hello,
i defined two transport protocols udp and tls
i assigned to one user transport=tlstrans
while the other transport=udptrans
both users registered successfully but when i try to make a call from user1 (tls) to user2 (udp) the call is successful, where is the call could not be reached when i make it from user2 to user1.
is this normal?
If configured properly then calling between any kind of devices works fine.
what do you mean by configured properly?
Generally every time a call doesn’t work, regardless of transport, it’s a configuration issue either in Asterisk or the endpoint.
As it is you haven’t provided any further information about how it’s not working, logging, configuration, etc, so I can’t elaborate any further.
When configured according to the documentation! You are asking us to repeat the documentation here, or at least to write your whole configuration here. You need to provide us with the configuration you tried and logging showing what is failing. We may then have some chance of working out what you are doing wrong.
;====================template
endpoint-basic
type=endpoint
context=phones
disallow=all
allow=gsm,alaw,ulaw,g723
;device_state_busy_at=1
direct_media=yes
dtmf_mode=rfc4733
auth-userpass
type=auth
auth_type=userpass
aor-single-reg
type=aor
max_contacts=1
remove_existing=no
;===============EXTENSION 5000
5000
transport=transport-tls
media_encryption=sdes
media_encryption_optimistic=yes
auth=auth5000
aors=5000
auth5000
password=123
username=5000
;===============EXTENSION 5001
5001
transport=simpletrans
auth=auth5001
aors=5001
auth5001
password=123
username=5001
making the call from 5001 to 5000
<--- Received SIP request (2275 bytes) from UDP:192.168.133.1:53023 --->
PUBLISH sip:5001@192.168.133.99 SIP/2.0
Via: SIP/2.0/UDP 10.6.140.189:53023;rport;branch=z9hG4bKPj40162ab307154243abd74189ecfcf150
Max-Forwards: 70
From: "5001" <sip:5001@192.168.133.99>;tag=2a169eab133f4534873faeb737d03fc8
To: "5001" <sip:5001@192.168.133.99>
Call-ID: 5663bb452ee14dc997da5beec03c7143
CSeq: 1 PUBLISH
Event: presence
Expires: 600
User-Agent: Blink 3.2.0 (Windows)
Content-Type: application/pidf+xml
Content-Length: 1822
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:agp-caps="urn:ag-projects:xml:ns:pidf:caps" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3A5001%40192.168.133.99"><tuple id="SID-62a15cb2-6886-4748-a9f7-faca43ae7fa9"><status><basic>open</basic><agp-pidf:extended>busy</agp-pidf:extended></status><caps:servcaps><caps:audio>true</caps:audio><caps:message>true</caps:message><caps:text>false</caps:text><agp-caps:file-transfer>true</agp-caps:file-transfer><agp-caps:screen-sharing-server>true</agp-caps:screen-sharing-server><agp-caps:screen-sharing-client>true</agp-caps:screen-sharing-client></caps:servcaps><c:display-name>5001</c:display-name><agp-pidf:device-info id="62a15cb2-6886-4748-a9f7-faca43ae7fa9"><agp-pidf:description>John-PC</agp-pidf:description><agp-pidf:user-agent>Blink 3.2.0 (Windows)</agp-pidf:user-agent><agp-pidf:time-offset>120</agp-pidf:time-offset></agp-pidf:device-info><rpid:user-input idle-threshold="600">active</rpid:user-input><dm:deviceID>62a15cb2-6886-4748-a9f7-faca43ae7fa9</dm:deviceID><contact>sip%3A5001%40192.168.133.99</contact><note>On the phone</note><timestamp>2021-12-06T12:43:13.073534+02:00</timestamp></tuple><dm:person id="PID-d5ff37c66914ed204768cfa079568924"><rpid:activities><rpid:busy/></rpid:activities><dm:timestamp>2021-12-06T12:43:13.073534+02:00</dm:timestamp></dm:person><dm:device id="DID-62a15cb2-6886-4748-a9f7-faca43ae7fa9"><dm:deviceID>62a15cb2-6886-4748-a9f7-faca43ae7fa9</dm:deviceID><dm:note>Blink 3.2.0 (Windows) at John-PC</dm:note><dm:timestamp>2021-12-06T12:43:13.073534+02:00</dm:timestamp></dm:device></presence>
<--- Transmitting SIP response (569 bytes) to UDP:192.168.133.1:53023 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.6.140.189:53023;rport=53023;received=192.168.133.1;branch=z9hG4bKPj40162ab307154243abd74189ecfcf150
Call-ID: 5663bb452ee14dc997da5beec03c7143
From: "5001" <sip:5001@192.168.133.99>;tag=2a169eab133f4534873faeb737d03fc8
To: "5001" <sip:5001@192.168.133.99>;tag=z9hG4bKPj40162ab307154243abd74189ecfcf150
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1638787392/ff8e9202f6fdb1e7b2b38c5b10349319",opaque="2e861b257056cce1",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.0.0-rc2
Content-Length: 0
<--- Received SIP request (2569 bytes) from UDP:192.168.133.1:53023 --->
PUBLISH sip:5001@192.168.133.99 SIP/2.0
Via: SIP/2.0/UDP 10.6.140.189:53023;rport;branch=z9hG4bKPjbbc9d189448e492e8d0dbc89ff0ee913
Max-Forwards: 70
From: "5001" <sip:5001@192.168.133.99>;tag=2a169eab133f4534873faeb737d03fc8
To: "5001" <sip:5001@192.168.133.99>
Call-ID: 5663bb452ee14dc997da5beec03c7143
CSeq: 2 PUBLISH
Event: presence
Expires: 600
User-Agent: Blink 3.2.0 (Windows)
Authorization: Digest username="5001", realm="asterisk", nonce="1638787392/ff8e9202f6fdb1e7b2b38c5b10349319", uri="sip:5001@192.168.133.99", response="6edc1900225ba589de816faf42a0523a", algorithm=md5, cnonce="42bf00ae441941a89c71ebe9419a27ed", opaque="2e861b257056cce1", qop=auth, nc=00000001
Content-Type: application/pidf+xml
Content-Length: 1822
<?xml version='1.0' encoding='UTF-8'?>
<presence xmlns:agp-caps="urn:ag-projects:xml:ns:pidf:caps" xmlns:agp-pidf="urn:ag-projects:xml:ns:pidf" xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns="urn:ietf:params:xml:ns:pidf" entity="sip%3A5001%40192.168.133.99"><tuple id="SID-62a15cb2-6886-4748-a9f7-faca43ae7fa9"><status><basic>open</basic><agp-pidf:extended>busy</agp-pidf:extended></status><caps:servcaps><caps:audio>true</caps:audio><caps:message>true</caps:message><caps:text>false</caps:text><agp-caps:file-transfer>true</agp-caps:file-transfer><agp-caps:screen-sharing-server>true</agp-caps:screen-sharing-server><agp-caps:screen-sharing-client>true</agp-caps:screen-sharing-client></caps:servcaps><c:display-name>5001</c:display-name><agp-pidf:device-info id="62a15cb2-6886-4748-a9f7-faca43ae7fa9"><agp-pidf:description>John-PC</agp-pidf:description><agp-pidf:user-agent>Blink 3.2.0 (Windows)</agp-pidf:user-agent><agp-pidf:time-offset>120</agp-pidf:time-offset></agp-pidf:device-info><rpid:user-input idle-threshold="600">active</rpid:user-input><dm:deviceID>62a15cb2-6886-4748-a9f7-faca43ae7fa9</dm:deviceID><contact>sip%3A5001%40192.168.133.99</contact><note>On the phone</note><timestamp>2021-12-06T12:43:13.073534+02:00</timestamp></tuple><dm:person id="PID-d5ff37c66914ed204768cfa079568924"><rpid:activities><rpid:busy/></rpid:activities><dm:timestamp>2021-12-06T12:43:13.073534+02:00</dm:timestamp></dm:person><dm:device id="DID-62a15cb2-6886-4748-a9f7-faca43ae7fa9"><dm:deviceID>62a15cb2-6886-4748-a9f7-faca43ae7fa9</dm:deviceID><dm:note>Blink 3.2.0 (Windows) at John-PC</dm:note><dm:timestamp>2021-12-06T12:43:13.073534+02:00</dm:timestamp></dm:device></presence>
[Dec 6 10:43:12] WARNING[6790]: res_pjsip_pubsub.c:3345 pubsub_on_rx_publish_request: No registered publish handler for event presence from 5001
<--- Transmitting SIP response (420 bytes) to UDP:192.168.133.1:53023 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 10.6.140.189:53023;rport=53023;received=192.168.133.1;branch=z9hG4bKPjbbc9d189448e492e8d0dbc89ff0ee913
Call-ID: 5663bb452ee14dc997da5beec03c7143
From: "5001" <sip:5001@192.168.133.99>;tag=2a169eab133f4534873faeb737d03fc8
To: "5001" <sip:5001@192.168.133.99>;tag=z9hG4bKPjbbc9d189448e492e8d0dbc89ff0ee913
CSeq: 2 PUBLISH
Server: Asterisk PBX 18.0.0-rc2
Content-Length: 0
<--- Received SIP request (997 bytes) from UDP:192.168.133.1:53023 --->
INVITE sip:5000@192.168.133.99 SIP/2.0
Via: SIP/2.0/UDP 10.6.140.189:53023;rport;branch=z9hG4bKPj7328781e3c0544da98b95511f40c2643
Max-Forwards: 70
From: "5001" <sip:5001@192.168.133.99>;tag=01b6048e5fc443b3b2f0819327c7630b
To: <sip:5000@192.168.133.99>
Contact: <sip:29673158@192.168.133.1:53023>
Call-ID: 1871d9bd67bf42319d43614e373342a0
CSeq: 900 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: replaces, norefersub, gruu
User-Agent: Blink 3.2.0 (Windows)
Content-Type: application/sdp
Content-Length: 427
v=0
o=- 3847783393 3847783393 IN IP4 192.168.133.1
s=Blink 3.2.0 (Windows)
t=0 0
m=audio 50018 RTP/AVP 113 9 0 8 101
c=IN IP4 192.168.133.1
a=rtcp:50019
a=rtpmap:113 opus/48000/2
a=fmtp:113 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=zrtp-hash:1.10 77429035c22b68889c4661a82a10adb3106ee9b3e2ed40699bdfb8435f5a8bbe
a=sendrecv
<--- Transmitting SIP response (563 bytes) to UDP:192.168.133.1:53023 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.6.140.189:53023;rport=53023;received=192.168.133.1;branch=z9hG4bKPj7328781e3c0544da98b95511f40c2643
Call-ID: 1871d9bd67bf42319d43614e373342a0
From: "5001" <sip:5001@192.168.133.99>;tag=01b6048e5fc443b3b2f0819327c7630b
To: <sip:5000@192.168.133.99>;tag=z9hG4bKPj7328781e3c0544da98b95511f40c2643
CSeq: 900 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1638787392/ff8e9202f6fdb1e7b2b38c5b10349319",opaque="2c482ada04940b0c",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.0.0-rc2
Content-Length: 0
<--- Received SIP request (416 bytes) from UDP:192.168.133.1:53023 --->
ACK sip:5000@192.168.133.99 SIP/2.0
Via: SIP/2.0/UDP 10.6.140.189:53023;rport;branch=z9hG4bKPj7328781e3c0544da98b95511f40c2643
Max-Forwards: 70
From: "5001" <sip:5001@192.168.133.99>;tag=01b6048e5fc443b3b2f0819327c7630b
To: <sip:5000@192.168.133.99>;tag=z9hG4bKPj7328781e3c0544da98b95511f40c2643
Call-ID: 1871d9bd67bf42319d43614e373342a0
CSeq: 900 ACK
User-Agent: Blink 3.2.0 (Windows)
Content-Length: 0
<--- Received SIP request (1291 bytes) from UDP:192.168.133.1:53023 --->
INVITE sip:5000@192.168.133.99 SIP/2.0
Via: SIP/2.0/UDP 10.6.140.189:53023;rport;branch=z9hG4bKPj143c4059bedb44db8b948660763a6145
Max-Forwards: 70
From: "5001" <sip:5001@192.168.133.99>;tag=01b6048e5fc443b3b2f0819327c7630b
To: <sip:5000@192.168.133.99>
Contact: <sip:29673158@192.168.133.1:53023>
Call-ID: 1871d9bd67bf42319d43614e373342a0
CSeq: 901 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: replaces, norefersub, gruu
User-Agent: Blink 3.2.0 (Windows)
Authorization: Digest username="5001", realm="asterisk", nonce="1638787392/ff8e9202f6fdb1e7b2b38c5b10349319", uri="sip:5000@192.168.133.99", response="d16b9d48d25f6702da59f2e21b4b7792", algorithm=md5, cnonce="9c36ecf3c6d34cb3b46ae1d7f7b14edf", opaque="2c482ada04940b0c", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 427
v=0
o=- 3847783393 3847783393 IN IP4 192.168.133.1
s=Blink 3.2.0 (Windows)
t=0 0
m=audio 50018 RTP/AVP 113 9 0 8 101
c=IN IP4 192.168.133.1
a=rtcp:50019
a=rtpmap:113 opus/48000/2
a=fmtp:113 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=zrtp-hash:1.10 77429035c22b68889c4661a82a10adb3106ee9b3e2ed40699bdfb8435f5a8bbe
a=sendrecv
== Setting global variable 'SIPDOMAIN' to '192.168.133.99'
<--- Transmitting SIP response (365 bytes) to UDP:192.168.133.1:53023 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.6.140.189:53023;rport=53023;received=192.168.133.1;branch=z9hG4bKPj143c4059bedb44db8b948660763a6145
Call-ID: 1871d9bd67bf42319d43614e373342a0
From: "5001" <sip:5001@192.168.133.99>;tag=01b6048e5fc443b3b2f0819327c7630b
To: <sip:5000@192.168.133.99>
CSeq: 901 INVITE
Server: Asterisk PBX 18.0.0-rc2
Content-Length: 0
-- Executing [5000@phones:1] Dial("PJSIP/5001-00000023", "PJSIP/5000") in new stack
-- Called PJSIP/5000
<--- Transmitting SIP request (1092 bytes) to TLS:192.168.133.1:55925 --->
INVITE sip:90354618@192.168.133.1:55925 SIP/2.0
Via: SIP/2.0/TLS 192.168.133.99:5061;rport;branch=z9hG4bKPj4378a878-39a9-449a-86f0-660eed799629;alias
From: "5001" <sip:5001@192.168.133.99>;tag=a9989993-30ab-44a3-a3e7-e579298378fe
To: <sip:90354618@192.168.133.1>
Contact: <sip:asterisk@192.168.133.99:5061;transport=TLS>
Call-ID: e28cff16-230e-403a-9909-b2f82650a9cd
CSeq: 16275 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.0.0-rc2
Content-Type: application/sdp
Content-Length: 394
v=0
o=- 356975371 356975371 IN IP4 192.168.133.99
s=Asterisk
c=IN IP4 192.168.133.99
t=0 0
m=audio 19818 RTP/AVP 8 0 3 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:SeyoCslBACq5od+7iyBKw6/TOrCHqDzFFrfb6ifZ
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
You failed to mark up your logs and configurations as preformatted text, so they are generally garbled.
Your log is incomplete but the call is successful to the point at which the log ends.
You haven’t included your transport definitions.
these are the transport definitions:
[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0
[transport-tls]
type=transport
protocol=tls
bind=192.168.133.99:5061
cert_file=/etc/asterisk/keys/asterisk.crt
priv_key_file=/etc/asterisk/keys/asterisk.key
ca_list_file=/etc/asterisk/keys/ca.crt
method=sslv23
require_client_cert=yes
verify_client=yes
verify_server=yes
and these are the complete logs :
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [5000@phones:2] Hangup("PJSIP/5001-0000002c", "") in new stack
== Spawn extension (phones, 5000, 2) exited non-zero on 'PJSIP/5001-0000002c'
<--- Transmitting SIP response (393 bytes) to UDP:192.168.133.96:10378 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.133.96:10378;rport=10378;received=192.168.133.96;branch=z9hG4bK124709598
Call-ID: 708086240-10378-5@BJC.BGI.BDD.JG
From: "5001" <sip:5001@192.168.133.99>;tag=779522668
To: <sip:5000@192.168.133.99>;tag=8799ddf3-10aa-4371-8492-8f2ccb419aa4
CSeq: 41 INVITE
Server: Asterisk PBX 18.0.0-rc2
Reason: Q.850;cause=18
Content-Length: 0
[Dec 6 11:22:57] ERROR[1205]: cdr_csv.c:275 writefile: Unable to open file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory
[Dec 6 11:22:57] WARNING[1205]: cdr_csv.c:308 csv_log: Unable to write CSV record to master '/var/log/asterisk//cdr-csv//Master.csv' : No such file or directory
<--- Received SIP request (310 bytes) from UDP:192.168.133.96:10378 --->
ACK sip:5000@192.168.133.99 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.96:10378;branch=z9hG4bK124709598;rport
From: "5001" <sip:5001@192.168.133.99>;tag=779522668
To: <sip:5000@192.168.133.99>;tag=8799ddf3-10aa-4371-8492-8f2ccb419aa4
Call-ID: 708086240-10378-5@BJC.BGI.BDD.JG
CSeq: 41 ACK
Content-Length: 0
but i am sorry i could not understand this:
You’ve been using the forum long enough to be aware of this. When adding logs or configurations, you should always mark them up using the “</>” tool in the forum text entry box editing menu. Otherwise various characters are treated as mark up.
I would try turning up the verbosity and seeing if you can get a explicit messages about the timeout that is reporting. However, my best guess is that this is about outgoing TLS, not mixed transports, and is probably the result of a router removing a temporary firewall or NAT rule and throwing away IP packets without sending an ICMP, or resetting the connection.
You should use something like sngrep, or tcpdump and wireshark, to see how well the TLS is working at the TCP level.
Also, it looks like you are screen scraping the logs. You need to enable the full log file and copy from that, or, at least, enable time stamps on the console.
oh yes i am copying the logs from the screen!
how could i enable full log file?
i am very grateful david for your patience.
this is the reason the call is being rejected:
37300 UDP
37400 TLS
a call is made from 37300 to 37400
<--- Received SIP request (1112 bytes) from UDP:192.168.133.1:56059 --->
INVITE sip:37400@192.168.133.98 SIP/2.0
Via: SIP/2.0/UDP 10.6.140.173:56059;rport;branch=z9hG4bKPjbc2d16c779084f7ca2acdf 499647096e
Max-Forwards: 70
From: "37300" <sip:37300@192.168.133.98>;tag=0a8728cdf1d744c98345ca6c3db5e1fd
To: <sip:37400@192.168.133.98>
Contact: <sip:60723459@192.168.133.1:56059>
Call-ID: 121813b08a244eef8a76e80a45af59ea
CSeq: 10738 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: replaces, norefersub, gruu
User-Agent: Blink 3.2.0 (Windows)
Content-Type: application/sdp
Content-Length: 536
v=0
o=- 3848636830 3848636830 IN IP4 192.168.133.1
s=Blink 3.2.0 (Windows)
t=0 0
m=audio 50006 RTP/SAVP 113 9 3 0 8 101
c=IN IP4 192.168.133.1
a=rtcp:50007
a=rtpmap:113 opus/48000/2
a=fmtp:113 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ku5CpXhw0/HuzVZ1rpAm/Lx2GMiq4+nVRgSN6x nf
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:UBAbutMhTwuJblrwNId/GglG9EsY9JEMRStb5X HG
a=sendrecv
<--- Transmitting SIP response (568 bytes) to UDP:192.168.133.1:56059 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.6.140.173:56059;rport=56059;received=192.168.133.1;branch=z9 hG4bKPjbc2d16c779084f7ca2acdf499647096e
Call-ID: 121813b08a244eef8a76e80a45af59ea
From: "37300" <sip:37300@192.168.133.98>;tag=0a8728cdf1d744c98345ca6c3db5e1fd
To: <sip:37400@192.168.133.98>;tag=z9hG4bKPjbc2d16c779084f7ca2acdf499647096e
CSeq: 10738 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1639640820/af44369121d4704ae3b8 59e37fd3bd57",opaque="25c01753170547be",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.0.0-rc2
Content-Length: 0
<--- Received SIP request (422 bytes) from UDP:192.168.133.1:56059 --->
ACK sip:37400@192.168.133.98 SIP/2.0
Via: SIP/2.0/UDP 10.6.140.173:56059;rport;branch=z9hG4bKPjbc2d16c779084f7ca2acdf 499647096e
Max-Forwards: 70
From: "37300" <sip:37300@192.168.133.98>;tag=0a8728cdf1d744c98345ca6c3db5e1fd
To: <sip:37400@192.168.133.98>;tag=z9hG4bKPjbc2d16c779084f7ca2acdf499647096e
Call-ID: 121813b08a244eef8a76e80a45af59ea
CSeq: 10738 ACK
User-Agent: Blink 3.2.0 (Windows)
Content-Length: 0
<--- Received SIP request (1408 bytes) from UDP:192.168.133.1:56059 --->
INVITE sip:37400@192.168.133.98 SIP/2.0
Via: SIP/2.0/UDP 10.6.140.173:56059;rport;branch=z9hG4bKPjae70374e7fcd4b9280d67f 159f3afc7a
Max-Forwards: 70
From: "37300" <sip:37300@192.168.133.98>;tag=0a8728cdf1d744c98345ca6c3db5e1fd
To: <sip:37400@192.168.133.98>
Contact: <sip:60723459@192.168.133.1:56059>
Call-ID: 121813b08a244eef8a76e80a45af59ea
CSeq: 10739 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: replaces, norefersub, gruu
User-Agent: Blink 3.2.0 (Windows)
Authorization: Digest username="37300", realm="asterisk", nonce="1639640820/af44 369121d4704ae3b859e37fd3bd57", uri="sip:37400@192.168.133.98", response="e1dde65 6e1b631d471c5f5f106116057", algorithm=md5, cnonce="dbbaf68a7a3a4d8885d51dd1bb1b1 a8c", opaque="25c01753170547be", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 536
v=0
o=- 3848636830 3848636830 IN IP4 192.168.133.1
s=Blink 3.2.0 (Windows)
t=0 0
m=audio 50006 RTP/SAVP 113 9 3 0 8 101
c=IN IP4 192.168.133.1
a=rtcp:50007
a=rtpmap:113 opus/48000/2
a=fmtp:113 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ku5CpXhw0/HuzVZ1rpAm/Lx2GMiq4+nVRgSN6x nf
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:UBAbutMhTwuJblrwNId/GglG9EsY9JEMRStb5X HG
a=sendrecv
== Setting global variable 'SIPDOMAIN' to '192.168.133.98'
<--- Transmitting SIP response (370 bytes) to UDP:192.168.133.1:56059 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.6.140.173:56059;rport=56059;received=192.168.133.1;branch=z9 hG4bKPjae70374e7fcd4b9280d67f159f3afc7a
Call-ID: 121813b08a244eef8a76e80a45af59ea
From: "37300" <sip:37300@192.168.133.98>;tag=0a8728cdf1d744c98345ca6c3db5e1fd
To: <sip:37400@192.168.133.98>
CSeq: 10739 INVITE
Server: Asterisk PBX 18.0.0-rc2
Content-Length: 0
<--- Transmitting SIP response (424 bytes) to UDP:192.168.133.1:56059 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.6.140.173:56059;rport=56059;received=192.168.133.1;branch=z9 hG4bKPjae70374e7fcd4b9280d67f159f3afc7a
Call-ID: 121813b08a244eef8a76e80a45af59ea
From: "37300" <sip:37300@192.168.133.98>;tag=0a8728cdf1d744c98345ca6c3db5e1fd
To: <sip:37400@192.168.133.98>;tag=63d0fd7e-c03a-49e3-a922-70a125d3a79d
CSeq: 10739 INVITE
Server: Asterisk PBX 18.0.0-rc2
Content-Length: 0
<--- Received SIP request (417 bytes) from UDP:192.168.133.1:56059 --->
ACK sip:37400@192.168.133.98 SIP/2.0
Via: SIP/2.0/UDP 10.6.140.173:56059;rport;branch=z9hG4bKPjae70374e7fcd4b9280d67f 159f3afc7a
Max-Forwards: 70
From: "37300" <sip:37300@192.168.133.98>;tag=0a8728cdf1d744c98345ca6c3db5e1fd
To: <sip:37400@192.168.133.98>;tag=63d0fd7e-c03a-49e3-a922-70a125d3a79d
Call-ID: 121813b08a244eef8a76e80a45af59ea
CSeq: 10739 ACK
User-Agent: Blink 3.2.0 (Windows)
Content-Length: 0
In such cases, it is typically due to mismatched media encryption options. I would have hoped the logs would include more information. They don’t seem to have been turned up.
I would note that sslv* is no no longer considered secure.
what is more secure?
but when a user is udp and the other is TLS it is then normal to have a mismatch in media encryption right?
plus i have this issue.
pjsip.conf:
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
[transport-tls]
type=transport
protocol=tls
bind=192.168.133.99:5061
cert_file=/etc/asterisk/keys/asterisk.crt
priv_key_file=/etc/asterisk/keys/asterisk.key
ca_list_file=/etc/asterisk/keys/ca.crt
method=sslv23
require_client_cert=yes
verify_client=yes
verify_server=yes
;template
endpoint-basic
type=endpoint
context=phones
disallow=all
allow=gsm,ulaw,alaw
direct_media=yes
dtmf_mode=rfc4733
auth-userpass
type=auth
auth_type=userpass
aor-single-reg
type=aor
max_contacts=1
remove_exisiting=no
;==========extension 37100
37100
transport=transport-tls
media_encryption=sdes
auth=auth37100
aors=37100
auth37100
password=123
username=37100
but the thing is that a from BLINK softphone when i add user 37100 it registers successfully on udp protocol of port 5060 whereas it is supposed not to !!!
The problem is with “37300”. You’ve provided configuration for “37100”.
yes true my friend
i already said i have another issue… this issue is related to registration and not the call.
anw here is the full configs of all the users:
;template
endpoint-basic
type=endpoint
context=phones
disallow=all
allow=gsm,ulaw,alaw
direct_media=yes
dtmf_mode=rfc4733
auth-userpass
type=auth
auth_type=userpass
aor-single-reg
type=aor
max_contacts=1
remove_exisiting=no
;==========extension 37100
37100
transport=transport-tls
media_encryption=sdes
auth=auth37100
aors=37100
auth37100
password=123
username=37100
37100
;==========extension 37400
37400
transport=transport-tls
media_encryption=sdes
auth=auth37400
aors=37400
auth37400
password=123
username=37400
37400
;==========extension 37300
37300
transport=transport-udp
auth=auth37300
aors=37300
auth37300
password=123
username=37300
37300
This post is entirely about calling, not registration. Please don’t suddenly ask for help with something else and change what is going on.
“37300” is calling in using SRTP-SDES. Your “37300” endpoint is not configured with it. The call is therefore rejected and fails.
noted my friend i will open another post for it
true !!! thanks that was it!!!
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