No call between two blink accounts

I am trying to make a all between 2 blink accounts!
the call is not happening due to “service unavailable” cause=34.

anyway i assume the problem is that when the blink account is registering it is adding a contact something like this (let’s say the username is 37200): == Contact 37200/sip:34257712@ipOfTheDesktop:randomPort

this number 34257712 might be making this problem leading to sip server not finding this account to make the all to it. meanwhile from blink i can make a call to another softphone application!

You haven’t provided enough information. You’d need to show the actual log of the call attempt in Asterisk.

pjsip.conf

[transport-tls]
type=transport
protocol=tls
bind=192.168.133.5:5061
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=alaw,ulaw,gsm
device_state_busy_at=1
direct_media=no
dtmf_mode=rfc4733
media_encryption=no

[auth-userpass](!)
type=auth
auth_type=userpass

[aor-single-reg](!)
type=aor
max_contacts=1
remove_existing=yes


;==============EXTENSION 37100
[37100](endpoint-basic)
transport=transport-tls
auth=auth37100
aors=37100

[auth37100](auth-userpass)
password=123
username=37100

[37100](aor-single-reg)
;==============EXTENSION 37200
[37200](endpoint-basic)
transport=transport-tls
auth=auth37200
aors=37200

[auth37200](auth-userpass)
password=123
username=37200

[37200](aor-single-reg)

extensions.conf

[phones]


exten => _X.,1,NoOp(${EXTEN})
same =>      n,Dial(PJSIP/${EXTEN})
same =>      n,Hangup()

call from 37200 to 37100:

<--- Received SIP request (2286 bytes) from TLS:192.168.133.90:49984 --->
PUBLISH sip:37200@192.168.133.5 SIP/2.0
Via: SIP/2.0/TLS 192.168.133.90:49984;rport;branch=z9hG4bKPj52651c1dfa4e4e61b57168d884010e56;alias
Max-Forwards: 70
From: "37200" <sip:37200@192.168.133.5>;tag=fc92cca5e32c42e4bdcdeaa51a89d158
To: "37200" <sip:37200@192.168.133.5>
Call-ID: efe04d17c952482caddf37d56b226e68
CSeq: 1 PUBLISH
Event: presence
Expires: 600
User-Agent: Blink 3.2.0 (Windows)
Content-Type: application/pidf+xml
Content-Length:  1823

<?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%3A37200%40192.168.133.5"><tuple id="SID-7f736476-8f2f-43b6-baec-1650c848307f"><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>37200</c:display-name><agp-pidf:device-info id="7f736476-8f2f-43b6-baec-1650c848307f"><agp-pidf:description>user-PC</agp-pidf:description><agp-pidf:user-agent>Blink 3.2.0 (Windows)</agp-pidf:user-agent><agp-pidf:time-offset>180</agp-pidf:time-offset></agp-pidf:device-info><rpid:user-input idle-threshold="600">active</rpid:user-input><dm:deviceID>7f736476-8f2f-43b6-baec-1650c848307f</dm:deviceID><contact>sip%3A37200%40192.168.133.5</contact><note>On the phone</note><timestamp>2022-10-13T11:49:28.127231+03:00</timestamp></tuple><dm:person id="PID-9c28ff1ea108daa61c7c917b690c9e01"><rpid:activities><rpid:busy/></rpid:activities><dm:timestamp>2022-10-13T11:49:28.127231+03:00</dm:timestamp></dm:person><dm:device id="DID-7f736476-8f2f-43b6-baec-1650c848307f"><dm:deviceID>7f736476-8f2f-43b6-baec-1650c848307f</dm:deviceID><dm:note>Blink 3.2.0 (Windows) at user-PC</dm:note><dm:timestamp>2022-10-13T11:49:28.127231+03:00</dm:timestamp></dm:device></presence>
<--- Transmitting SIP response (580 bytes) to TLS:192.168.133.90:49984 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 192.168.133.90:49984;rport=49984;received=192.168.133.90;branch=z9hG4bKPj52651c1dfa4e4e61b57168d884010e56;alias
Call-ID: efe04d17c952482caddf37d56b226e68
From: "37200" <sip:37200@192.168.133.5>;tag=fc92cca5e32c42e4bdcdeaa51a89d158
To: "37200" <sip:37200@192.168.133.5>;tag=z9hG4bKPj52651c1dfa4e4e61b57168d884010e56
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1663256344/af471c4dc156a10e1ce0aff2a8124acb",opaque="5568806e49271cbb",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.0.0-rc2
Content-Length:  0


<--- Received SIP request (2581 bytes) from TLS:192.168.133.90:49984 --->
PUBLISH sip:37200@192.168.133.5 SIP/2.0
Via: SIP/2.0/TLS 192.168.133.90:49984;rport;branch=z9hG4bKPj53321038725e4044a727baae578a5806;alias
Max-Forwards: 70
From: "37200" <sip:37200@192.168.133.5>;tag=fc92cca5e32c42e4bdcdeaa51a89d158
To: "37200" <sip:37200@192.168.133.5>
Call-ID: efe04d17c952482caddf37d56b226e68
CSeq: 2 PUBLISH
Event: presence
Expires: 600
User-Agent: Blink 3.2.0 (Windows)
Authorization: Digest username="37200", realm="asterisk", nonce="1663256344/af471c4dc156a10e1ce0aff2a8124acb", uri="sip:37200@192.168.133.5", response="486b2572923f513058fab2bcad19853e", algorithm=md5, cnonce="52a076a11632473b93fd570fdafcf86f", opaque="5568806e49271cbb", qop=auth, nc=00000001
Content-Type: application/pidf+xml
Content-Length:  1823

<?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%3A37200%40192.168.133.5"><tuple id="SID-7f736476-8f2f-43b6-baec-1650c848307f"><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>37200</c:display-name><agp-pidf:device-info id="7f736476-8f2f-43b6-baec-1650c848307f"><agp-pidf:description>user-PC</agp-pidf:description><agp-pidf:user-agent>Blink 3.2.0 (Windows)</agp-pidf:user-agent><agp-pidf:time-offset>180</agp-pidf:time-offset></agp-pidf:device-info><rpid:user-input idle-threshold="600">active</rpid:user-input><dm:deviceID>7f736476-8f2f-43b6-baec-1650c848307f</dm:deviceID><contact>sip%3A37200%40192.168.133.5</contact><note>On the phone</note><timestamp>2022-10-13T11:49:28.127231+03:00</timestamp></tuple><dm:person id="PID-9c28ff1ea108daa61c7c917b690c9e01"><rpid:activities><rpid:busy/></rpid:activities><dm:timestamp>2022-10-13T11:49:28.127231+03:00</dm:timestamp></dm:person><dm:device id="DID-7f736476-8f2f-43b6-baec-1650c848307f"><dm:deviceID>7f736476-8f2f-43b6-baec-1650c848307f</dm:deviceID><dm:note>Blink 3.2.0 (Windows) at user-PC</dm:note><dm:timestamp>2022-10-13T11:49:28.127231+03:00</dm:timestamp></dm:device></presence>
[Sep 15 15:39:04] WARNING[2414]: res_pjsip_pubsub.c:3345 pubsub_on_rx_publish_request: No registered publish handler for event presence from 37200
<--- Transmitting SIP response (431 bytes) to TLS:192.168.133.90:49984 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/TLS 192.168.133.90:49984;rport=49984;received=192.168.133.90;branch=z9hG4bKPj53321038725e4044a727baae578a5806;alias
Call-ID: efe04d17c952482caddf37d56b226e68
From: "37200" <sip:37200@192.168.133.5>;tag=fc92cca5e32c42e4bdcdeaa51a89d158
To: "37200" <sip:37200@192.168.133.5>;tag=z9hG4bKPj53321038725e4044a727baae578a5806
CSeq: 2 PUBLISH
Server: Asterisk PBX 18.0.0-rc2
Content-Length:  0


<--- Received SIP request (941 bytes) from TLS:192.168.133.90:49984 --->
INVITE sip:37100@192.168.133.5 SIP/2.0
Via: SIP/2.0/TLS 192.168.133.90:49984;rport;branch=z9hG4bKPj4ce7c12e30744ab5a5b70b61a94e077d;alias
Max-Forwards: 70
From: "37200" <sip:37200@192.168.133.5>;tag=e08d4dea55904f0db4f89b0f87424339
To: <sip:37100@192.168.133.5>
Contact: <sip:81036572@192.168.133.90:49982;transport=tls>
Call-ID: 2629e8da9c1d45119fe6f39c72944907
CSeq: 13659 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:   345

v=0
o=- 3874650568 3874650568 IN IP4 192.168.133.90
s=Blink 3.2.0 (Windows)
t=0 0
m=audio 50020 RTP/AVP 113 3 0 8 101
c=IN IP4 192.168.133.90
a=rtcp:50021
a=rtpmap:113 opus/48000/2
a=fmtp:113 useinbandfec=1
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=sendrecv

<--- Transmitting SIP response (575 bytes) to TLS:192.168.133.90:49984 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 192.168.133.90:49984;rport=49984;received=192.168.133.90;branch=z9hG4bKPj4ce7c12e30744ab5a5b70b61a94e077d;alias
Call-ID: 2629e8da9c1d45119fe6f39c72944907
From: "37200" <sip:37200@192.168.133.5>;tag=e08d4dea55904f0db4f89b0f87424339
To: <sip:37100@192.168.133.5>;tag=z9hG4bKPj4ce7c12e30744ab5a5b70b61a94e077d
CSeq: 13659 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1663256344/af471c4dc156a10e1ce0aff2a8124acb",opaque="3fba83d13d8a0e02",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.0.0-rc2
Content-Length:  0


<--- Received SIP request (427 bytes) from TLS:192.168.133.90:49984 --->
ACK sip:37100@192.168.133.5 SIP/2.0
Via: SIP/2.0/TLS 192.168.133.90:49984;rport;branch=z9hG4bKPj4ce7c12e30744ab5a5b70b61a94e077d;alias
Max-Forwards: 70
From: "37200" <sip:37200@192.168.133.5>;tag=e08d4dea55904f0db4f89b0f87424339
To: <sip:37100@192.168.133.5>;tag=z9hG4bKPj4ce7c12e30744ab5a5b70b61a94e077d
Call-ID: 2629e8da9c1d45119fe6f39c72944907
CSeq: 13659 ACK
User-Agent: Blink 3.2.0 (Windows)
Content-Length:  0


<--- Received SIP request (1236 bytes) from TLS:192.168.133.90:49984 --->
INVITE sip:37100@192.168.133.5 SIP/2.0
Via: SIP/2.0/TLS 192.168.133.90:49984;rport;branch=z9hG4bKPjecfa9edbafda4a838665293a3514e381;alias
Max-Forwards: 70
From: "37200" <sip:37200@192.168.133.5>;tag=e08d4dea55904f0db4f89b0f87424339
To: <sip:37100@192.168.133.5>
Contact: <sip:81036572@192.168.133.90:49982;transport=tls>
Call-ID: 2629e8da9c1d45119fe6f39c72944907
CSeq: 13660 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="37200", realm="asterisk", nonce="1663256344/af471c4dc156a10e1ce0aff2a8124acb", uri="sip:37100@192.168.133.5", response="c58994c37c90a83e8e9abbfa7b80cbca", algorithm=md5, cnonce="1824f3e75e8f45d3bdb195b3be9bd6eb", opaque="3fba83d13d8a0e02", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   345

v=0
o=- 3874650568 3874650568 IN IP4 192.168.133.90
s=Blink 3.2.0 (Windows)
t=0 0
m=audio 50020 RTP/AVP 113 3 0 8 101
c=IN IP4 192.168.133.90
a=rtcp:50021
a=rtpmap:113 opus/48000/2
a=fmtp:113 useinbandfec=1
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=sendrecv

  == Setting global variable 'SIPDOMAIN' to '192.168.133.5'
<--- Transmitting SIP response (377 bytes) to TLS:192.168.133.90:49984 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.133.90:49984;rport=49984;received=192.168.133.90;branch=z9hG4bKPjecfa9edbafda4a838665293a3514e381;alias
Call-ID: 2629e8da9c1d45119fe6f39c72944907
From: "37200" <sip:37200@192.168.133.5>;tag=e08d4dea55904f0db4f89b0f87424339
To: <sip:37100@192.168.133.5>
CSeq: 13660 INVITE
Server: Asterisk PBX 18.0.0-rc2
Content-Length:  0


    -- Executing [37100@phones:1] NoOp("PJSIP/37200-00000026", "37100") in new stack
    -- Executing [37100@phones:2] Dial("PJSIP/37200-00000026", "PJSIP/37100") in new stack
    -- Called PJSIP/37100
<--- Transmitting SIP request (999 bytes) to TLS:192.168.133.85:50097 --->
INVITE sip:69571804@192.168.133.85:50097;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.133.5:5061;rport;branch=z9hG4bKPj21351e7f-0190-4994-b5bb-7c945e0cce24;alias
From: "37200" <sip:37200@192.168.133.5>;tag=0695cc9d-ba57-43a4-8fbd-833b31bc9a9d
To: <sip:69571804@192.168.133.85>
Contact: <sip:asterisk@192.168.133.5:5061;transport=TLS>
Call-ID: 40a8b9f3-fbbe-43ca-8348-2d070a3b0e92
CSeq: 31918 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
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:   286

v=0
o=- 2007370001 2007370001 IN IP4 192.168.133.5
s=Asterisk
c=IN IP4 192.168.133.5
t=0 0
m=audio 17678 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [37100@phones:3] Hangup("PJSIP/37200-00000026", "") in new stack
  == Spawn extension (phones, 37100, 3) exited non-zero on 'PJSIP/37200-00000026'
<--- Transmitting SIP response (455 bytes) to TLS:192.168.133.90:49984 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TLS 192.168.133.90:49984;rport=49984;received=192.168.133.90;branch=z9hG4bKPjecfa9edbafda4a838665293a3514e381;alias
Call-ID: 2629e8da9c1d45119fe6f39c72944907
From: "37200" <sip:37200@192.168.133.5>;tag=e08d4dea55904f0db4f89b0f87424339
To: <sip:37100@192.168.133.5>;tag=0aeb5efa-b8ed-4439-8991-24f73b2168d3
CSeq: 13660 INVITE
Server: Asterisk PBX 18.0.0-rc2
Reason: Q.850;cause=34
Content-Length:  0

this same scenario used to work months ago.
i just can t figure it out!!

Your logs appear to indicate that the endpoint never responds to the INVITE sent by Asterisk. It could be that something on your machine running the endpoint is blocking the TCP port 50097 eg. local system firewall. You might want to look at more Asterisk logs as well to see what the connectivity history looks like for this endpoint, or if there is some sort of TLS cert expiration issue or an out-of-sync clock on the endpoint.

looks like it is a certificate issue!
it is so werid! the endpoints are registering but whenever i try making a call from one endpoint to another it fails.
from pjsip.conf i set verify server to NO, then the call was a success! but when it is set to yes! then it always says service unavailable!

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