search

InnovateAsterisk / Browser-Phone

A fully featured browser based WebRTC SIP phone for Asterisk
https://www.innovateasterisk.com
GNU Affero General Public License v3.0
499 stars 245 forks source link

Outgoing from WebRTC issue #441

Closed hayzamjs closed 1 year ago

hayzamjs commented 1 year ago

Hello,

I've setup 2 extensions in pjsip as follows:

;==========General Settings==========
[general]

;==========UDP Transport==========
[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0:5060

;==========WSS Transport==========
[transport-wss]
type = transport
protocol = wss
bind = 0.0.0.0

;==========Extensions Template==========
[endpoint-ext](!)
type=endpoint
context=sip_internal
webrtc=yes
disallow=all
;allow=alaw
;allow=ulaw
;allow=g729
;allow=gsm
allow=opus

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

[aor-single-reg](!)
type=aor
max_contacts=10

;==========Extension 1337==========
[1337](endpoint-ext)
auth=1337
aors=1337

[1337](auth-userpass)
username=1337
password=1337

[1337](aor-single-reg)

;==========Extension 1338==========
[1338](endpoint-ext)
auth=1338
aors=1338

[1338](auth-userpass)
username=1338
password=1338

[1338](aor-single-reg)

I am able to dial from a normal sip client (I'm using Zoiper(extension 1338)), to WebRTC (extension 1337) without any issues and I can hear audio both ways. But the issue is dialing from BrowserPhone to Zoiper, I don't see a ring in the zoiper client nor do I see anything in the asterisk CLI logs, I don't see even the ring event. Any pointers on how this could be solved?

My end goal is to have all the extensions be able to call each other seemlessly

EDIT: edited the conf

InnovateAsterisk commented 1 year ago

[transport-wss] type = transport protocol = wss bind = 0.0.0.0:5060 <—-

I don’t think you bind it like this. You can probably take the bind line out. The port is opened and controlled with http.conf

InnovateAsterisk commented 1 year ago

“ nor do I see anything in the asterisk CLI logs”

If there is no activity in the CLI when you make a call from webrtc extension, then there must be some kind of error shown.

First put pjsip logging on so that you can see the sip messages. Remember Asterisk is not a proxy, so all the sip messages are terminated on asterisk, this means that if the messages doesn’t appear in the CLI, there is another problem. Asterisk will have no choice but to handle the sip messages, meaning either dial plan will execute or you will get an error.

hayzamjs commented 1 year ago

I've updated the pjsip.conf and removed the port, without the 0.0.0.0 in bind I get this error:

[Feb 21 23:33:05] ERROR[17497]: res_pjsip/config_transport.c:711 transport_apply: Transport 'transport-wss' could not be started as binding not specified

None the less calling from zoiper to BrowserPhone results in the following pjsip log:

DifOS*CLI>
    -- [difuse_lte_trunk] Trying to connect on /dev/ttyUSB2...
    -- [difuse_lte_trunk] Quectel has connected, initializing...
[Feb 21 23:30:54] ERROR[17249]: at_response.c:410 log_cmd_response_error: [difuse_lte_trunk] Getting IMSI number failed
    -- [difuse_lte_trunk] Error initializing Quectel
    -- [difuse_lte_trunk] Quectel has disconnected
<--- Received SIP request (969 bytes) from UDP:49.37.212.107:50816 --->
INVITE sip:1337@redacted-domain-name.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.29.4:50816;branch=z9hG4bK-524287-1---f2a2882c70c1a598;rport
Max-Forwards: 70
Contact: <sip:1338@49.37.212.107:50816;transport=UDP>
To: <sip:1337@redacted-domain-name.com>
From: <sip:1338@redacted-domain-name.com;transport=UDP>;tag=fb22d750
Call-ID: M9U5QZ-mJAYc6tQ1-UPIGg..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.5.10 v2.10.17.3
Allow-Events: presence, kpml, talk
Content-Length: 375

v=0
o=Z 122335893 1 IN IP4 49.37.212.107
s=Z
c=IN IP4 49.37.212.107
t=0 0
m=audio 57885 RTP/AVPF 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir

<--- Transmitting SIP response (528 bytes) to UDP:49.37.212.107:50816 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.29.4:50816;rport=50816;received=49.37.212.107;branch=z9hG4bK-524287-1---f2a2882c70c1a598
Call-ID: M9U5QZ-mJAYc6tQ1-UPIGg..
From: <sip:1338@redacted-domain-name.com>;tag=fb22d750
To: <sip:1337@redacted-domain-name.com>;tag=z9hG4bK-524287-1---f2a2882c70c1a598
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1677002455/3aae6128b1b968f936fa2cb1015e6d6b",opaque="4393276d3465c61d",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.0.1
Content-Length:  0

<--- Received SIP request (388 bytes) from UDP:49.37.212.107:50816 --->
ACK sip:1337@redacted-domain-name.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.29.4:50816;branch=z9hG4bK-524287-1---f2a2882c70c1a598;rport
Max-Forwards: 70
To: <sip:1337@redacted-domain-name.com>;tag=z9hG4bK-524287-1---f2a2882c70c1a598
From: <sip:1338@redacted-domain-name.com;transport=UDP>;tag=fb22d750
Call-ID: M9U5QZ-mJAYc6tQ1-UPIGg..
CSeq: 1 ACK
Content-Length: 0

<--- Received SIP request (1279 bytes) from UDP:49.37.212.107:50816 --->
INVITE sip:1337@redacted-domain-name.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.29.4:50816;branch=z9hG4bK-524287-1---3102073222c9a41c;rport
Max-Forwards: 70
Contact: <sip:1338@49.37.212.107:50816;transport=UDP>
To: <sip:1337@redacted-domain-name.com>
From: <sip:1338@redacted-domain-name.com;transport=UDP>;tag=fb22d750
Call-ID: M9U5QZ-mJAYc6tQ1-UPIGg..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.5.10 v2.10.17.3
Authorization: Digest username="1338",realm="asterisk",nonce="1677002455/3aae6128b1b968f936fa2cb1015e6d6b",uri="sip:1337@redacted-domain-name.com;transport=UDP",response="fc9a5a20b9063fef0984c15b92f6728a",cnonce="4f7c0915b1b5b3c3df82f8fcb40fb86b",nc=00000001,qop=auth,algorithm=MD5,opaque="4393276d3465c61d"
Allow-Events: presence, kpml, talk
Content-Length: 375

v=0
o=Z 122335893 1 IN IP4 49.37.212.107
s=Z
c=IN IP4 49.37.212.107
t=0 0
m=audio 57885 RTP/AVPF 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir

<--- Transmitting SIP response (336 bytes) to UDP:49.37.212.107:50816 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.29.4:50816;rport=50816;received=49.37.212.107;branch=z9hG4bK-524287-1---3102073222c9a41c
Call-ID: M9U5QZ-mJAYc6tQ1-UPIGg..
From: <sip:1338@redacted-domain-name.com>;tag=fb22d750
To: <sip:1337@redacted-domain-name.com>
CSeq: 2 INVITE
Server: Asterisk PBX 20.0.1
Content-Length:  0

    -- Executing [1337@sip_internal:1] Dial("PJSIP/1338-00000002", "PJSIP/1337") in new stack
    -- Called PJSIP/1337
<--- Transmitting SIP request (2459 bytes) to WSS:49.37.212.107:33586 --->
INVITE sip:qdmuvf6r@49.37.212.107:33586;transport=ws SIP/2.0
Via: SIP/2.0/WSS 45.x.x.x:8089;rport;branch=z9hG4bKPj7e989c91-70b7-490b-9f0e-6989b94fc1f0;alias
From: <sip:1338@DifOS>;tag=6facbffb-9109-4c61-b56b-386bd02fe364
To: <sip:qdmuvf6r@49.37.212.107>
Contact: <sip:asterisk@DifOS:5060;transport=ws>
Call-ID: 978e1fee-5243-4d7c-88fb-82725ed0c2d7
CSeq: 15815 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.0.1
Content-Type: application/sdp
Content-Length:  1768

v=0
o=- 1161121378 1161121378 IN IP4 45.x.x.x
s=Asterisk
c=IN IP4 45.x.x.x
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 15512 UDP/TLS/RTP/SAVPF 8 0 3 107 18 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 65:C5:56:DA:37:5B:B0:56:35:A5:86:C4:74:7F:C0:F7:A1:AC:BA:EC:BB:FD:AE:B9:6E:CD:2E:0F:36:CB:8E:03
a=ice-ufrag:11e4cde86d61b55242359e093a0019c6
a=ice-pwd:0eab063d5247a2d34951fac507ab8b9c
a=candidate:Hafef801 1 UDP 2130706431 10.254.248.1 15512 typ host
a=candidate:Ha010201 1 UDP 2130706431 10.1.2.1 15512 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 15512 typ host
a=candidate:H2d74e58d 1 UDP 2130706431 45.x.x.x 15512 typ host
a=candidate:Hafe650a 1 UDP 2130706431 10.254.101.10 15512 typ host
a=candidate:Haca000c 1 UDP 2130706431 10.202.0.12 15512 typ host
a=candidate:Ha010142 1 UDP 2130706431 10.1.1.66 15512 typ host
a=candidate:Had2000b 1 UDP 2130706431 10.210.0.11 15512 typ host
a=candidate:H5145ffc5 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0e 15512 typ host
a=candidate:Hcdc1d3e3 1 UDP 2130706431 fdda:71ba:e799::1 15512 typ host
a=candidate:H5145ffc6 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0f 15512 typ host
a=candidate:H2bf3c030 1 UDP 2130706431 fe80::800c:43ff:fe26:6000 15512 typ host
a=candidate:He38cf972 1 UDP 2130706431 fe80::20c:43ff:fe26:6000 15512 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:1840929981 cname:04fcc8ea-dcde-434d-84cb-6c295aa6c886
a=msid:d890ab76-7a29-4926-9cbc-f0f43b14a7cd 77684561-fa49-4661-9384-785ebd090c9d
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Received SIP response (463 bytes) from WSS:49.37.212.107:33586 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 45.x.x.x:8089;rport;branch=z9hG4bKPj7e989c91-70b7-490b-9f0e-6989b94fc1f0;alias
From: <sip:1338@DifOS>;tag=6facbffb-9109-4c61-b56b-386bd02fe364
To: <sip:qdmuvf6r@49.37.212.107>
CSeq: 15815 INVITE
Call-ID: 978e1fee-5243-4d7c-88fb-82725ed0c2d7
Supported: outbound
User-Agent: Browser Phone 0.3.22 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:109.0) Gecko/20100101 Firefox/110.0
Content-Length: 0

<--- Received SIP response (529 bytes) from WSS:49.37.212.107:33586 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 45.x.x.x:8089;rport;branch=z9hG4bKPj7e989c91-70b7-490b-9f0e-6989b94fc1f0;alias
From: <sip:1338@DifOS>;tag=6facbffb-9109-4c61-b56b-386bd02fe364
To: <sip:qdmuvf6r@49.37.212.107>;tag=eiocn9unbe
CSeq: 15815 INVITE
Call-ID: 978e1fee-5243-4d7c-88fb-82725ed0c2d7
Supported: outbound
User-Agent: Browser Phone 0.3.22 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:109.0) Gecko/20100101 Firefox/110.0
Contact: <sip:qdmuvf6r@192.0.2.35;transport=wss>
Content-Length: 0

    -- PJSIP/1337-00000003 is ringing
<--- Transmitting SIP response (525 bytes) to UDP:49.37.212.107:50816 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.29.4:50816;rport=50816;received=49.37.212.107;branch=z9hG4bK-524287-1---3102073222c9a41c
Call-ID: M9U5QZ-mJAYc6tQ1-UPIGg..
From: <sip:1338@redacted-domain-name.com>;tag=fb22d750
To: <sip:1337@redacted-domain-name.com>;tag=0d2ec97f-688f-46e2-b9b0-2487acb46b10
CSeq: 2 INVITE
Server: Asterisk PBX 20.0.1
Contact: <sip:45.x.x.x:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0

45.x.x.x is the PBX server

And calling from BrowserPhone to Zoiper yields this:

<--- Received SIP request (3885 bytes) from WSS:49.37.212.107:33678 --->
INVITE sip:1338@redacted-domain-name.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.35;branch=z9hG4bK4402786
To: <sip:1338@redacted-domain-name.com>
From: "1337" <sip:1337@redacted-domain-name.com>;tag=h80j799lml
CSeq: 1 INVITE
Call-ID: 5s85umevn8uhucadvnlf
Max-Forwards: 70
Contact: <sip:qdmuvf6r@192.0.2.35;transport=wss;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.22 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:109.0) Gecko/20100101 Firefox/110.0
Content-Type: application/sdp
Content-Length: 3285

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 5029958294352283709 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 47:AC:10:F4:65:6C:6D:E9:E1:6C:EA:08:0E:66:1A:B7:44:CB:52:70:5A:A7:87:28:E6:4C:9E:93:9D:ED:93:AE
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 63743 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 49.37.212.107
a=candidate:0 1 UDP 2122252543 172.22.240.1 63740 typ host
a=candidate:2 1 UDP 2122187007 2402:8100:3900:bf28:8d73:b54b:aada:5c08 63741 typ host
a=candidate:4 1 UDP 2122121471 fd8d:3f44:87b2:0:d875:dec8:ac63:e48a 63742 typ host
a=candidate:6 1 UDP 2121990399 192.168.1.142 63743 typ host
a=candidate:8 1 UDP 2122055935 192.168.56.1 63744 typ host
a=candidate:10 1 UDP 2121924863 192.168.191.1 63745 typ host
a=candidate:12 1 UDP 2121859327 192.168.150.1 63746 typ host
a=candidate:14 1 TCP 2105524479 172.22.240.1 9 typ host tcptype active
a=candidate:15 1 TCP 2105458943 2402:8100:3900:bf28:8d73:b54b:aada:5c08 9 typ host tcptype active
a=candidate:16 1 TCP 2105393407 fd8d:3f44:87b2:0:d875:dec8:ac63:e48a 9 typ host tcptype active
a=candidate:17 1 TCP 2105262335 192.168.1.142 9 typ host tcptype active
a=candidate:18 1 TCP 2105327871 192.168.56.1 9 typ host tcptype active
a=candidate:19 1 TCP 2105196799 192.168.191.1 9 typ host tcptype active
a=candidate:20 1 TCP 2105131263 192.168.150.1 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 172.22.240.1 63747 typ host
a=candidate:2 2 UDP 2122187006 2402:8100:3900:bf28:8d73:b54b:aada:5c08 63748 typ host
a=candidate:4 2 UDP 2122121470 fd8d:3f44:87b2:0:d875:dec8:ac63:e48a 63749 typ host
a=candidate:6 2 UDP 2121990398 192.168.1.142 63750 typ host
a=candidate:8 2 UDP 2122055934 192.168.56.1 63751 typ host
a=candidate:10 2 UDP 2121924862 192.168.191.1 63752 typ host
a=candidate:12 2 UDP 2121859326 192.168.150.1 63753 typ host
a=candidate:14 2 TCP 2105524478 172.22.240.1 9 typ host tcptype active
a=candidate:15 2 TCP 2105458942 2402:8100:3900:bf28:8d73:b54b:aada:5c08 9 typ host tcptype active
a=candidate:16 2 TCP 2105393406 fd8d:3f44:87b2:0:d875:dec8:ac63:e48a 9 typ host tcptype active
a=candidate:17 2 TCP 2105262334 192.168.1.142 9 typ host tcptype active
a=candidate:18 2 TCP 2105327870 192.168.56.1 9 typ host tcptype active
a=candidate:19 2 TCP 2105196798 192.168.191.1 9 typ host tcptype active
a=candidate:20 2 TCP 2105131262 192.168.150.1 9 typ host tcptype active
a=candidate:7 1 UDP 1685790719 49.37.212.107 63743 typ srflx raddr 192.168.1.142 rport 63743
a=candidate:7 2 UDP 1685790718 49.37.212.107 63750 typ srflx raddr 192.168.1.142 rport 63750
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:db572d09c82ad39dff2205e24b6eef0c
a=ice-ufrag:2be08586
a=mid:0
a=msid:{77fb5137-09eb-466e-a98d-0275f525694c} {93791b6b-cb53-4822-90f8-2625a5be7c41}
a=rtcp:63750 IN IP4 49.37.212.107
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:3076815183 cname:{13aa93d6-735c-49e9-88a2-965876e8f199}

<--- Transmitting SIP response (483 bytes) to WSS:49.37.212.107:33678 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.35;rport=33678;received=49.37.212.107;branch=z9hG4bK4402786
Call-ID: 5s85umevn8uhucadvnlf
From: "1337" <sip:1337@redacted-domain-name.com>;tag=h80j799lml
To: <sip:1338@redacted-domain-name.com>;tag=z9hG4bK4402786
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1677002690/48669c0e50d9c2de51534e277d035dc0",opaque="3717a8070d973dc0",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.0.1
Content-Length:  0

<--- Received SIP request (309 bytes) from WSS:49.37.212.107:33678 --->
ACK sip:1338@redacted-domain-name.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.35;branch=z9hG4bK4402786
To: <sip:1338@redacted-domain-name.com>;tag=z9hG4bK4402786
From: "1337" <sip:1337@redacted-domain-name.com>;tag=h80j799lml
Call-ID: 5s85umevn8uhucadvnlf
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

<--- Received SIP request (4170 bytes) from WSS:49.37.212.107:33678 --->
INVITE sip:1338@redacted-domain-name.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.35;branch=z9hG4bK7514954
To: <sip:1338@redacted-domain-name.com>
From: "1337" <sip:1337@redacted-domain-name.com>;tag=h80j799lml
CSeq: 2 INVITE
Call-ID: 5s85umevn8uhucadvnlf
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="1337", realm="asterisk", nonce="1677002690/48669c0e50d9c2de51534e277d035dc0", uri="sip:1338@redacted-domain-name.com", response="5abb94cf8896c48aa654403781ea29c6", opaque="3717a8070d973dc0", qop=auth, cnonce="r1bae217rn7u", nc=00000001
Contact: <sip:qdmuvf6r@192.0.2.35;transport=wss;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.22 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:109.0) Gecko/20100101 Firefox/110.0
Content-Type: application/sdp
Content-Length: 3285

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 5029958294352283709 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 47:AC:10:F4:65:6C:6D:E9:E1:6C:EA:08:0E:66:1A:B7:44:CB:52:70:5A:A7:87:28:E6:4C:9E:93:9D:ED:93:AE
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 63743 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 49.37.212.107
a=candidate:0 1 UDP 2122252543 172.22.240.1 63740 typ host
a=candidate:2 1 UDP 2122187007 2402:8100:3900:bf28:8d73:b54b:aada:5c08 63741 typ host
a=candidate:4 1 UDP 2122121471 fd8d:3f44:87b2:0:d875:dec8:ac63:e48a 63742 typ host
a=candidate:6 1 UDP 2121990399 192.168.1.142 63743 typ host
a=candidate:8 1 UDP 2122055935 192.168.56.1 63744 typ host
a=candidate:10 1 UDP 2121924863 192.168.191.1 63745 typ host
a=candidate:12 1 UDP 2121859327 192.168.150.1 63746 typ host
a=candidate:14 1 TCP 2105524479 172.22.240.1 9 typ host tcptype active
a=candidate:15 1 TCP 2105458943 2402:8100:3900:bf28:8d73:b54b:aada:5c08 9 typ host tcptype active
a=candidate:16 1 TCP 2105393407 fd8d:3f44:87b2:0:d875:dec8:ac63:e48a 9 typ host tcptype active
a=candidate:17 1 TCP 2105262335 192.168.1.142 9 typ host tcptype active
a=candidate:18 1 TCP 2105327871 192.168.56.1 9 typ host tcptype active
a=candidate:19 1 TCP 2105196799 192.168.191.1 9 typ host tcptype active
a=candidate:20 1 TCP 2105131263 192.168.150.1 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 172.22.240.1 63747 typ host
a=candidate:2 2 UDP 2122187006 2402:8100:3900:bf28:8d73:b54b:aada:5c08 63748 typ host
a=candidate:4 2 UDP 2122121470 fd8d:3f44:87b2:0:d875:dec8:ac63:e48a 63749 typ host
a=candidate:6 2 UDP 2121990398 192.168.1.142 63750 typ host
a=candidate:8 2 UDP 2122055934 192.168.56.1 63751 typ host
a=candidate:10 2 UDP 2121924862 192.168.191.1 63752 typ host
a=candidate:12 2 UDP 2121859326 192.168.150.1 63753 typ host
a=candidate:14 2 TCP 2105524478 172.22.240.1 9 typ host tcptype active
a=candidate:15 2 TCP 2105458942 2402:8100:3900:bf28:8d73:b54b:aada:5c08 9 typ host tcptype active
a=candidate:16 2 TCP 2105393406 fd8d:3f44:87b2:0:d875:dec8:ac63:e48a 9 typ host tcptype active
a=candidate:17 2 TCP 2105262334 192.168.1.142 9 typ host tcptype active
a=candidate:18 2 TCP 2105327870 192.168.56.1 9 typ host tcptype active
a=candidate:19 2 TCP 2105196798 192.168.191.1 9 typ host tcptype active
a=candidate:20 2 TCP 2105131262 192.168.150.1 9 typ host tcptype active
a=candidate:7 1 UDP 1685790719 49.37.212.107 63743 typ srflx raddr 192.168.1.142 rport 63743
a=candidate:7 2 UDP 1685790718 49.37.212.107 63750 typ srflx raddr 192.168.1.142 rport 63750
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:db572d09c82ad39dff2205e24b6eef0c
a=ice-ufrag:2be08586
a=mid:0
a=msid:{77fb5137-09eb-466e-a98d-0275f525694c} {93791b6b-cb53-4822-90f8-2625a5be7c41}
a=rtcp:63750 IN IP4 49.37.212.107
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:3076815183 cname:{13aa93d6-735c-49e9-88a2-965876e8f199}

<--- Transmitting SIP response (312 bytes) to WSS:49.37.212.107:33678 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 192.0.2.35;rport=33678;received=49.37.212.107;branch=z9hG4bK7514954
Call-ID: 5s85umevn8uhucadvnlf
From: "1337" <sip:1337@redacted-domain-name.com>;tag=h80j799lml
To: <sip:1338@redacted-domain-name.com>
CSeq: 2 INVITE
Server: Asterisk PBX 20.0.1
Content-Length:  0

    -- Executing [1338@sip_internal:1] Dial("PJSIP/1337-00000004", "PJSIP/1338") in new stack
    -- Called PJSIP/1338
<--- Transmitting SIP request (2448 bytes) to UDP:192.168.29.4:55000 --->
INVITE sip:1338@192.168.29.4:55000;ob SIP/2.0
Via: SIP/2.0/UDP 45.x.x.x:5060;rport;branch=z9hG4bKPj7fcdcbe5-d9f9-4285-9011-d622fb69cbdb
From: "1337" <sip:1337@45.x.x.x>;tag=c0cb38be-6786-4b54-a310-debbfac127b9
To: <sip:1338@192.168.29.4;ob>
Contact: <sip:asterisk@45.x.x.x:5060>
Call-ID: 90e1d8fa-e2f8-4805-bdbb-dc44e82139a2
CSeq: 16397 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.0.1
Content-Type: application/sdp
Content-Length:  1768

v=0
o=- 1814262825 1814262825 IN IP4 45.x.x.x
s=Asterisk
c=IN IP4 45.x.x.x
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 17062 UDP/TLS/RTP/SAVPF 8 0 107 18 3 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 21:58:7A:CF:37:90:BF:8D:C9:68:2E:6B:D0:4E:73:5F:CF:F2:8B:C6:B0:2E:39:D5:5D:EA:67:FB:57:85:E1:0D
a=ice-ufrag:78cf11a1645ce22949e232911fb21154
a=ice-pwd:393bf3d77dbe2cdc1af7fb1b456d014e
a=candidate:Hafef801 1 UDP 2130706431 10.254.248.1 17062 typ host
a=candidate:Ha010201 1 UDP 2130706431 10.1.2.1 17062 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 17062 typ host
a=candidate:H2d74e58d 1 UDP 2130706431 45.x.x.x 17062 typ host
a=candidate:Hafe650a 1 UDP 2130706431 10.254.101.10 17062 typ host
a=candidate:Haca000c 1 UDP 2130706431 10.202.0.12 17062 typ host
a=candidate:Ha010142 1 UDP 2130706431 10.1.1.66 17062 typ host
a=candidate:Had2000b 1 UDP 2130706431 10.210.0.11 17062 typ host
a=candidate:H5145ffc5 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0e 17062 typ host
a=candidate:Hcdc1d3e3 1 UDP 2130706431 fdda:71ba:e799::1 17062 typ host
a=candidate:H5145ffc6 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0f 17062 typ host
a=candidate:H2bf3c030 1 UDP 2130706431 fe80::800c:43ff:fe26:6000 17062 typ host
a=candidate:He38cf972 1 UDP 2130706431 fe80::20c:43ff:fe26:6000 17062 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:1107486638 cname:d129e21d-5d32-47fb-bd66-40b937d1b1eb
a=msid:d9112def-69c5-47c3-8975-341c98a77033 b4ddb4a4-7b3e-45b2-8ad3-09e96a2a11c7
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Transmitting SIP request (2448 bytes) to UDP:192.168.29.4:55000 --->
INVITE sip:1338@192.168.29.4:55000;ob SIP/2.0
Via: SIP/2.0/UDP 45.x.x.x:5060;rport;branch=z9hG4bKPj7fcdcbe5-d9f9-4285-9011-d622fb69cbdb
From: "1337" <sip:1337@45.x.x.x>;tag=c0cb38be-6786-4b54-a310-debbfac127b9
To: <sip:1338@192.168.29.4;ob>
Contact: <sip:asterisk@45.x.x.x:5060>
Call-ID: 90e1d8fa-e2f8-4805-bdbb-dc44e82139a2
CSeq: 16397 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.0.1
Content-Type: application/sdp
Content-Length:  1768

v=0
o=- 1814262825 1814262825 IN IP4 45.x.x.x
s=Asterisk
c=IN IP4 45.x.x.x
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 17062 UDP/TLS/RTP/SAVPF 8 0 107 18 3 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 21:58:7A:CF:37:90:BF:8D:C9:68:2E:6B:D0:4E:73:5F:CF:F2:8B:C6:B0:2E:39:D5:5D:EA:67:FB:57:85:E1:0D
a=ice-ufrag:78cf11a1645ce22949e232911fb21154
a=ice-pwd:393bf3d77dbe2cdc1af7fb1b456d014e
a=candidate:Hafef801 1 UDP 2130706431 10.254.248.1 17062 typ host
a=candidate:Ha010201 1 UDP 2130706431 10.1.2.1 17062 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 17062 typ host
a=candidate:H2d74e58d 1 UDP 2130706431 45.x.x.x 17062 typ host
a=candidate:Hafe650a 1 UDP 2130706431 10.254.101.10 17062 typ host
a=candidate:Haca000c 1 UDP 2130706431 10.202.0.12 17062 typ host
a=candidate:Ha010142 1 UDP 2130706431 10.1.1.66 17062 typ host
a=candidate:Had2000b 1 UDP 2130706431 10.210.0.11 17062 typ host
a=candidate:H5145ffc5 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0e 17062 typ host
a=candidate:Hcdc1d3e3 1 UDP 2130706431 fdda:71ba:e799::1 17062 typ host
a=candidate:H5145ffc6 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0f 17062 typ host
a=candidate:H2bf3c030 1 UDP 2130706431 fe80::800c:43ff:fe26:6000 17062 typ host
a=candidate:He38cf972 1 UDP 2130706431 fe80::20c:43ff:fe26:6000 17062 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:1107486638 cname:d129e21d-5d32-47fb-bd66-40b937d1b1eb
a=msid:d9112def-69c5-47c3-8975-341c98a77033 b4ddb4a4-7b3e-45b2-8ad3-09e96a2a11c7
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Transmitting SIP request (2448 bytes) to UDP:192.168.29.4:55000 --->
INVITE sip:1338@192.168.29.4:55000;ob SIP/2.0
Via: SIP/2.0/UDP 45.x.x.x:5060;rport;branch=z9hG4bKPj7fcdcbe5-d9f9-4285-9011-d622fb69cbdb
From: "1337" <sip:1337@45.x.x.x>;tag=c0cb38be-6786-4b54-a310-debbfac127b9
To: <sip:1338@192.168.29.4;ob>
Contact: <sip:asterisk@45.x.x.x:5060>
Call-ID: 90e1d8fa-e2f8-4805-bdbb-dc44e82139a2
CSeq: 16397 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.0.1
Content-Type: application/sdp
Content-Length:  1768

v=0
o=- 1814262825 1814262825 IN IP4 45.x.x.x
s=Asterisk
c=IN IP4 45.x.x.x
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 17062 UDP/TLS/RTP/SAVPF 8 0 107 18 3 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 21:58:7A:CF:37:90:BF:8D:C9:68:2E:6B:D0:4E:73:5F:CF:F2:8B:C6:B0:2E:39:D5:5D:EA:67:FB:57:85:E1:0D
a=ice-ufrag:78cf11a1645ce22949e232911fb21154
a=ice-pwd:393bf3d77dbe2cdc1af7fb1b456d014e
a=candidate:Hafef801 1 UDP 2130706431 10.254.248.1 17062 typ host
a=candidate:Ha010201 1 UDP 2130706431 10.1.2.1 17062 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 17062 typ host
a=candidate:H2d74e58d 1 UDP 2130706431 45.x.x.x 17062 typ host
a=candidate:Hafe650a 1 UDP 2130706431 10.254.101.10 17062 typ host
a=candidate:Haca000c 1 UDP 2130706431 10.202.0.12 17062 typ host
a=candidate:Ha010142 1 UDP 2130706431 10.1.1.66 17062 typ host
a=candidate:Had2000b 1 UDP 2130706431 10.210.0.11 17062 typ host
a=candidate:H5145ffc5 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0e 17062 typ host
a=candidate:Hcdc1d3e3 1 UDP 2130706431 fdda:71ba:e799::1 17062 typ host
a=candidate:H5145ffc6 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0f 17062 typ host
a=candidate:H2bf3c030 1 UDP 2130706431 fe80::800c:43ff:fe26:6000 17062 typ host
a=candidate:He38cf972 1 UDP 2130706431 fe80::20c:43ff:fe26:6000 17062 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:1107486638 cname:d129e21d-5d32-47fb-bd66-40b937d1b1eb
a=msid:d9112def-69c5-47c3-8975-341c98a77033 b4ddb4a4-7b3e-45b2-8ad3-09e96a2a11c7
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Transmitting SIP request (2448 bytes) to UDP:192.168.29.4:55000 --->
INVITE sip:1338@192.168.29.4:55000;ob SIP/2.0
Via: SIP/2.0/UDP 45.x.x.x:5060;rport;branch=z9hG4bKPj7fcdcbe5-d9f9-4285-9011-d622fb69cbdb
From: "1337" <sip:1337@45.x.x.x>;tag=c0cb38be-6786-4b54-a310-debbfac127b9
To: <sip:1338@192.168.29.4;ob>
Contact: <sip:asterisk@45.x.x.x:5060>
Call-ID: 90e1d8fa-e2f8-4805-bdbb-dc44e82139a2
CSeq: 16397 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.0.1
Content-Type: application/sdp
Content-Length:  1768

v=0
o=- 1814262825 1814262825 IN IP4 45.x.x.x
s=Asterisk
c=IN IP4 45.x.x.x
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 17062 UDP/TLS/RTP/SAVPF 8 0 107 18 3 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 21:58:7A:CF:37:90:BF:8D:C9:68:2E:6B:D0:4E:73:5F:CF:F2:8B:C6:B0:2E:39:D5:5D:EA:67:FB:57:85:E1:0D
a=ice-ufrag:78cf11a1645ce22949e232911fb21154
a=ice-pwd:393bf3d77dbe2cdc1af7fb1b456d014e
a=candidate:Hafef801 1 UDP 2130706431 10.254.248.1 17062 typ host
a=candidate:Ha010201 1 UDP 2130706431 10.1.2.1 17062 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 17062 typ host
a=candidate:H2d74e58d 1 UDP 2130706431 45.x.x.x 17062 typ host
a=candidate:Hafe650a 1 UDP 2130706431 10.254.101.10 17062 typ host
a=candidate:Haca000c 1 UDP 2130706431 10.202.0.12 17062 typ host
a=candidate:Ha010142 1 UDP 2130706431 10.1.1.66 17062 typ host
a=candidate:Had2000b 1 UDP 2130706431 10.210.0.11 17062 typ host
a=candidate:H5145ffc5 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0e 17062 typ host
a=candidate:Hcdc1d3e3 1 UDP 2130706431 fdda:71ba:e799::1 17062 typ host
a=candidate:H5145ffc6 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0f 17062 typ host
a=candidate:H2bf3c030 1 UDP 2130706431 fe80::800c:43ff:fe26:6000 17062 typ host
a=candidate:He38cf972 1 UDP 2130706431 fe80::20c:43ff:fe26:6000 17062 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:1107486638 cname:d129e21d-5d32-47fb-bd66-40b937d1b1eb
a=msid:d9112def-69c5-47c3-8975-341c98a77033 b4ddb4a4-7b3e-45b2-8ad3-09e96a2a11c7
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Transmitting SIP request (2446 bytes) to UDP:192.168.29.4:55000 --->
INVITE sip:1338@192.168.29.4:55000;ob SIP/2.0
Via: SIP/2.0/UDP 45.x.x.x:5060;rport;branch=z9hG4bKPjf1d8d597-0eb8-4c05-b054-ed5ed9e97c55
From: "1337" <sip:1337@45.x.x.x>;tag=50d92bce-0ad3-4e5e-a951-1cdc5f8a6696
To: <sip:1338@192.168.29.4;ob>
Contact: <sip:asterisk@45.x.x.x:5060>
Call-ID: 91936cc1-00e0-4445-b445-8b553a6448db
CSeq: 4510 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.0.1
Content-Type: application/sdp
Content-Length:  1767

v=0
o=- 2036252927 2036252927 IN IP4 45.x.x.x
s=Asterisk
c=IN IP4 45.x.x.x
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 18162 UDP/TLS/RTP/SAVPF 8 0 107 18 3 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 01:C2:88:61:6B:FD:5B:FB:E0:91:71:A0:43:D6:04:76:BB:5B:C9:AD:A0:77:05:A3:82:83:73:26:90:7F:0F:99
a=ice-ufrag:416415ed26225f807cdf298855a88759
a=ice-pwd:1fe4cf602de1cb3e664f669f385dd3f1
a=candidate:Hafef801 1 UDP 2130706431 10.254.248.1 18162 typ host
a=candidate:Ha010201 1 UDP 2130706431 10.1.2.1 18162 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 18162 typ host
a=candidate:H2d74e58d 1 UDP 2130706431 45.x.x.x 18162 typ host
a=candidate:Hafe650a 1 UDP 2130706431 10.254.101.10 18162 typ host
a=candidate:Haca000c 1 UDP 2130706431 10.202.0.12 18162 typ host
a=candidate:Ha010142 1 UDP 2130706431 10.1.1.66 18162 typ host
a=candidate:Had2000b 1 UDP 2130706431 10.210.0.11 18162 typ host
a=candidate:H5145ffc5 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0e 18162 typ host
a=candidate:Hcdc1d3e3 1 UDP 2130706431 fdda:71ba:e799::1 18162 typ host
a=candidate:H5145ffc6 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0f 18162 typ host
a=candidate:H2bf3c030 1 UDP 2130706431 fe80::800c:43ff:fe26:6000 18162 typ host
a=candidate:He38cf972 1 UDP 2130706431 fe80::20c:43ff:fe26:6000 18162 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:678212404 cname:6d6c31fc-bb9d-457f-aaf3-498aef15ccc4
a=msid:327f8d0a-6b5c-4635-b98b-69b2d4ff73a3 3242fc28-b392-45f1-beee-e2b26ba7f075
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Received SIP request (449 bytes) from WSS:49.37.212.107:33678 --->
CANCEL sip:1338@redacted-domain-name.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.35;branch=z9hG4bK7514954
To: <sip:1338@redacted-domain-name.com>
From: "1337" <sip:1337@redacted-domain-name.com>;tag=h80j799lml
CSeq: 2 CANCEL
Call-ID: 5s85umevn8uhucadvnlf
Max-Forwards: 70
Supported: outbound
User-Agent: Browser Phone 0.3.22 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:109.0) Gecko/20100101 Firefox/110.0
Content-Length: 0

<--- Transmitting SIP response (349 bytes) to WSS:49.37.212.107:33678 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.35;rport=33678;received=49.37.212.107;branch=z9hG4bK7514954
Call-ID: 5s85umevn8uhucadvnlf
From: "1337" <sip:1337@redacted-domain-name.com>;tag=h80j799lml
To: <sip:1338@redacted-domain-name.com>;tag=35511041-3979-4de1-8f77-2ba609672a03
CSeq: 2 CANCEL
Server: Asterisk PBX 20.0.1
Content-Length:  0

<--- Transmitting SIP response (365 bytes) to WSS:49.37.212.107:33678 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/WSS 192.0.2.35;rport=33678;received=49.37.212.107;branch=z9hG4bK7514954
Call-ID: 5s85umevn8uhucadvnlf
From: "1337" <sip:1337@redacted-domain-name.com>;tag=h80j799lml
To: <sip:1338@redacted-domain-name.com>;tag=35511041-3979-4de1-8f77-2ba609672a03
CSeq: 2 INVITE
Server: Asterisk PBX 20.0.1
Content-Length:  0

  == Spawn extension (sip_internal, 1338, 1) exited non-zero on 'PJSIP/1337-00000004'
<--- Received SIP request (331 bytes) from WSS:49.37.212.107:33678 --->
ACK sip:1338@redacted-domain-name.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.35;branch=z9hG4bK7514954
To: <sip:1338@redacted-domain-name.com>;tag=35511041-3979-4de1-8f77-2ba609672a03
From: "1337" <sip:1337@redacted-domain-name.com>;tag=h80j799lml
Call-ID: 5s85umevn8uhucadvnlf
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0

<--- Transmitting SIP request (2448 bytes) to UDP:192.168.29.4:55000 --->
INVITE sip:1338@192.168.29.4:55000;ob SIP/2.0
Via: SIP/2.0/UDP 45.x.x.x:5060;rport;branch=z9hG4bKPj7fcdcbe5-d9f9-4285-9011-d622fb69cbdb
From: "1337" <sip:1337@45.x.x.x>;tag=c0cb38be-6786-4b54-a310-debbfac127b9
To: <sip:1338@192.168.29.4;ob>
Contact: <sip:asterisk@45.x.x.x:5060>
Call-ID: 90e1d8fa-e2f8-4805-bdbb-dc44e82139a2
CSeq: 16397 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.0.1
Content-Type: application/sdp
Content-Length:  1768

v=0
o=- 1814262825 1814262825 IN IP4 45.x.x.x
s=Asterisk
c=IN IP4 45.x.x.x
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 17062 UDP/TLS/RTP/SAVPF 8 0 107 18 3 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 21:58:7A:CF:37:90:BF:8D:C9:68:2E:6B:D0:4E:73:5F:CF:F2:8B:C6:B0:2E:39:D5:5D:EA:67:FB:57:85:E1:0D
a=ice-ufrag:78cf11a1645ce22949e232911fb21154
a=ice-pwd:393bf3d77dbe2cdc1af7fb1b456d014e
a=candidate:Hafef801 1 UDP 2130706431 10.254.248.1 17062 typ host
a=candidate:Ha010201 1 UDP 2130706431 10.1.2.1 17062 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 17062 typ host
a=candidate:H2d74e58d 1 UDP 2130706431 45.x.x.x 17062 typ host
a=candidate:Hafe650a 1 UDP 2130706431 10.254.101.10 17062 typ host
a=candidate:Haca000c 1 UDP 2130706431 10.202.0.12 17062 typ host
a=candidate:Ha010142 1 UDP 2130706431 10.1.1.66 17062 typ host
a=candidate:Had2000b 1 UDP 2130706431 10.210.0.11 17062 typ host
a=candidate:H5145ffc5 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0e 17062 typ host
a=candidate:Hcdc1d3e3 1 UDP 2130706431 fdda:71ba:e799::1 17062 typ host
a=candidate:H5145ffc6 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0f 17062 typ host
a=candidate:H2bf3c030 1 UDP 2130706431 fe80::800c:43ff:fe26:6000 17062 typ host
a=candidate:He38cf972 1 UDP 2130706431 fe80::20c:43ff:fe26:6000 17062 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:1107486638 cname:d129e21d-5d32-47fb-bd66-40b937d1b1eb
a=msid:d9112def-69c5-47c3-8975-341c98a77033 b4ddb4a4-7b3e-45b2-8ad3-09e96a2a11c7
a=rtcp-fb:* transport-cc
a=mid:audio-0

    -- [difuse_lte_trunk] Trying to connect on /dev/ttyUSB2...
    -- [difuse_lte_trunk] Quectel has connected, initializing...
[Feb 21 23:34:59] ERROR[17831]: at_response.c:410 log_cmd_response_error: [difuse_lte_trunk] Getting IMSI number failed
    -- [difuse_lte_trunk] Error initializing Quectel
    -- [difuse_lte_trunk] Quectel has disconnected
<--- Received SIP request (920 bytes) from UDP:49.37.212.107:50816 --->
REGISTER sip:redacted-domain-name.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.29.4:50816;branch=z9hG4bK-524287-1---9d1f181c396e9a0d;rport
Max-Forwards: 70
Contact: <sip:1338@49.37.212.107:50816;transport=UDP;rinstance=68fd78f6cf3cd9b6>
To: <sip:1338@redacted-domain-name.com;transport=UDP>
From: <sip:1338@redacted-domain-name.com;transport=UDP>;tag=fc0dad56
Call-ID: qiiclAXPtEYI-aQl2GtHLQ..
CSeq: 11 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.5.10 v2.10.17.3
Authorization: Digest username="1338",realm="asterisk",nonce="1677002647/9ad65505ba7bd80102a583a9c7201fd0",uri="sip:redacted-domain-name.com;transport=UDP",response="e968b124273f726eb7a18de020e29920",cnonce="4e4c82fa2ae28846e391c3f5fb03853c",nc=00000002,qop=auth,algorithm=MD5,opaque="0155d69b1391dcda"
Allow-Events: presence, kpml, talk
Content-Length: 0

<--- Transmitting SIP response (542 bytes) to UDP:49.37.212.107:50816 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.29.4:50816;rport=50816;received=49.37.212.107;branch=z9hG4bK-524287-1---9d1f181c396e9a0d
Call-ID: qiiclAXPtEYI-aQl2GtHLQ..
From: <sip:1338@redacted-domain-name.com>;tag=fc0dad56
To: <sip:1338@redacted-domain-name.com>;tag=z9hG4bK-524287-1---9d1f181c396e9a0d
CSeq: 11 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1677002700/e1a14fc8041c0899afa5ddfdda1f3c0a",opaque="3d96995911223f7f",stale=true,algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.0.1
Content-Length:  0

<--- Received SIP request (920 bytes) from UDP:49.37.212.107:50816 --->
REGISTER sip:redacted-domain-name.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.29.4:50816;branch=z9hG4bK-524287-1---fc49c4ba12b87582;rport
Max-Forwards: 70
Contact: <sip:1338@49.37.212.107:50816;transport=UDP;rinstance=68fd78f6cf3cd9b6>
To: <sip:1338@redacted-domain-name.com;transport=UDP>
From: <sip:1338@redacted-domain-name.com;transport=UDP>;tag=fc0dad56
Call-ID: qiiclAXPtEYI-aQl2GtHLQ..
CSeq: 12 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.5.10 v2.10.17.3
Authorization: Digest username="1338",realm="asterisk",nonce="1677002700/e1a14fc8041c0899afa5ddfdda1f3c0a",uri="sip:redacted-domain-name.com;transport=UDP",response="7561c54484e971dce41a0d70da052c1d",cnonce="0f82402f93262b4f18c143cc9696cbc9",nc=00000001,qop=auth,algorithm=MD5,opaque="3d96995911223f7f"
Allow-Events: presence, kpml, talk
Content-Length: 0

<--- Transmitting SIP response (709 bytes) to UDP:49.37.212.107:50816 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.29.4:50816;rport=50816;received=49.37.212.107;branch=z9hG4bK-524287-1---fc49c4ba12b87582
Call-ID: qiiclAXPtEYI-aQl2GtHLQ..
From: <sip:1338@redacted-domain-name.com>;tag=fc0dad56
To: <sip:1338@redacted-domain-name.com>;tag=z9hG4bK-524287-1---fc49c4ba12b87582
CSeq: 12 REGISTER
Date: Tue, 21 Feb 2023 18:05:01 GMT
Contact: <sip:1338@192.168.29.4:55000;ob>;expires=212
Contact: <sip:1338@49.37.212.107:33417;transport=tls>;expires=2813
Contact: <sip:1338@49.37.212.107:33092;transport=tls>;expires=1748
Contact: <sip:1338@49.37.212.107:50816;transport=UDP;rinstance=68fd78f6cf3cd9b6>;expires=59
Expires: 60
Server: Asterisk PBX 20.0.1
Content-Length:  0

<--- Transmitting SIP request (2445 bytes) to UDP:192.168.29.4:55000 --->
INVITE sip:1338@192.168.29.4:55000;ob SIP/2.0
Via: SIP/2.0/UDP 45.x.x.x:5060;rport;branch=z9hG4bKPjbcd2dbed-6603-4784-baa1-5a6f077c835a
From: "1337" <sip:1337@45.x.x.x>;tag=c0fc635c-9d04-46da-b68b-3d10578807c7
To: <sip:1338@192.168.29.4;ob>
Contact: <sip:asterisk@45.x.x.x:5060>
Call-ID: d8df02f2-ad08-4f31-9088-f6b3342947ef
CSeq: 8060 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.0.1
Content-Type: application/sdp
Content-Length:  1766

v=0
o=- 933388450 933388450 IN IP4 45.x.x.x
s=Asterisk
c=IN IP4 45.x.x.x
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 16782 UDP/TLS/RTP/SAVPF 8 0 107 18 3 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 6B:DD:7F:19:6A:5B:D4:AF:F0:AE:CA:D2:DB:68:BF:3A:ED:B8:97:2A:DA:0A:28:BE:AC:73:E2:51:29:01:0C:B0
a=ice-ufrag:2295c12e186e20de30b8ba62321c348a
a=ice-pwd:5c3210b05cbf5e9326878d7f4a9cc62b
a=candidate:Hafef801 1 UDP 2130706431 10.254.248.1 16782 typ host
a=candidate:Ha010201 1 UDP 2130706431 10.1.2.1 16782 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 16782 typ host
a=candidate:H2d74e58d 1 UDP 2130706431 45.x.x.x 16782 typ host
a=candidate:Hafe650a 1 UDP 2130706431 10.254.101.10 16782 typ host
a=candidate:Haca000c 1 UDP 2130706431 10.202.0.12 16782 typ host
a=candidate:Ha010142 1 UDP 2130706431 10.1.1.66 16782 typ host
a=candidate:Had2000b 1 UDP 2130706431 10.210.0.11 16782 typ host
a=candidate:H5145ffc5 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0e 16782 typ host
a=candidate:Hcdc1d3e3 1 UDP 2130706431 fdda:71ba:e799::1 16782 typ host
a=candidate:H5145ffc6 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0f 16782 typ host
a=candidate:H2bf3c030 1 UDP 2130706431 fe80::800c:43ff:fe26:6000 16782 typ host
a=candidate:He38cf972 1 UDP 2130706431 fe80::20c:43ff:fe26:6000 16782 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:1204905885 cname:f2120f50-e650-43e8-96b9-6de04054d384
a=msid:dd9dcb23-1447-40c7-a38a-2c79f10dd971 a13c5038-c146-45e4-9607-489feabe7d17
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Transmitting SIP request (2448 bytes) to UDP:192.168.29.4:55000 --->
INVITE sip:1338@192.168.29.4:55000;ob SIP/2.0
Via: SIP/2.0/UDP 45.x.x.x:5060;rport;branch=z9hG4bKPj7fcdcbe5-d9f9-4285-9011-d622fb69cbdb
From: "1337" <sip:1337@45.x.x.x>;tag=c0cb38be-6786-4b54-a310-debbfac127b9
To: <sip:1338@192.168.29.4;ob>
Contact: <sip:asterisk@45.x.x.x:5060>
Call-ID: 90e1d8fa-e2f8-4805-bdbb-dc44e82139a2
CSeq: 16397 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.0.1
Content-Type: application/sdp
Content-Length:  1768

v=0
o=- 1814262825 1814262825 IN IP4 45.x.x.x
s=Asterisk
c=IN IP4 45.x.x.x
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 17062 UDP/TLS/RTP/SAVPF 8 0 107 18 3 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 21:58:7A:CF:37:90:BF:8D:C9:68:2E:6B:D0:4E:73:5F:CF:F2:8B:C6:B0:2E:39:D5:5D:EA:67:FB:57:85:E1:0D
a=ice-ufrag:78cf11a1645ce22949e232911fb21154
a=ice-pwd:393bf3d77dbe2cdc1af7fb1b456d014e
a=candidate:Hafef801 1 UDP 2130706431 10.254.248.1 17062 typ host
a=candidate:Ha010201 1 UDP 2130706431 10.1.2.1 17062 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 17062 typ host
a=candidate:H2d74e58d 1 UDP 2130706431 45.x.x.x 17062 typ host
a=candidate:Hafe650a 1 UDP 2130706431 10.254.101.10 17062 typ host
a=candidate:Haca000c 1 UDP 2130706431 10.202.0.12 17062 typ host
a=candidate:Ha010142 1 UDP 2130706431 10.1.1.66 17062 typ host
a=candidate:Had2000b 1 UDP 2130706431 10.210.0.11 17062 typ host
a=candidate:H5145ffc5 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0e 17062 typ host
a=candidate:Hcdc1d3e3 1 UDP 2130706431 fdda:71ba:e799::1 17062 typ host
a=candidate:H5145ffc6 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0f 17062 typ host
a=candidate:H2bf3c030 1 UDP 2130706431 fe80::800c:43ff:fe26:6000 17062 typ host
a=candidate:He38cf972 1 UDP 2130706431 fe80::20c:43ff:fe26:6000 17062 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:1107486638 cname:d129e21d-5d32-47fb-bd66-40b937d1b1eb
a=msid:d9112def-69c5-47c3-8975-341c98a77033 b4ddb4a4-7b3e-45b2-8ad3-09e96a2a11c7
a=rtcp-fb:* transport-cc
a=mid:audio-0

    -- [difuse_lte_trunk] Trying to connect on /dev/ttyUSB2...
    -- [difuse_lte_trunk] Quectel has connected, initializing...
[Feb 21 23:35:09] ERROR[17837]: at_response.c:410 log_cmd_response_error: [difuse_lte_trunk] Getting IMSI number failed
    -- [difuse_lte_trunk] Error initializing Quectel
    -- [difuse_lte_trunk] Quectel has disconnected
<--- Transmitting SIP request (2446 bytes) to UDP:192.168.29.4:55000 --->
INVITE sip:1338@192.168.29.4:55000;ob SIP/2.0
Via: SIP/2.0/UDP 45.x.x.x:5060;rport;branch=z9hG4bKPjf1d8d597-0eb8-4c05-b054-ed5ed9e97c55
From: "1337" <sip:1337@45.x.x.x>;tag=50d92bce-0ad3-4e5e-a951-1cdc5f8a6696
To: <sip:1338@192.168.29.4;ob>
Contact: <sip:asterisk@45.x.x.x:5060>
Call-ID: 91936cc1-00e0-4445-b445-8b553a6448db
CSeq: 4510 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.0.1
Content-Type: application/sdp
Content-Length:  1767

v=0
o=- 2036252927 2036252927 IN IP4 45.x.x.x
s=Asterisk
c=IN IP4 45.x.x.x
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 18162 UDP/TLS/RTP/SAVPF 8 0 107 18 3 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 01:C2:88:61:6B:FD:5B:FB:E0:91:71:A0:43:D6:04:76:BB:5B:C9:AD:A0:77:05:A3:82:83:73:26:90:7F:0F:99
a=ice-ufrag:416415ed26225f807cdf298855a88759
a=ice-pwd:1fe4cf602de1cb3e664f669f385dd3f1
a=candidate:Hafef801 1 UDP 2130706431 10.254.248.1 18162 typ host
a=candidate:Ha010201 1 UDP 2130706431 10.1.2.1 18162 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 18162 typ host
a=candidate:H2d74e58d 1 UDP 2130706431 45.x.x.x 18162 typ host
a=candidate:Hafe650a 1 UDP 2130706431 10.254.101.10 18162 typ host
a=candidate:Haca000c 1 UDP 2130706431 10.202.0.12 18162 typ host
a=candidate:Ha010142 1 UDP 2130706431 10.1.1.66 18162 typ host
a=candidate:Had2000b 1 UDP 2130706431 10.210.0.11 18162 typ host
a=candidate:H5145ffc5 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0e 18162 typ host
a=candidate:Hcdc1d3e3 1 UDP 2130706431 fdda:71ba:e799::1 18162 typ host
a=candidate:H5145ffc6 1 UDP 2130706431 fe80::e810:3fff:fe06:3f0f 18162 typ host
a=candidate:H2bf3c030 1 UDP 2130706431 fe80::800c:43ff:fe26:6000 18162 typ host
a=candidate:He38cf972 1 UDP 2130706431 fe80::20c:43ff:fe26:6000 18162 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:678212404 cname:6d6c31fc-bb9d-457f-aaf3-498aef15ccc4
a=msid:327f8d0a-6b5c-4635-b98b-69b2d4ff73a3 3242fc28-b392-45f1-beee-e2b26ba7f075
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Received SIP request (531 bytes) from UDP:10.254.254.2:5060 --->
OPTIONS sip:10.254.248.1 SIP/2.0
Via: SIP/2.0/UDP 10.254.254.2:5060;branch=z9hG4bK5a791c9d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.254.254.2>;tag=as56ef7c24
To: <sip:10.254.248.1>
Contact: <sip:asterisk@10.254.254.2:5060>
Call-ID: 3b402342708c304d6ccdb0b37ed0db1e@10.254.254.2:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.15.1
Date: Tue, 21 Feb 2023 18:05:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

[Feb 21 23:35:12] NOTICE[17752]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'OPTIONS' from '"asterisk" <sip:asterisk@10.254.254.2>' failed for '10.254.254.2:5060' (callid: 3b402342708c304d6ccdb0b37ed0db1e@10.254.254.2:5060) - No matching endpoint found
<--- Transmitting SIP response (500 bytes) to UDP:10.254.254.2:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.254.254.2:5060;rport=5060;received=10.254.254.2;branch=z9hG4bK5a791c9d
Call-ID: 3b402342708c304d6ccdb0b37ed0db1e@10.254.254.2:5060
From: "asterisk" <sip:asterisk@10.254.254.2>;tag=as56ef7c24
To: <sip:10.254.248.1>;tag=z9hG4bK5a791c9d
CSeq: 102 OPTIONS
WWW-Authenticate: Digest realm="asterisk",nonce="1677002712/f2fb04d4b407e768e28b75611756464d",opaque="0448deaa5ee6ae12",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.0.1
Content-Length:

45.x.x.x is my PBX server and 49.37.212.107 is my PC with the webrtc client

I'm guessing it reaches asterisk but then fails to go forward from there

EDIT: You can ignore the quectel error messages those are from my LTE dongle

InnovateAsterisk commented 1 year ago

Yes, i see even in my own config there is a bind= line. But still the message flow comes over the http server. https://github.com/InnovateAsterisk/Browser-Phone/blob/12061962af0ba09899c432cdc1e18a2f2297740e/config/pjsip.conf#L16

So from the trace above you can see that the call gets from the browser phone to asterisk. (so as a sanity test you could get it to play music on hold). But, the problem is coming from the Asterisk box, reaching Zoiper.

Take not of the multiple attempts to reach the zoiper device:

<--- Transmitting SIP request (2446 bytes) to UDP:192.168.29.4:55000 --->
INVITE sip:1338@192.168.29.4:55000;ob SIP/2.0

This means that Asterisk thinks extension 1338 is registered at a device with IP address 192.168.29.4. I'm not sure if this is correct. You may need to be re-writing your contact if you are setup with nat.

InnovateAsterisk commented 1 year ago

If you are going to be using a "regular" extension, it would probably be better to make a template for no-webrtc, with all the correct settings for say Zoiper, and then assigned that template to the Zoiper extension.

hayzamjs commented 1 year ago

My config now (with zoiper as the endpoint-ext-normal) given below and it works well! I was being dumb and using my double natted router to set both the UDP client and the webrtc client up, once I switched over the UDP client to my phone over LTE it worked with the latest config.

;==========UDP Transport==========
[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0:5060

;==========WSS Transport==========
[transport-wss]
type = transport
protocol = wss
bind=0.0.0.0

;==========Extensions Template==========
[endpoint-ext-webrtc](!)
type=endpoint
context=sip_internal
webrtc=yes
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=opus

[endpoint-ext-normal](!)
type=endpoint
context=sip_internal
webrtc=yes
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=opus

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

[aor-single-reg](!)
type=aor
max_contacts=10

;==========Extension 1337==========
[1337](endpoint-ext-webrtc)
auth=1337
aors=1337

[1337](auth-userpass)
username=1337
password=1337

[1337](aor-single-reg)

;==========Extension 1338==========
[1338](endpoint-ext-normal)
auth=1338
aors=1338

[1338](auth-userpass)
username=1338
password=1338

[1338](aor-single-reg)

My only issue that remains now is the fact that I can't let users interchangeably use their extension from a normal sip client as well as a WebRTC sip client, is there something I can do for that?

Thank you for all the responses so far!

InnovateAsterisk commented 1 year ago

My only issue that remains now is the fact that I can't let users interchangeably use their extension from a normal sip client as well as a WebRTC sip client, is there something I can do for that?

There basically isn't anything you can do about this. They may be able to register, but the calling will fail. DTLS isn't very well supported by typical UDP clients, and Unencrypted RTP isn't supported in WebRTC.

The only solution for this is to have a proxy in front of the Asterisk box, that strips the ICE and DTLS to make it a "normal" UDP call, but this requires a considerable configuration change.