Call ends directly after answering the call

the call directly ends after answering(holding the handset of the analog phone) an extension which is connected to my phone port at my pstn gateway (asm200).

i got an asterisk server with multiple endpoints(sip phones) that are working fine, recently i connected my pstn gateway with an analog phone connected to its phone/fxs port.
i created a trunk and an endpoint for this analog phone and then i configured my asm200.
my gateway and asterisk server have static ips.

at pjsip.conf

[gateway] ;; this is my trunk to the sip/pstn gateway
type=aor
contact=sip:xxx.xxx.xxx.110:5060

[gateway]
type=endpoint
context=phones
allow=all
aors=gateway

[gateway]
;type=identify
match=xxx.xxx.xxx.110
endpoint=gateway

[37400] ;; this is my extension to the analog phone
type = endpoint
context = phones
disallow = all
allow = ulaw,alaw,gsm
aors = 37400
auth = auth37400
device_state_busy_at=1

[37400]
type = aor
max_contacts = 1

[auth37400]
type=auth
auth_type=userpass
password=123
username=37400

so now i got the analog phone registered and shown as an endpoint but the weird thing is that it tries to register in 2 different accounts but i dont think it is a problem.
anyway i remove type=identify since they are both trusted with static ips.

when i make the call and answer it this is what i got where it ends directly :
– Executing [5537400@phones:2] Dial(β€œPJSIP/37200-000000a6”, β€œPJSIP/37400@gateway,25”) in new stack
– Called PJSIP/37400@gateway
– PJSIP/gateway-000000a7 is making progress passing it to PJSIP/37200-000000a6
– PJSIP/gateway-000000a7 is making progress passing it to PJSIP/37200-000000a6
– PJSIP/gateway-000000a7 answered PJSIP/37200-000000a6
– Channel PJSIP/gateway-000000a7 joined β€˜simple_bridge’ basic-bridge <2f9828d8-a284-417c-8cf4-28b676df764c>
– Channel PJSIP/37200-000000a6 joined β€˜simple_bridge’ basic-bridge <2f9828d8-a284-417c-8cf4-28b676df764c>
– Channel PJSIP/gateway-000000a7 left β€˜native_rtp’ basic-bridge <2f9828d8-a284-417c-8cf4-28b676df764c>
– Channel PJSIP/37200-000000a6 left β€˜native_rtp’ basic-bridge <2f9828d8-a284-417c-8cf4-28b676df764c>

is this your system ?
Asterisk <β€”> PSTN Gateway <β€”> Analog Phone(37400)
if it is right, you don’t need to register analog phone.

and could you append sip log?
at CLI, type command β€œpjsip set logger on”

1 Like

@hsunryou yes this is what it was.
but i now i tested my endpoint 44200 <β€”> asterisk1192.168.133.222 <β€”>asterisk2 192.168.133.111<β€”>endpoint 37301
which was working so fine when the ip of asterisk 1 was 192.168.137.222!!!
but when i modified the ip of asterisk2 and changed it in all configurations
i try to make the call and it gives this error which i cant understand ! (call ends when i answer)
which is the same error of Asterisk <β€”> PSTN Gateway <β€”> Analog Phone(37400)

now this is the CLI of 37200 <β€”> asterisk1192.168.133.111 <β€”>asterisk2 192.168.133.222<β€”>endpoint 44200
with the trunk called asterisk2
INVITE sip:0744200@192.168.133.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.2:5060;branch=z9hG4bK.gkNex-Ete;rport
From: sip:37301@192.168.133.111;tag=YS2WnVOO8
To: sip:0744200@192.168.133.111
CSeq: 20 INVITE
Call-ID: SUgERujZGF
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 498
Contact: sip:37301@192.168.133.2;transport=udp;expires=599;+sip.instance=β€œurn:uuid:2c74159b-a9ce-00d3-8921-0ee561834c37”
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)

v=0
o=37301 3132 2316 IN IP4 192.168.133.2
s=Talk
c=IN IP4 192.168.133.2
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 3 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<β€” Transmitting SIP response (477 bytes) to UDP:192.168.133.2:5060 β€”>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.133.2:5060;rport=5060;received=192.168.133.2;branch=z9hG4bK.gkNex-Ete
Call-ID: SUgERujZGF
From: sip:37301@192.168.133.111;tag=YS2WnVOO8
To: sip:0744200@192.168.133.111;tag=z9hG4bK.gkNex-Ete
CSeq: 20 INVITE
WWW-Authenticate: Digest realm=β€œasterisk”,nonce=β€œ1582798429/9269dbf5a18714bb0dc6ed40da73c276”,opaque=β€œ17c62c06790f2e60”,algorithm=md5,qop=β€œauth”
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP request (396 bytes) from UDP:192.168.133.2:5060 β€”>
ACK sip:0744200@192.168.133.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.2:5060;branch=z9hG4bK.gkNex-Ete;rport
Call-ID: SUgERujZGF
From: sip:37301@192.168.133.111;tag=YS2WnVOO8
To: sip:0744200@192.168.133.111;tag=z9hG4bK.gkNex-Ete
Contact: sip:37301@192.168.133.2;transport=udp;expires=599;+sip.instance=β€œurn:uuid:2c74159b-a9ce-00d3-8921-0ee561834c37”
Max-Forwards: 70
CSeq: 20 ACK

<β€” Received SIP request (1389 bytes) from UDP:192.168.133.2:5060 β€”>
INVITE sip:0744200@192.168.133.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.2:5060;branch=z9hG4bK.3XGa100cA;rport
From: sip:37301@192.168.133.111;tag=YS2WnVOO8
To: sip:0744200@192.168.133.111
CSeq: 21 INVITE
Call-ID: SUgERujZGF
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 498
Contact: sip:37301@192.168.133.2;transport=udp;expires=599;+sip.instance=β€œurn:uuid:2c74159b-a9ce-00d3-8921-0ee561834c37”
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582798429/9269dbf5a18714bb0dc6ed40da73c276”, algorithm=md5, opaque=β€œ17c62c06790f2e60”, username=β€œ37301”, uri="sip:0744200@192.168.133.111", response=β€œ66c44a8d5ccd2dcf2eda6e9240697bb3”, cnonce=β€œGNalh-PtMNk9AtmK”, nc=00000001, qop=auth

v=0
o=37301 3132 2316 IN IP4 192.168.133.2
s=Talk
c=IN IP4 192.168.133.2
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 3 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

== Setting global variable β€˜SIPDOMAIN’ to β€˜192.168.133.111’
<β€” Transmitting SIP response (303 bytes) to UDP:192.168.133.2:5060 β€”>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.133.2:5060;rport=5060;received=192.168.133.2;branch=z9hG4bK.3XGa100cA
Call-ID: SUgERujZGF
From: sip:37301@192.168.133.111;tag=YS2WnVOO8
To: sip:0744200@192.168.133.111
CSeq: 21 INVITE
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

-- Executing [0744200@phones:1] Dial("PJSIP/37301-00000016", "PJSIP/44200@asterisk2,,25") in new stack
-- Called PJSIP/44200@asterisk2

<β€” Transmitting SIP request (647 bytes) to UDP:192.168.133.222:5060 β€”>
INVITE sip:44200@192.168.133.222:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj7cb7d080-5938-4c7a-a001-daa3b78a7b61
From: sip:37301@192.168.133.111;tag=a07b5d57-41bb-422e-86f5-6023efc3fb41
To: sip:44200@192.168.133.222
Contact: sip:asterisk@192.168.133.111:5060
Call-ID: cf933fdb-becc-4981-87eb-150c00cdf29f
CSeq: 4299 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 certified/16.3-cert1
Content-Length: 0

<β€” Received SIP response (388 bytes) from UDP:192.168.133.222:5060 β€”>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.133.111:5060;rport=5060;received=192.168.133.111;branch=z9hG4bKPj7cb7d080-5938-4c7a-a001-daa3b78a7b61
Call-ID: cf933fdb-becc-4981-87eb-150c00cdf29f
From: sip:37301@192.168.133.111;tag=a07b5d57-41bb-422e-86f5-6023efc3fb41
To: sip:44200@192.168.133.222
CSeq: 4299 INVITE
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP response (578 bytes) from UDP:192.168.133.222:5060 β€”>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.133.111:5060;rport=5060;received=192.168.133.111;branch=z9hG4bKPj7cb7d080-5938-4c7a-a001-daa3b78a7b61
Call-ID: cf933fdb-becc-4981-87eb-150c00cdf29f
From: sip:37301@192.168.133.111;tag=a07b5d57-41bb-422e-86f5-6023efc3fb41
To: sip:44200@192.168.133.222;tag=1ffbdafb-3dd1-4938-a7a9-633bcb440d9a
CSeq: 4299 INVITE
Server: Asterisk PBX certified/16.3-cert1
Contact: sip:192.168.133.222:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Content-Length: 0

-- PJSIP/asterisk2-00000017 is ringing
-- PJSIP/asterisk2-00000017 is ringing

<β€” Transmitting SIP response (493 bytes) to UDP:192.168.133.2:5060 β€”>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.133.2:5060;rport=5060;received=192.168.133.2;branch=z9hG4bK.3XGa100cA
Call-ID: SUgERujZGF
From: sip:37301@192.168.133.111;tag=YS2WnVOO8
To: sip:0744200@192.168.133.111;tag=a5441eeb-001d-44ce-beeb-19eff782294b
CSeq: 21 INVITE
Server: Asterisk PBX certified/16.3-cert1
Contact: sip:192.168.133.111:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length: 0

<β€” Received SIP response (674 bytes) from UDP:192.168.133.222:5060 β€”>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.111:5060;rport=5060;received=192.168.133.111;branch=z9hG4bKPj7cb7d080-5938-4c7a-a001-daa3b78a7b61
Call-ID: cf933fdb-becc-4981-87eb-150c00cdf29f
From: sip:37301@192.168.133.111;tag=a07b5d57-41bb-422e-86f5-6023efc3fb41
To: sip:44200@192.168.133.222;tag=1ffbdafb-3dd1-4938-a7a9-633bcb440d9a
CSeq: 4299 INVITE
Server: Asterisk PBX certified/16.3-cert1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: sip:192.168.133.222:5060
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Length: 0

-- PJSIP/asterisk2-00000017 answered PJSIP/37301-00000016

<β€” Transmitting SIP request (436 bytes) to UDP:192.168.133.222:5060 β€”>
ACK sip:192.168.133.222:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj31d87846-2028-4967-be95-27c435bd9a6e
From: sip:37301@192.168.133.111;tag=a07b5d57-41bb-422e-86f5-6023efc3fb41
To: sip:44200@192.168.133.222;tag=1ffbdafb-3dd1-4938-a7a9-633bcb440d9a
Call-ID: cf933fdb-becc-4981-87eb-150c00cdf29f
CSeq: 4299 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Transmitting SIP request (436 bytes) to UDP:192.168.133.222:5060 β€”>
BYE sip:192.168.133.222:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj837c3ea1-fdbb-4414-9ca0-6cf10dbade2f
From: sip:37301@192.168.133.111;tag=a07b5d57-41bb-422e-86f5-6023efc3fb41
To: sip:44200@192.168.133.222;tag=1ffbdafb-3dd1-4938-a7a9-633bcb440d9a
Call-ID: cf933fdb-becc-4981-87eb-150c00cdf29f
CSeq: 4300 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Transmitting SIP response (848 bytes) to UDP:192.168.133.2:5060 β€”>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.2:5060;rport=5060;received=192.168.133.2;branch=z9hG4bK.3XGa100cA
Call-ID: SUgERujZGF
From: sip:37301@192.168.133.111;tag=YS2WnVOO8
To: sip:0744200@192.168.133.111;tag=a5441eeb-001d-44ce-beeb-19eff782294b
CSeq: 21 INVITE
Server: Asterisk PBX certified/16.3-cert1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: sip:192.168.133.111:5060
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 278

v=0
o=- 3132 2318 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.111
t=0 0
m=audio 10664 RTP/AVP 0 8 3 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Channel PJSIP/asterisk2-00000017 joined 'simple_bridge' basic-bridge <7eef12a1-7b50-452a-b745-ce0b65700a41>
-- Channel PJSIP/37301-00000016 joined 'simple_bridge' basic-bridge <7eef12a1-7b50-452a-b745-ce0b65700a41>

<β€” Received SIP request (469 bytes) from UDP:192.168.133.222:5060 β€”>
BYE sip:asterisk@192.168.133.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.222:5060;rport;branch=z9hG4bKPj53e497ce-d675-4789-abc5-dd9ac7d49252
From: sip:44200@192.168.133.222;tag=1ffbdafb-3dd1-4938-a7a9-633bcb440d9a
To: sip:37301@192.168.133.111;tag=a07b5d57-41bb-422e-86f5-6023efc3fb41
Call-ID: cf933fdb-becc-4981-87eb-150c00cdf29f
CSeq: 8205 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP response (422 bytes) from UDP:192.168.133.222:5060 β€”>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.111:5060;rport=5060;received=192.168.133.111;branch=z9hG4bKPj837c3ea1-fdbb-4414-9ca0-6cf10dbade2f
Call-ID: cf933fdb-becc-4981-87eb-150c00cdf29f
From: sip:37301@192.168.133.111;tag=a07b5d57-41bb-422e-86f5-6023efc3fb41
To: sip:44200@192.168.133.222;tag=1ffbdafb-3dd1-4938-a7a9-633bcb440d9a
CSeq: 4300 BYE
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Transmitting SIP response (422 bytes) to UDP:192.168.133.222:5060 β€”>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.222:5060;rport=5060;received=192.168.133.222;branch=z9hG4bKPj53e497ce-d675-4789-abc5-dd9ac7d49252
Call-ID: cf933fdb-becc-4981-87eb-150c00cdf29f
From: sip:44200@192.168.133.222;tag=1ffbdafb-3dd1-4938-a7a9-633bcb440d9a
To: sip:37301@192.168.133.111;tag=a07b5d57-41bb-422e-86f5-6023efc3fb41
CSeq: 8205 BYE
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

-- Channel PJSIP/asterisk2-00000017 left 'simple_bridge' basic-bridge <7eef12a1-7b50-452a-b745-ce0b65700a41>
-- Channel PJSIP/37301-00000016 left 'simple_bridge' basic-bridge <7eef12a1-7b50-452a-b745-ce0b65700a41>

== Spawn extension (phones, 0744200, 1) exited non-zero on β€˜PJSIP/37301-00000016’
<β€” Received SIP request (619 bytes) from UDP:192.168.133.2:5060 β€”>
ACK sip:192.168.133.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.2:5060;rport;branch=z9hG4bK.RxeWnMiyq
From: sip:37301@192.168.133.111;tag=YS2WnVOO8
To: sip:0744200@192.168.133.111;tag=a5441eeb-001d-44ce-beeb-19eff782294b
CSeq: 21 ACK
Call-ID: SUgERujZGF
Max-Forwards: 70
Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582798429/9269dbf5a18714bb0dc6ed40da73c276”, algorithm=md5, opaque=β€œ17c62c06790f2e60”, username=β€œ37301”, uri="sip:0744200@192.168.133.111", response=β€œ66c44a8d5ccd2dcf2eda6e9240697bb3”, cnonce=β€œGNalh-PtMNk9AtmK”, nc=00000001, qop=auth
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)

<β€” Transmitting SIP request (423 bytes) to UDP:192.168.133.2:5060 β€”>
BYE sip:37301@192.168.133.2;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj48bb8a01-0775-4e83-97ae-f4b1be40b6ec
From: sip:0744200@192.168.133.111;tag=a5441eeb-001d-44ce-beeb-19eff782294b
To: sip:37301@192.168.133.111;tag=YS2WnVOO8
Call-ID: SUgERujZGF
CSeq: 12568 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP response (364 bytes) from UDP:192.168.133.2:5060 β€”>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj48bb8a01-0775-4e83-97ae-f4b1be40b6ec
From: sip:0744200@192.168.133.111;tag=a5441eeb-001d-44ce-beeb-19eff782294b
To: sip:37301@192.168.133.111;tag=YS2WnVOO8
Call-ID: SUgERujZGF
CSeq: 12568 BYE
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Supported: replaces, outbound, gruu

i repeat i tested the call before and it was perfect i dont know what happened now.
thanks in advance
best regards

there is this BYE request that is being sent before even i answer the call…
so when i am answering it is hanging up … i

CLI of sip phone (37301)<β€”>Asterisk <β€”> PSTN Gateway <β€”> Analog Phone(37400) when allow=all

<β€” Received SIP request (1083 bytes) from UDP:192.168.133.9:5060 β€”>
INVITE sip:5537400@192.168.133.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.deKa0SJHD;rport
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111
CSeq: 20 INVITE
Call-ID: qAJzmzMXW4
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 495
Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=β€œurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6”
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)

v=0
o=37200 860 3348 IN IP4 192.168.133.9
s=Talk
c=IN IP4 192.168.133.9
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr

<β€” Transmitting SIP response (477 bytes) to UDP:192.168.133.9:5060 β€”>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.deKa0SJHD
Call-ID: qAJzmzMXW4
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111;tag=z9hG4bK.deKa0SJHD
CSeq: 20 INVITE
WWW-Authenticate: Digest realm=β€œasterisk”,nonce=β€œ1582803850/c68d9c4942ea27702747fb2ae45a5fd2”,opaque=β€œ6fdadc21648d812e”,algorithm=md5,qop=β€œauth”
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP request (384 bytes) from UDP:192.168.133.9:5060 β€”>
ACK sip:5537400@192.168.133.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.deKa0SJHD;rport
Call-ID: qAJzmzMXW4
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111;tag=z9hG4bK.deKa0SJHD
Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=β€œurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6”
Max-Forwards: 70
CSeq: 20 ACK

<β€” Received SIP request (1368 bytes) from UDP:192.168.133.9:5060 β€”>
INVITE sip:5537400@192.168.133.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.64jPZgZQ4;rport
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111
CSeq: 21 INVITE
Call-ID: qAJzmzMXW4
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 495
Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=β€œurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6”
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582803850/c68d9c4942ea27702747fb2ae45a5fd2”, algorithm=md5, opaque=β€œ6fdadc21648d812e”, username=β€œ37200”, uri="sip:5537400@192.168.133.111", response=β€œ2f8bf71cf466f61770b1b7e676b9f74c”, cnonce=β€œfnYHwNjtNYFfx2iS”, nc=00000001, qop=auth

v=0
o=37200 860 3348 IN IP4 192.168.133.9
s=Talk
c=IN IP4 192.168.133.9
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr

== Setting global variable β€˜SIPDOMAIN’ to β€˜192.168.133.111’
<β€” Transmitting SIP response (303 bytes) to UDP:192.168.133.9:5060 β€”>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.64jPZgZQ4
Call-ID: qAJzmzMXW4
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111
CSeq: 21 INVITE
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

-- Executing [5537400@phones:1] NoOp("PJSIP/37200-0000005c", "") in new stack
-- Executing [5537400@phones:2] Dial("PJSIP/37200-0000005c", "PJSIP/37400@gateway,,25") in new stack
-- Called PJSIP/37400@gateway

<β€” Transmitting SIP request (648 bytes) to UDP:192.168.133.110:5060 β€”>
INVITE sip:37400@192.168.133.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj81916518-66c5-4dba-bad6-aba3fe4cc705
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110
Contact: sip:asterisk@192.168.133.111:5060
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
CSeq: 12306 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 certified/16.3-cert1
Content-Length: 0

<β€” Received SIP response (409 bytes) from UDP:192.168.133.110:5060 β€”>
SIP/2.0 100 Trying
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
Content-Length: 0
CSeq: 12306 INVITE
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj81916518-66c5-4dba-bad6-aba3fe4cc705
Quintum: 0b03313336

<β€” Received SIP response (794 bytes) from UDP:192.168.133.110:5060 β€”>
SIP/2.0 183 Session Progress
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
Contact: sip:37400@192.168.133.110
Content-Length: 282
Content-Type: application/sdp
CSeq: 12306 INVITE
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj81916518-66c5-4dba-bad6-aba3fe4cc705
Quintum: 070e9777887800313106001e03808081

v=0
o=Quintum 31 31 IN IP4 192.168.133.110
s=VoipCall
c=IN IP4 192.168.133.110
t=0 0
m=audio 10308 RTP/AVP 0 8 0 101
c=IN IP4 192.168.133.110
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv

-- PJSIP/gateway-0000005d is making progress passing it to PJSIP/37200-0000005c
-- PJSIP/gateway-0000005d is making progress passing it to PJSIP/37200-0000005c

<β€” Transmitting SIP response (790 bytes) to UDP:192.168.133.9:5060 β€”>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.64jPZgZQ4
Call-ID: qAJzmzMXW4
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
CSeq: 21 INVITE
Server: Asterisk PBX certified/16.3-cert1
Contact: sip:192.168.133.111:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Content-Length: 254

v=0
o=- 860 3350 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.111
t=0 0
m=audio 14112 RTP/AVP 0 8 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<β€” Transmitting SIP response (790 bytes) to UDP:192.168.133.9:5060 β€”>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.64jPZgZQ4
Call-ID: qAJzmzMXW4
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
CSeq: 21 INVITE
Server: Asterisk PBX certified/16.3-cert1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: sip:192.168.133.111:5060
Content-Type: application/sdp
Content-Length: 254

v=0
o=- 860 3350 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.111
t=0 0
m=audio 14112 RTP/AVP 0 8 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<β€” Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β€”>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582803821/abda582c4f90086ba17566405b3c9887”,opaque=β€œ455484dc74b416e2”,algorithm=md5,qop=auth, username=β€œ37400”, uri=β€œsip:192.168.133.111”, response=β€œ47a5d9673ce59d2e9856077099ab93f3”, cnonce=β€œ95d98670”, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 639 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0425

<β€” Transmitting SIP response (548 bytes) to UDP:192.168.133.110:5060 β€”>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0425
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-0425
CSeq: 639 REGISTER
WWW-Authenticate: Digest realm=β€œasterisk”,nonce=β€œ1582803851/0a8b6a3ea35c8b932bdd2e7d7008585d”,opaque=β€œ6191bf2630478c73”,stale=true,algorithm=md5,qop=β€œauth”
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β€”>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582803851/0a8b6a3ea35c8b932bdd2e7d7008585d”,opaque=β€œ6191bf2630478c73”,algorithm=md5,qop=auth, username=β€œ37400”, uri=β€œsip:192.168.133.111”, response=β€œccf9e19eb04dcd85904ae482cb684567”, cnonce=β€œ95d98670”, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 640 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0427

<β€” Transmitting SIP response (523 bytes) to UDP:192.168.133.110:5060 β€”>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0427
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-0427
CSeq: 640 REGISTER
Date: Thu, 27 Feb 2020 11:44:11 GMT
Contact: sip:37400@192.168.133.110:5060
Contact: sip:37400@192.168.133.110;expires=59
Expires: 60
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP response (737 bytes) from UDP:192.168.133.110:5060 β€”>
SIP/2.0 200 OK
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
Contact: sip:37400@192.168.133.110
Content-Length: 282
Content-Type: application/sdp
CSeq: 12306 INVITE
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj81916518-66c5-4dba-bad6-aba3fe4cc705

v=0
o=Quintum 32 32 IN IP4 192.168.133.110
s=VoipCall
c=IN IP4 192.168.133.110
t=0 0
m=audio 10308 RTP/AVP 0 8 0 101
c=IN IP4 192.168.133.110
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv

-- PJSIP/gateway-0000005d answered PJSIP/37200-0000005c

<β€” Transmitting SIP request (674 bytes) to UDP:192.168.133.110:5060 β€”>
ACK sip:37400@192.168.133.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPja4bac9e7-55dc-4548-a96c-a73eae2dfd9b
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
CSeq: 12306 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 31 33 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.111
t=0 0
m=audio 17872 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<β€” Transmitting SIP response (824 bytes) to UDP:192.168.133.9:5060 β€”>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.64jPZgZQ4
Call-ID: qAJzmzMXW4
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
CSeq: 21 INVITE
Server: Asterisk PBX certified/16.3-cert1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: sip:192.168.133.111:5060
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 254

v=0
o=- 860 3350 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.111
t=0 0
m=audio 14112 RTP/AVP 0 8 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Channel PJSIP/gateway-0000005d joined 'simple_bridge' basic-bridge <20a71dcf-7ed0-443b-a4e6-890e38589f5a>
-- Channel PJSIP/37200-0000005c joined 'simple_bridge' basic-bridge <20a71dcf-7ed0-443b-a4e6-890e38589f5a>

<β€” Transmitting SIP request (917 bytes) to UDP:192.168.133.110:5060 β€”>
INVITE sip:37400@192.168.133.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjd2e1b922-8e89-4568-b134-ec5e7a289362
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
Contact: sip:asterisk@192.168.133.111:5060
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
CSeq: 12307 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 certified/16.3-cert1
Content-Type: application/sdp
Content-Length: 224

v=0
o=- 31 34 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.9
t=0 0
m=audio 7078 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<β€” Received SIP request (619 bytes) from UDP:192.168.133.9:5060 β€”>
ACK sip:192.168.133.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.9:5060;rport;branch=z9hG4bK.ASt31QCn7
From: sip:37200@192.168.133.111;tag=jmyXFMtGh
To: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
CSeq: 21 ACK
Call-ID: qAJzmzMXW4
Max-Forwards: 70
Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582803850/c68d9c4942ea27702747fb2ae45a5fd2”, algorithm=md5, opaque=β€œ6fdadc21648d812e”, username=β€œ37200”, uri="sip:5537400@192.168.133.111", response=β€œ2f8bf71cf466f61770b1b7e676b9f74c”, cnonce=β€œfnYHwNjtNYFfx2iS”, nc=00000001, qop=auth
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)

<β€” Transmitting SIP request (900 bytes) to UDP:192.168.133.9:5060 β€”>
INVITE sip:37200@192.168.133.9;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj01db1d75-244e-44a8-a7f1-7306378fd8d3
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Contact: sip:192.168.133.111:5060
Call-ID: qAJzmzMXW4
CSeq: 31911 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 certified/16.3-cert1
Content-Type: application/sdp
Content-Length: 230

v=0
o=- 860 3351 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.110
t=0 0
m=audio 10308 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<β€” Received SIP response (286 bytes) from UDP:192.168.133.9:5060 β€”>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj01db1d75-244e-44a8-a7f1-7306378fd8d3
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31911 INVITE

<β€” Received SIP response (764 bytes) from UDP:192.168.133.9:5060 β€”>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj01db1d75-244e-44a8-a7f1-7306378fd8d3
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31911 INVITE
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=β€œurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6”
Content-Type: application/sdp
Content-Length: 146

v=0
o=37200 860 3350 IN IP4 192.168.133.9
s=Talk
c=IN IP4 192.168.133.9
t=0 0
m=audio 7078 RTP/AVP 0 100
a=rtpmap:100 telephone-event/8000

<β€” Transmitting SIP request (399 bytes) to UDP:192.168.133.9:5060 β€”>
ACK sip:37200@192.168.133.9;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj5444a648-3ecf-43df-8369-b9ab1b4c6386
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31911 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP request (402 bytes) from UDP:192.168.133.110:5060 β€”>
BYE sip:asterisk@192.168.133.111:5060 SIP/2.0
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
Content-Length: 0
CSeq: 12307 BYE
From: sip:37400@192.168.133.110;tag=c0a8856e-35
Max-Forwards: 70
To: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0428

<β€” Transmitting SIP response (376 bytes) to UDP:192.168.133.110:5060 β€”>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0428
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
From: sip:37400@192.168.133.110;tag=c0a8856e-35
To: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
CSeq: 12307 BYE
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

-- Channel PJSIP/gateway-0000005d left 'native_rtp' basic-bridge <20a71dcf-7ed0-443b-a4e6-890e38589f5a>
-- Channel PJSIP/37200-0000005c left 'native_rtp' basic-bridge <20a71dcf-7ed0-443b-a4e6-890e38589f5a>

<β€” Transmitting SIP request (900 bytes) to UDP:192.168.133.9:5060 β€”>
INVITE sip:37200@192.168.133.9;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjffb03d42-0e07-41c3-b1be-6e5ba7109d52
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Contact: sip:192.168.133.111:5060
Call-ID: qAJzmzMXW4
CSeq: 31912 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 certified/16.3-cert1
Content-Type: application/sdp
Content-Length: 230

v=0
o=- 860 3352 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.111
t=0 0
m=audio 14112 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

== Spawn extension (phones, 5537400, 2) exited non-zero on β€˜PJSIP/37200-0000005c’
<β€” Received SIP response (286 bytes) from UDP:192.168.133.9:5060 β€”>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjffb03d42-0e07-41c3-b1be-6e5ba7109d52
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31912 INVITE

<β€” Received SIP response (764 bytes) from UDP:192.168.133.9:5060 β€”>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjffb03d42-0e07-41c3-b1be-6e5ba7109d52
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31912 INVITE
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=β€œurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6”
Content-Type: application/sdp
Content-Length: 146

v=0
o=37200 860 3352 IN IP4 192.168.133.9
s=Talk
c=IN IP4 192.168.133.9
t=0 0
m=audio 7078 RTP/AVP 0 100
a=rtpmap:100 telephone-event/8000

<β€” Transmitting SIP request (399 bytes) to UDP:192.168.133.9:5060 β€”>
ACK sip:37200@192.168.133.9;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj57b41596-43d1-4f2b-8ea1-64615c1b9643
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31912 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Transmitting SIP request (399 bytes) to UDP:192.168.133.9:5060 β€”>
BYE sip:37200@192.168.133.9;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj7f970460-3a57-4fdf-ace6-c9914416ea1e
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31913 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP response (358 bytes) from UDP:192.168.133.9:5060 β€”>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj7f970460-3a57-4fdf-ace6-c9914416ea1e
From: sip:5537400@192.168.133.111;tag=d2b97259-c6b5-4789-a066-82f663888789
To: sip:37200@192.168.133.111;tag=jmyXFMtGh
Call-ID: qAJzmzMXW4
CSeq: 31913 BYE
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Supported: replaces, outbound

<β€” Transmitting SIP request (917 bytes) to UDP:192.168.133.110:5060 β€”>
INVITE sip:37400@192.168.133.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjd2e1b922-8e89-4568-b134-ec5e7a289362
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
Contact: sip:asterisk@192.168.133.111:5060
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
CSeq: 12307 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 certified/16.3-cert1
Content-Type: application/sdp
Content-Length: 224

v=0
o=- 31 34 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.9
t=0 0
m=audio 7078 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<β€” Transmitting SIP request (917 bytes) to UDP:192.168.133.110:5060 β€”>
INVITE sip:37400@192.168.133.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjd2e1b922-8e89-4568-b134-ec5e7a289362
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
Contact: sip:asterisk@192.168.133.111:5060
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
CSeq: 12307 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 certified/16.3-cert1
Content-Type: application/sdp
Content-Length: 224

v=0
o=- 31 34 IN IP4 192.168.133.111
s=Asterisk
c=IN IP4 192.168.133.9
t=0 0
m=audio 7078 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<β€” Received SIP response (360 bytes) from UDP:192.168.133.110:5060 β€”>
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
Content-Length: 0
CSeq: 12307 INVITE
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjd2e1b922-8e89-4568-b134-ec5e7a289362

<β€” Transmitting SIP request (413 bytes) to UDP:192.168.133.110:5060 β€”>
ACK sip:37400@192.168.133.110 SIP/2.0
Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPjd2e1b922-8e89-4568-b134-ec5e7a289362
From: sip:37200@192.168.133.111;tag=3683f132-4de2-48a2-b836-2a0b873b9f82
To: sip:37400@192.168.133.110;tag=c0a8856e-35
Call-ID: 01768be2-4a75-491b-b650-e240256d97c2
CSeq: 12307 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β€”>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582803851/0a8b6a3ea35c8b932bdd2e7d7008585d”,opaque=β€œ6191bf2630478c73”,algorithm=md5,qop=auth, username=β€œ37400”, uri=β€œsip:192.168.133.111”, response=β€œccf9e19eb04dcd85904ae482cb684567”, cnonce=β€œ95d98670”, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 641 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0429

<β€” Transmitting SIP response (523 bytes) to UDP:192.168.133.110:5060 β€”>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0429
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-0429
CSeq: 641 REGISTER
Date: Thu, 27 Feb 2020 11:44:41 GMT
Contact: sip:37400@192.168.133.110:5060
Contact: sip:37400@192.168.133.110;expires=59
Expires: 60
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β€”>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582803851/0a8b6a3ea35c8b932bdd2e7d7008585d”,opaque=β€œ6191bf2630478c73”,algorithm=md5,qop=auth, username=β€œ37400”, uri=β€œsip:192.168.133.111”, response=β€œccf9e19eb04dcd85904ae482cb684567”, cnonce=β€œ95d98670”, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 642 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042a

<β€” Transmitting SIP response (548 bytes) to UDP:192.168.133.110:5060 β€”>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042a
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-042a
CSeq: 642 REGISTER
WWW-Authenticate: Digest realm=β€œasterisk”,nonce=β€œ1582803910/bd7dfb7b5ddf63009ca0392a4302337c”,opaque=β€œ2e731e5e691d79d7”,stale=true,algorithm=md5,qop=β€œauth”
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β€”>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582803910/bd7dfb7b5ddf63009ca0392a4302337c”,opaque=β€œ2e731e5e691d79d7”,algorithm=md5,qop=auth, username=β€œ37400”, uri=β€œsip:192.168.133.111”, response=β€œ43ae11a82dafa21df4f4d4f4b6e3e630”, cnonce=β€œ95d98670”, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 643 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042c

<β€” Transmitting SIP response (523 bytes) to UDP:192.168.133.110:5060 β€”>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042c
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-042c
CSeq: 643 REGISTER
Date: Thu, 27 Feb 2020 11:45:10 GMT
Contact: sip:37400@192.168.133.110:5060
Contact: sip:37400@192.168.133.110;expires=59
Expires: 60
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β€”>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582803910/bd7dfb7b5ddf63009ca0392a4302337c”,opaque=β€œ2e731e5e691d79d7”,algorithm=md5,qop=auth, username=β€œ37400”, uri=β€œsip:192.168.133.111”, response=β€œ43ae11a82dafa21df4f4d4f4b6e3e630”, cnonce=β€œ95d98670”, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 644 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042d

<β€” Transmitting SIP response (523 bytes) to UDP:192.168.133.110:5060 β€”>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042d
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-042d
CSeq: 644 REGISTER
Date: Thu, 27 Feb 2020 11:45:40 GMT
Contact: sip:37400@192.168.133.110:5060
Contact: sip:37400@192.168.133.110;expires=59
Expires: 60
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

<β€” Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β€”>
REGISTER sip:192.168.133.111 SIP/2.0
Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582803910/bd7dfb7b5ddf63009ca0392a4302337c”,opaque=β€œ2e731e5e691d79d7”,algorithm=md5,qop=auth, username=β€œ37400”, uri=β€œsip:192.168.133.111”, response=β€œ43ae11a82dafa21df4f4d4f4b6e3e630”, cnonce=β€œ95d98670”, nc=00000001
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
Contact: sip:37400@192.168.133.110
Content-Length: 0
CSeq: 645 REGISTER
Expires: 30
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
Max-Forwards: 70
To: sip:37400@192.168.133.111
User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04
Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042e

<β€” Transmitting SIP response (548 bytes) to UDP:192.168.133.110:5060 β€”>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-042e
Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110
From: sip:37400@192.168.133.111;tag=c0a8856e-2c
To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-042e
CSeq: 645 REGISTER
WWW-Authenticate: Digest realm=β€œasterisk”,nonce=β€œ1582803969/b5bcfa6c274e8881485c7c8b12db59d9”,opaque=β€œ77f67719619803ee”,stale=true,algorithm=md5,qop=β€œauth”
Server: Asterisk PBX certified/16.3-cert1
Content-Length: 0

i solved the problem between the 2 asterisks where i edited this in asterisk1 and asterisk2 trunk :
before:
allow=all
after:
allow=alaw,ulaw,gsm

and it worked. it looked like it was a delay while negotiating between the 2 endpoints about the codecs leading to this problem.

but now the problem is still the same at :
sip phone (37301)<β€”>Asterisk <β€”> PSTN Gateway <β€”> Analog Phone(37400)

although i edited : allow=all to allow=alaw,ulaw
at both pjsip.conf and gateway interface but same problem still arises.

CLI of sip phone (37301)<β€”>Asterisk <β€”> PSTN Gateway <β€”> Analog Phone(37400) when allow=alaw,ulaw :

<β€” Received SIP request (1084 bytes) from UDP:192.168.133.9:5060 β€”>

INVITE sip:5537400@192.168.133.111 SIP/2.0

Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.MHXMQ-0fT;rport

From: sip:37200@192.168.133.111;tag=qUh9CY-11

To: sip:5537400@192.168.133.111

CSeq: 20 INVITE

Call-ID: 3XoezsrI0y

Max-Forwards: 70

Supported: replaces, outbound

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE

Content-Type: application/sdp

Content-Length: 496

Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=β€œurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6”

User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)

v=0

o=37200 3674 1098 IN IP4 192.168.133.9

s=Talk

c=IN IP4 192.168.133.9

t=0 0

a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics

m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100

a=rtpmap:96 opus/48000/2

a=fmtp:96 useinbandfec=1

a=rtpmap:97 speex/16000

a=fmtp:97 vbr=on

a=rtpmap:98 speex/8000

a=fmtp:98 vbr=on

a=rtpmap:101 telephone-event/48000

a=rtpmap:99 telephone-event/16000

a=rtpmap:100 telephone-event/8000

a=rtcp-fb:* trr-int 5000

a=rtcp-fb:* ccm tmmbr

<β€” Transmitting SIP response (477 bytes) to UDP:192.168.133.9:5060 β€”>

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.MHXMQ-0fT

Call-ID: 3XoezsrI0y

From: sip:37200@192.168.133.111;tag=qUh9CY-11

To: sip:5537400@192.168.133.111;tag=z9hG4bK.MHXMQ-0fT

CSeq: 20 INVITE

WWW-Authenticate: Digest realm=β€œasterisk”,nonce=β€œ1582804077/d5099a10150776cea86a990f9bacef3a”,opaque=β€œ099c157b20166471”,algorithm=md5,qop=β€œauth”

Server: Asterisk PBX certified/16.3-cert1

Content-Length: 0

<β€” Received SIP request (384 bytes) from UDP:192.168.133.9:5060 β€”>

ACK sip:5537400@192.168.133.111 SIP/2.0

Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.MHXMQ-0fT;rport

Call-ID: 3XoezsrI0y

From: sip:37200@192.168.133.111;tag=qUh9CY-11

To: sip:5537400@192.168.133.111;tag=z9hG4bK.MHXMQ-0fT

Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=β€œurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6”

Max-Forwards: 70

CSeq: 20 ACK

<β€” Received SIP request (1369 bytes) from UDP:192.168.133.9:5060 β€”>

INVITE sip:5537400@192.168.133.111 SIP/2.0

Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.it15xmX4J;rport

From: sip:37200@192.168.133.111;tag=qUh9CY-11

To: sip:5537400@192.168.133.111

CSeq: 21 INVITE

Call-ID: 3XoezsrI0y

Max-Forwards: 70

Supported: replaces, outbound

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE

Content-Type: application/sdp

Content-Length: 496

Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=β€œurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6”

User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)

Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582804077/d5099a10150776cea86a990f9bacef3a”, algorithm=md5, opaque=β€œ099c157b20166471”, username=β€œ37200”, uri="sip:5537400@192.168.133.111", response=β€œf4a36af3fae7f4d0c44551385533cff6”, cnonce=β€œ0au5DX0xXFqVqnmQ”, nc=00000001, qop=auth

v=0

o=37200 3674 1098 IN IP4 192.168.133.9

s=Talk

c=IN IP4 192.168.133.9

t=0 0

a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics

m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100

a=rtpmap:96 opus/48000/2

a=fmtp:96 useinbandfec=1

a=rtpmap:97 speex/16000

a=fmtp:97 vbr=on

a=rtpmap:98 speex/8000

a=fmtp:98 vbr=on

a=rtpmap:101 telephone-event/48000

a=rtpmap:99 telephone-event/16000

a=rtpmap:100 telephone-event/8000

a=rtcp-fb:* trr-int 5000

a=rtcp-fb:* ccm tmmbr

== Setting global variable β€˜SIPDOMAIN’ to β€˜192.168.133.111’

<β€” Transmitting SIP response (303 bytes) to UDP:192.168.133.9:5060 β€”>

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.it15xmX4J

Call-ID: 3XoezsrI0y

From: sip:37200@192.168.133.111;tag=qUh9CY-11

To: sip:5537400@192.168.133.111

CSeq: 21 INVITE

Server: Asterisk PBX certified/16.3-cert1

Content-Length: 0

– Executing [5537400@phones:1] NoOp(β€œPJSIP/37200-00000060”, β€œβ€) in new stack

– Executing [5537400@phones:2] Dial(β€œPJSIP/37200-00000060”, β€œPJSIP/37400@gateway,25”) in new stack

– Called PJSIP/37400@gateway

<β€” Transmitting SIP request (946 bytes) to UDP:192.168.133.110:5060 β€”>

INVITE sip:37400@192.168.133.110:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj627ecd31-350b-4748-8c27-862b5e02a83e

From: sip:37200@192.168.133.111;tag=c515cd63-3d77-49f8-9648-90d4b13cbaa6

To: sip:37400@192.168.133.110

Contact: sip:asterisk@192.168.133.111:5060

Call-ID: da614c3f-5e5e-4d87-956d-a73c8a37025c

CSeq: 1334 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 certified/16.3-cert1

Content-Type: application/sdp

Content-Length: 265

v=0

o=- 478466055 478466055 IN IP4 192.168.133.111

s=Asterisk

c=IN IP4 192.168.133.111

t=0 0

m=audio 10596 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:150

a=sendrecv

<β€” Received SIP response (408 bytes) from UDP:192.168.133.110:5060 β€”>

SIP/2.0 100 Trying

Call-ID: da614c3f-5e5e-4d87-956d-a73c8a37025c

Content-Length: 0

CSeq: 1334 INVITE

From: sip:37200@192.168.133.111;tag=c515cd63-3d77-49f8-9648-90d4b13cbaa6

To: sip:37400@192.168.133.110;tag=c0a8856e-37

User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04

Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj627ecd31-350b-4748-8c27-862b5e02a83e

Quintum: 0b03313432

<β€” Received SIP response (400 bytes) from UDP:192.168.133.110:5060 β€”>

SIP/2.0 503 Service Unavailable

Call-ID: da614c3f-5e5e-4d87-956d-a73c8a37025c

Content-Length: 0

CSeq: 1334 INVITE

From: sip:37200@192.168.133.111;tag=c515cd63-3d77-49f8-9648-90d4b13cbaa6

To: sip:37400@192.168.133.110;tag=c0a8856e-37

User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04

Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj627ecd31-350b-4748-8c27-862b5e02a83e

<β€” Transmitting SIP request (417 bytes) to UDP:192.168.133.110:5060 β€”>

ACK sip:37400@192.168.133.110:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.133.111:5060;rport;branch=z9hG4bKPj627ecd31-350b-4748-8c27-862b5e02a83e

From: sip:37200@192.168.133.111;tag=c515cd63-3d77-49f8-9648-90d4b13cbaa6

To: sip:37400@192.168.133.110;tag=c0a8856e-37

Call-ID: da614c3f-5e5e-4d87-956d-a73c8a37025c

CSeq: 1334 ACK

Max-Forwards: 70

User-Agent: Asterisk PBX certified/16.3-cert1

Content-Length: 0

== Everyone is busy/congested at this time (1:0/1/0)

– Executing [5537400@phones:3] Hangup(β€œPJSIP/37200-00000060”, β€œβ€) in new stack

== Spawn extension (phones, 5537400, 3) exited non-zero on β€˜PJSIP/37200-00000060’

<β€” Transmitting SIP response (381 bytes) to UDP:192.168.133.9:5060 β€”>

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 192.168.133.9:5060;rport=5060;received=192.168.133.9;branch=z9hG4bK.it15xmX4J

Call-ID: 3XoezsrI0y

From: sip:37200@192.168.133.111;tag=qUh9CY-11

To: sip:5537400@192.168.133.111;tag=cf2d1439-fbeb-4401-9d48-4fde41e92fb7

CSeq: 21 INVITE

Server: Asterisk PBX certified/16.3-cert1

Reason: Q.850;cause=34

Content-Length: 0

<β€” Received SIP request (403 bytes) from UDP:192.168.133.9:5060 β€”>

ACK sip:5537400@192.168.133.111 SIP/2.0

Via: SIP/2.0/UDP 192.168.133.9:5060;branch=z9hG4bK.it15xmX4J;rport

Call-ID: 3XoezsrI0y

From: sip:37200@192.168.133.111;tag=qUh9CY-11

To: sip:5537400@192.168.133.111;tag=cf2d1439-fbeb-4401-9d48-4fde41e92fb7

Contact: sip:37200@192.168.133.9;transport=udp;+sip.instance=β€œurn:uuid:313ef843-7fe8-43c9-88d4-20fac982aeb6”

Max-Forwards: 70

CSeq: 21 ACK

<β€” Received SIP request (680 bytes) from UDP:192.168.133.110:5060 β€”>

REGISTER sip:192.168.133.111 SIP/2.0

Authorization: Digest realm=β€œasterisk”, nonce=β€œ1582804058/2caf4a8cd201defcb1c394f6c3005605”,opaque=β€œ269bba1646f13712”,algorithm=md5,qop=auth, username=β€œ37400”, uri=β€œsip:192.168.133.111”, response=β€œbecbc1a281630d9bcdc83e6643a9ad45”, cnonce=β€œ95d98670”, nc=00000001

Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110

Contact: sip:37400@192.168.133.110

Content-Length: 0

CSeq: 652 REGISTER

Expires: 30

From: sip:37400@192.168.133.111;tag=c0a8856e-2c

Max-Forwards: 70

To: sip:37400@192.168.133.111

User-Agent: Quintum/1.0.0 SN/0030E1106AC1 SW/P108-09-04

Via: SIP/2.0/UDP 192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0438

<β€” Transmitting SIP response (523 bytes) to UDP:192.168.133.110:5060 β€”>

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.133.110;rport=5060;received=192.168.133.110;branch=z9hG4bK-tenor-c0a8-856e-0438

Call-ID: call-71A0A2AC-0DA2-2110-0B0D-0@192.168.133.110

From: sip:37400@192.168.133.111;tag=c0a8856e-2c

To: sip:37400@192.168.133.111;tag=z9hG4bK-tenor-c0a8-856e-0438

CSeq: 652 REGISTER

Date: Thu, 27 Feb 2020 11:48:07 GMT

Contact: sip:37400@192.168.133.110:5060

Contact: sip:37400@192.168.133.110;expires=59

Expires: 60

Server: Asterisk PBX certified/16.3-cert1

Content-Length: 0

so now i understand what is happening when i set allow=all ! it is going through this unwanted delay which leads to sending BYE request ! but when i set allow=alaw,ulaw then i dont understand what is happening since i did configure both to same codecs

It’s trying to do direct media, that is have each side send media directly. When that is done a remote side sends a BYE terminating the call. Set β€œdirect_media” to β€œno” on the endpoints.

pjsip.conf

[gateway] ;; this is my trunk to the sip/pstn gateway
type=aor
contact=sip:37400@192.168.133.110:5060

[gateway]
type=endpoint
context=phones
allow=all
aors=gateway
direct_media=no
[gateway]
;type=identify
match=193.168.133.110
endpoint=gateway

[37400] ;; this is my extension to the analog phone
type = endpoint
context = phones
disallow = all
allow = ulaw,alaw
aors = 37400
uth = auth37400
device_state_busy_at=1
direct_media=no
[37400]
type = aor
max_contacts = 2
contact=sip:37400@192.168.133.110:5060
[auth37400]
type=auth
auth_type=userpass
password=123
username=37400

[37200]
type = endpoint
context = phones
disallow = all
allow = ulaw,alaw,gsm
aors = 37200
auth = auth37200
device_state_busy_at=1
direct_media=no
[37200]
type = aor
max_contacts = 1

[auth37200]
type=auth
auth_type=userpass
password=123
username=37200

here at pjsip.cong i set direct_medoa to no at gateway and endpoints.
if allow=all at gateway then it rings but still directly ends after answering and if allow=alaw,ulaw then it does not ring at all …
do i have to set direct media to no from the interface of the gateway
but i really it is not about direct media since same problem was solved but changing audio codecs when i was to make a call from one asterisk to another

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