 1 udp 2122260223 10.35.100.184 60927 typ host generation 0 network-id 2 network-cost 50
a=candidate:398949648 1 udp 2122194687 192.168.43.16 50223 typ host generation 0 network-id 1 network-cost 10
a=candidate:3366238450 1 tcp 1518280447 10.35.100.184 9 typ host tcptype active generation 0 network-id 2 network-cost 50
a=candidate:1497661920 1 tcp 1518214911 192.168.43.16 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=candidate:2525985700 1 udp 1685987071 119.160.118.159 29667 typ srflx raddr 192.168.43.16 rport 50223 generation 0 network-id 1 network-cost 10
a=ice-ufrag:vown
a=ice-pwd:Kt2cP3DHqSnlHq68I0xEhwir
a=ice-options:trickle
a=fingerprint:sha-256 B4:87:43:3C:C1:AB:B3:90:20:AE:E5:17:18:21:58:62:4C:ED:87:F2:95:6C:98:46:FF:C8:89:A9:C6:9B:A5:46
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:ggDdjlX6em2z7nNjbQKyafYeg47bH73uGdKb fba6c5b2-50e6-4efe-b27e-4936719e610c
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2633459723 cname:wLfc3FQShW5JiVwF
a=ssrc:2633459723 msid:ggDdjlX6em2z7nNjbQKyafYeg47bH73uGdKb fba6c5b2-50e6-4efe-b27e-4936719e610c
a=ssrc:2633459723 mslabel:ggDdjlX6em2z7nNjbQKyafYeg47bH73uGdKb
a=ssrc:2633459723 label:fba6c5b2-50e6-4efe-b27e-4936719e610c
<------------->
--- (13 headers 48 lines) ---
Using INVITE request as basis request - 6eoain49tac83mlu25nu
Found peer '1061' for '1061' from 119.160.118.159:51097
<--- Reliably Transmitting (NAT) to 119.160.118.159:51097 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 58dc6ttjhp38.invalid;branch=z9hG4bK6232492;received=119.160.118.159;rport=51097
From: "1061" <sip:10...@52.15.202.126>;tag=3d6qd8aknj
To: <sip:10...@52.15.202.126>;tag=as146111c7
Call-ID: 6eoain49tac83mlu25nu
CSeq: 6557 INVITE
Server: Asterisk PBX 16.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="52.15.202.126", nonce="136647f1"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '6eoain49tac83mlu25nu' in 6400 ms (Method: INVITE)
<--- SIP read from WS:119.160.118.159:51097 --->
ACK sip:10...@52.15.202.126 SIP/2.0
Via: SIP/2.0/WSS 58dc6ttjhp38.invalid;branch=z9hG4bK6232492
To: <sip:10...@52.15.202.126>;tag=as146111c7
From: "1061" <sip:10...@52.15.202.126>;tag=3d6qd8aknj
Call-ID: 6eoain49tac83mlu25nu
Content-Length: 0
CSeq: 6557 ACK
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from WS:119.160.118.159:51097 --->
INVITE sip:10...@52.15.202.126 SIP/2.0
Via: SIP/2.0/WSS 58dc6ttjhp38.invalid;branch=z9hG4bK3704401
Max-Forwards: 70
To: <sip:10...@52.15.202.126>
From: "1061" <sip:10...@52.15.202.126>;tag=3d6qd8aknj
Call-ID: 6eoain49tac83mlu25nu
CSeq: 6558 INVITE
Authorization: Digest algorithm=MD5, username="1061", realm="52.15.202.126", nonce="136647f1", uri="sip:10...@52.15.202.126", response="a132ccc61d902b2f388bb3c25acb462b"
Contact: <sip:qtel...@58dc6ttjhp38.invalid;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.7.8
Content-Type: application/sdp
Content-Length: 2283
v=0
o=- 1630447477073986558 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS ggDdjlX6em2z7nNjbQKyafYeg47bH73uGdKb
m=audio 29667 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 119.160.118.159
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2250720258 1 udp 2122260223 10.35.100.184 60927 typ host generation 0 network-id 2 network-cost 50
a=candidate:398949648 1 udp 2122194687 192.168.43.16 50223 typ host generation 0 network-id 1 network-cost 10
a=candidate:3366238450 1 tcp 1518280447 10.35.100.184 9 typ host tcptype active generation 0 network-id 2 network-cost 50
a=candidate:1497661920 1 tcp 1518214911 192.168.43.16 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=candidate:2525985700 1 udp 1685987071 119.160.118.159 29667 typ srflx raddr 192.168.43.16 rport 50223 generation 0 network-id 1 network-cost 10
a=ice-ufrag:vown
a=ice-pwd:Kt2cP3DHqSnlHq68I0xEhwir
a=ice-options:trickle
a=fingerprint:sha-256 B4:87:43:3C:C1:AB:B3:90:20:AE:E5:17:18:21:58:62:4C:ED:87:F2:95:6C:98:46:FF:C8:89:A9:C6:9B:A5:46
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:ggDdjlX6em2z7nNjbQKyafYeg47bH73uGdKb fba6c5b2-50e6-4efe-b27e-4936719e610c
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2633459723 cname:wLfc3FQShW5JiVwF
a=ssrc:2633459723 msid:ggDdjlX6em2z7nNjbQKyafYeg47bH73uGdKb fba6c5b2-50e6-4efe-b27e-4936719e610c
a=ssrc:2633459723 mslabel:ggDdjlX6em2z7nNjbQKyafYeg47bH73uGdKb
a=ssrc:2633459723 label:fba6c5b2-50e6-4efe-b27e-4936719e610c
<------------->
--- (14 headers 48 lines) ---
Using INVITE request as basis request - 6eoain49tac83mlu25nu
Found peer '1061' for '1061' from 119.160.118.159:51097
== Using SIP RTP CoS mark 5
Got SDP version 2 and unique parts [- 1630447477073986558 IN IP4 127.0.0.1]
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 110
Found RTP audio format 112
Found RTP audio format 113
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found unknown media description format telephone-event for ID 110
Found unknown media description format telephone-event for ID 112
Found unknown media description format telephone-event for ID 113
Found audio description format telephone-event for ID 126
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 119.160.118.159:29667
Looking for 1061 in default (domain 52.15.202.126)
sip_route_dump: route/path hop: <sip:qtel...@58dc6ttjhp38.invalid;transport=ws;ob>
<--- Transmitting (NAT) to 119.160.118.159:51097 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 58dc6ttjhp38.invalid;branch=z9hG4bK3704401;received=119.160.118.159;rport=51097
From: "1061" <sip:10...@52.15.202.126>;tag=3d6qd8aknj
To: <sip:10...@52.15.202.126>
Call-ID: 6eoain49tac83mlu25nu
CSeq: 6558 INVITE
Server: Asterisk PBX 16.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:10...@192.168.2.179:5060;transport=ws>
Content-Length: 0
<------------>
-- Executing [1061@default:1] Dial("SIP/1061-00000000", "SIP/1061") in new stack
[May 30 20:00:36] WARNING[3236][C-00000001]: app_dial.c:2576 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
-- No devices or endpoints to dial (technology/resource)
-- Auto fallthrough, channel 'SIP/1061-00000000' status is 'CHANUNAVAIL'
<--- Reliably Transmitting (NAT) to 119.160.118.159:51097 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/WSS 58dc6ttjhp38.invalid;branch=z9hG4bK3704401;received=119.160.118.159;rport=51097
From: "1061" <sip:10...@52.15.202.126>;tag=3d6qd8aknj
To: <sip:10...@52.15.202.126>;tag=as2ae6a58a
Call-ID: 6eoain49tac83mlu25nu
CSeq: 6558 INVITE
Server: Asterisk PBX 16.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0
<------------>
<--- SIP read from WS:119.160.118.159:51097 --->
ACK sip:10...@52.15.202.126 SIP/2.0
Via: SIP/2.0/WSS 58dc6ttjhp38.invalid;branch=z9hG4bK3704401
To: <sip:10...@52.15.202.126>;tag=as2ae6a58a
From: "1061" <sip:10...@52.15.202.126>;tag=3d6qd8aknj
Call-ID: 6eoain49tac83mlu25nu
Content-Length: 0
CSeq: 6558 ACK
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '6eoain49tac83mlu25nu' Method: ACK
Reliably Transmitting (NAT) to 119.160.118.159:51097:
OPTIONS sip:qtel...@58dc6ttjhp38.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.2.179:5060;branch=z9hG4bK3804e3ab;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@192.168.2.179>;tag=as2062ea27
To: <sip:qtel...@58dc6ttjhp38.invalid;transport=ws>
Contact: <sip:aste...@192.168.2.179:5060;transport=ws>
Call-ID: 4ed803d571ab2dd1...@192.168.2.179:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.10.0
Date: Sat, 30 May 2020 15:00:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[May 30 20:00:37] ERROR[1111]: chan_sip.c:4344 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
-------------------------------------------------------------------
thats the end point
[1061] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=password ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=udp,ws,wss ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=fingerprint ; Tell Asterisk to verify DTLS fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
rtcp_mux=yes ; Tell Asterisk to do RTCP mux
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Muhammad Zain
Eric Green
On Saturday, May 30, 2020 at 2:32:31 PM UTC-4, Muhammad Zain wrote:
same issue at jsSip but when i add contact_uri parameter then it solved, does sip.js has the same kind of parameter ?Muhammad Zain HaiderSoftware EngineerLondon | San Francisco | New York | Sydney | Paris | Dubai25 Sharif Colony, Canal Park, Gulberg II, Lahore, Pakistan.Pinpoint - The New Advanced Footfall Solution From AdMaximContents of this email message and any files attached are confidential and may contain confidential/privileged information. If you are not the intended recipient please notify the sender by replying to this e-mail and please delete the message from your system immediately. You should not copy it or use it for any purpose nor disclose its contents to any other person.
On Sat, May 30, 2020 at 8:05 PM Muhammad Zain <muhamm...@admaxim.com> wrote:
i just debug and found this,
debugger logs :-
-- Registered SIP '1061' at 119.160.118.159:51097
[May 30 20:00:09] ERROR[3232]: chan_sip.c:4344 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
[May 30 20:00:13] NOTICE[1111]: chan_sip.c:30545 sip_poke_noanswer: Peer '1061' is now UNREACHABLE! Last qualify: 0
ip-192-168-2-179*CLI> SIP set debug peer 1061
SIP Debugging Enabled for IP: 119.160.118.159
Reliably Transmitting (NAT) to 119.160.118.159:51097:
OPTIONS sip:qtel6nbk@58dc6ttjhp38.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.2.179:5060;branch=z9hG4bK11719d37;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@192.168.2.179>;tag=as16ede78d
To: <sip:qtel6nbk@58dc6ttjhp38.invalid;transport=ws>
Contact: <sip:aste...@192.168.2.179:5060;transport=ws>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.10.0
Date: Sat, 30 May 2020 15:00:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[May 30 20:00:23] ERROR[1111]: chan_sip.c:4344 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
Really destroying SIP dialog '0ea798a904d5d54c1887bee913841e...@192.168.2.179:5060' Method: OPTIONS
<--- SIP read from WS:119.160.118.159:51097 --->
INVITE sip:10...@52.15.202.126 SIP/2.0
Via: SIP/2.0/WSS 58dc6ttjhp38.invalid;branch=z9hG4bK6232492
Max-Forwards: 70
To: <sip:10...@52.15.202.126>
From: "1061" <sip:10...@52.15.202.126>;tag=3d6qd8aknj
Call-ID: 6eoain49tac83mlu25nu
CSeq: 6557 INVITE
Contact: <sip:qtel6nbk@58dc6ttjhp38.invalid;transport=ws;ob>
Contact: <sip:qtel6nbk@58dc6ttjhp38.invalid;transport=ws;ob>
sip_route_dump: route/path hop: <sip:qtel6nbk@58dc6ttjhp38.invalid;transport=ws;ob>
OPTIONS sip:qtel6nbk@58dc6ttjhp38.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.2.179:5060;branch=z9hG4bK3804e3ab;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@192.168.2.179>;tag=as2062ea27
To: <sip:qtel6nbk@58dc6ttjhp38.invalid;transport=ws>
Contact: <sip:aste...@192.168.2.179:5060;transport=ws>
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Muhammad Zain
Hi. We are aware of this issue and are tracking it on Github - https://github.com/onsip/SIP.js/issues/791
On Saturday, May 30, 2020 at 2:32:31 PM UTC-4, Muhammad Zain wrote:
same issue at jsSip but when i add contact_uri parameter then it solved, does sip.js has the same kind of parameter ?Muhammad Zain HaiderSoftware EngineerLondon | San Francisco | New York | Sydney | Paris | Dubai25 Sharif Colony, Canal Park, Gulberg II, Lahore, Pakistan.Pinpoint - The New Advanced Footfall Solution From AdMaximContents of this email message and any files attached are confidential and may contain confidential/privileged information. If you are not the intended recipient please notify the sender by replying to this e-mail and please delete the message from your system immediately. You should not copy it or use it for any purpose nor disclose its contents to any other person.
On Sat, May 30, 2020 at 8:05 PM Muhammad Zain <muhamm...@admaxim.com> wrote:
i just debug and found this,
debugger logs :-
-- Registered SIP '1061' at 119.160.118.159:51097
[May 30 20:00:09] ERROR[3232]: chan_sip.c:4344 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
[May 30 20:00:13] NOTICE[1111]: chan_sip.c:30545 sip_poke_noanswer: Peer '1061' is now UNREACHABLE! Last qualify: 0
ip-192-168-2-179*CLI> SIP set debug peer 1061
SIP Debugging Enabled for IP: 119.160.118.159
Reliably Transmitting (NAT) to 119.160.118.159:51097:
OPTIONS sip:qtel...@58dc6ttjhp38.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.2.179:5060;branch=z9hG4bK11719d37;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@192.168.2.179>;tag=as16ede78d
To: <sip:qtel...@58dc6ttjhp38.invalid;transport=ws>
Contact: <sip:aste...@192.168.2.179:5060;transport=ws>
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.10.0
Date: Sat, 30 May 2020 15:00:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
[May 30 20:00:23] ERROR[1111]: chan_sip.c:4344 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
Really destroying SIP dialog '0ea798a904d5d54c...@192.168.2.179:5060' Method: OPTIONS
<--- SIP read from WS:119.160.118.159:51097 --->
INVITE sip:10...@52.15.202.126 SIP/2.0
Via: SIP/2.0/WSS 58dc6ttjhp38.invalid;branch=z9hG4bK6232492
Max-Forwards: 70
To: <sip:10...@52.15.202.126>
From: "1061" <sip:10...@52.15.202.126>;tag=3d6qd8aknj
Call-ID: 6eoain49tac83mlu25nu
CSeq: 6557 INVITE
Contact: <sip:qtel...@58dc6ttjhp38.invalid;transport=ws;ob>
Contact: <sip:qtel...@58dc6ttjhp38.invalid;transport=ws;ob>
sip_route_dump: route/path hop: <sip:qtel...@58dc6ttjhp38.invalid;transport=ws;ob>
OPTIONS sip:qtel...@58dc6ttjhp38.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.2.179:5060;branch=z9hG4bK3804e3ab;rport
Max-Forwards: 70
From: "asterisk" <sip:aste...@192.168.2.179>;tag=as2062ea27
To: <sip:qtel...@58dc6ttjhp38.invalid;transport=ws>
Contact: <sip:aste...@192.168.2.179:5060;transport=ws>
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MD. AMIRUL ISLAM
contact = response.parseHeader('contact', contacts);
if(contact.uri.user === this.ua.contact.uri.user) {
expires = contact.getParam('expires');
break;
} else {
contact = null;
}
}
Via: SIP/2.0/WS 20.147.223.220:5060;branch=z9hG4bK7b866875;rport
Max-Forwards: 70
From: "Unknown" <sip:Unk...@20.147.223.220>;tag=as1735d89c
To: <sip:089q...@1ejlevqrbh76.invalid;transport=ws>
Contact: <sip:Unk...@20.147.223.220:5060;transport=ws>
Call-ID: 4c2b11167087ddc7...@20.147.223.220:5060
CSeq: 102 OPTIONS
User-Agent: IPBX-2.11.0(13.30.0)
Date: Mon, 09 Oct 2023 20:36:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0