Troubleshoot: Reason: Q.850;cause=34

NoFate · June 7, 2022, 11:19am

Hi
I’m using sip.linhome.org as trunk for outgoing call , so my softphones are registered to sip.linhome.org

Dialplan:

exten => 1,1,NoOp()
 same => n,Progress()
 same => n,Set(CALLERID(num)=xxxx)
 same => n,Set(__DYNAMIC_FEATURES=door)
 same => n,Dial(PJSIP/yyyy@trunk-linhome)

Trunk setup:

[trunk-linhome-auth]
type=auth
auth_type=userpass
password=1234
username=xxx

[trunk-linhome-aor]
type=aor
contact=sip:sip.linhome.org
qualify_frequency=30

[trunk-linhome-registration]
type=registration
outbound_auth=trunk-linhome-auth
server_uri=sip:sip.linhome.org
client_uri=sip:xxxx@sip.linhome.org
retry_interval=30
 
[trunk-linhome]
type=endpoint
context=default
disallow=all
allow=ulaw,alaw
allow=h264
outbound_auth=trunk-linhome-auth
aors=trunk-linhome-aor
from_domain=sip.linhome.org
direct_media=yes

 
[trunk-linhome-identify]
type=identify
endpoint=trunk-linhome
match=sip.linhome.org

Everything works as expected, i can dial out using this trunk!

But in the linhome app, i want to make some changes, so i compiled a new version , but i have issues with it… I see the audio call coming in, but it immediately aborts… when i make a video call, i can also answer, but then there is no audio … so i think its an audio problem, but i have no idea how to troubleshoot
i was comparing log between the working app and the compiled app, but i dont see any reason why it should not work ??

log working:

2022/06/07 12:13:44.040155 10.8.0.2:5060 -> 192.168.0.17:5050
INVITE sip:1@192.168.0.17:5050 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.2:5060;branch=z9hG4bK.Ttj4sxkSi;rport
From: "6000" <sip:6000@192.168.0.17>;tag=NS1DLkbUN
To: sip:1@192.168.0.17
CSeq: 20 INVITE
Call-ID: JXdZpiSjS0
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 222
Contact: <sip:6000@10.8.0.2;transport=udp>;expires=3599;+sip.instance="<urn:uuid:f95ada1c-90bb-0060-b75e-4f7e6d20508b>";+org.linphone.specs="ephemeral/1.1,groupchat/1.1"
User-Agent: Linphone Desktop/4.4.1 (XS-LT-335) Windows 10 Version 2009, Qt 5.15.2 LinphoneCore/5.1.19-1-g6cdd0918e

v=0
o=6000 305 1144 IN IP4 10.8.0.2
s=Talk
c=IN IP4 10.8.0.2
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 65418 RTP/AVP 0 8
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr


2022/06/07 12:13:44.040532 192.168.0.17:5050 -> 10.8.0.2:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.8.0.2:5060;rport=5060;received=10.8.0.2;branch=z9hG4bK.Ttj4sxkSi
Call-ID: JXdZpiSjS0
From: "6000" <sip:6000@192.168.0.17>;tag=NS1DLkbUN
To: <sip:1@192.168.0.17>;tag=z9hG4bK.Ttj4sxkSi
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1654596824/1c30b85ac7b776caed8a539e155f8f96",opaque="2c80a15e43c3453f",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.10.1
Content-Length:  0



2022/06/07 12:13:44.127680 10.8.0.2:5060 -> 192.168.0.17:5050
ACK sip:1@192.168.0.17:5050 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.2:5060;branch=z9hG4bK.Ttj4sxkSi;rport
Call-ID: JXdZpiSjS0
From: "6000" <sip:6000@192.168.0.17>;tag=NS1DLkbUN
To: <sip:1@192.168.0.17>;tag=z9hG4bK.Ttj4sxkSi
Contact: <sip:6000@10.8.0.2;transport=udp>;expires=3599;+sip.instance="<urn:uuid:f95ada1c-90bb-0060-b75e-4f7e6d20508b>";+org.linphone.specs="ephemeral/1.1,groupchat/1.1"
Max-Forwards: 70
CSeq: 20 ACK



2022/06/07 12:13:44.127811 10.8.0.2:5060 -> 192.168.0.17:5050
INVITE sip:1@192.168.0.17:5050 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.2:5060;branch=z9hG4bK.t15HjTt-7;rport
From: "6000" <sip:6000@192.168.0.17>;tag=NS1DLkbUN
To: sip:1@192.168.0.17
CSeq: 21 INVITE
Call-ID: JXdZpiSjS0
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 222
Contact: <sip:6000@10.8.0.2;transport=udp>;expires=3599;+sip.instance="<urn:uuid:f95ada1c-90bb-0060-b75e-4f7e6d20508b>";+org.linphone.specs="ephemeral/1.1,groupchat/1.1"
User-Agent: Linphone Desktop/4.4.1 (XS-LT-335) Windows 10 Version 2009, Qt 5.15.2 LinphoneCore/5.1.19-1-g6cdd0918e
Authorization:  Digest realm="asterisk", nonce="1654596824/1c30b85ac7b776caed8a539e155f8f96", algorithm=md5, opaque="2c80a15e43c3453f", username="6000",  uri="sip:1@192.168.0.17:5050", response="7cdd8342dd7e841c1e628c306fa861b2", cnonce="NHhGi0HZL44k6ozZ", nc=00000001, qop=auth

v=0
o=6000 305 1144 IN IP4 10.8.0.2
s=Talk
c=IN IP4 10.8.0.2
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 65418 RTP/AVP 0 8
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr


2022/06/07 12:13:44.128456 192.168.0.17:5050 -> 10.8.0.2:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.0.2:5060;rport=5060;received=10.8.0.2;branch=z9hG4bK.t15HjTt-7
Call-ID: JXdZpiSjS0
From: "6000" <sip:6000@192.168.0.17>;tag=NS1DLkbUN
To: <sip:1@192.168.0.17>
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.1
Content-Length:  0



2022/06/07 12:13:44.154610 192.168.0.17:5050 -> 10.8.0.2:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.8.0.2:5060;rport=5060;received=10.8.0.2;branch=z9hG4bK.t15HjTt-7
Call-ID: JXdZpiSjS0
From: "6000" <sip:6000@192.168.0.17>;tag=NS1DLkbUN
To: <sip:1@192.168.0.17>;tag=b4e07756-35d6-4483-bf14-e96c28632788
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.1
Contact: <sip:192.168.0.17:5050>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   192

v=0
o=- 305 1146 IN IP4 192.168.0.17
s=Asterisk
c=IN IP4 192.168.0.17
t=0 0
m=audio 18788 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:150
a=sendrecv


2022/06/07 12:13:44.282608 192.168.0.17:5050 -> 10.8.0.2:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.8.0.2:5060;rport=5060;received=10.8.0.2;branch=z9hG4bK.t15HjTt-7
Call-ID: JXdZpiSjS0
From: "6000" <sip:6000@192.168.0.17>;tag=NS1DLkbUN
To: <sip:1@192.168.0.17>;tag=b4e07756-35d6-4483-bf14-e96c28632788
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.0.17:5050>
Content-Type: application/sdp
Content-Length:   192

v=0
o=- 305 1146 IN IP4 192.168.0.17
s=Asterisk
c=IN IP4 192.168.0.17
t=0 0
m=audio 18788 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:150
a=sendrecv


2022/06/07 12:13:44.422904 192.168.0.17:5050 -> 10.8.0.2:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.8.0.2:5060;rport=5060;received=10.8.0.2;branch=z9hG4bK.t15HjTt-7
Call-ID: JXdZpiSjS0
From: "6000" <sip:6000@192.168.0.17>;tag=NS1DLkbUN
To: <sip:1@192.168.0.17>;tag=b4e07756-35d6-4483-bf14-e96c28632788
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.0.17:5050>
Content-Type: application/sdp
Content-Length:   192

v=0
o=- 305 1146 IN IP4 192.168.0.17
s=Asterisk
c=IN IP4 192.168.0.17
t=0 0
m=audio 18788 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:150
a=sendrecv


2022/06/07 12:13:50.624163 192.168.0.17:5050 -> 10.8.0.2:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.0.2:5060;rport=5060;received=10.8.0.2;branch=z9hG4bK.t15HjTt-7
Call-ID: JXdZpiSjS0
From: "6000" <sip:6000@192.168.0.17>;tag=NS1DLkbUN
To: <sip:1@192.168.0.17>;tag=b4e07756-35d6-4483-bf14-e96c28632788
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.0.17:5050>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   192

v=0
o=- 305 1146 IN IP4 192.168.0.17
s=Asterisk
c=IN IP4 192.168.0.17
t=0 0
m=audio 18788 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:150
a=sendrecv


2022/06/07 12:13:50.889432 10.8.0.2:5060 -> 192.168.0.17:5050
ACK sip:192.168.0.17:5050 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.2:5060;rport;branch=z9hG4bK.LAJLbRIs1
From: "6000" <sip:6000@192.168.0.17>;tag=NS1DLkbUN
To: <sip:1@192.168.0.17>;tag=b4e07756-35d6-4483-bf14-e96c28632788
CSeq: 21 ACK
Call-ID: JXdZpiSjS0
Max-Forwards: 70
Authorization:  Digest realm="asterisk", nonce="1654596824/1c30b85ac7b776caed8a539e155f8f96", algorithm=md5, opaque="2c80a15e43c3453f", username="6000",  uri="sip:1@192.168.0.17:5050", response="7cdd8342dd7e841c1e628c306fa861b2", cnonce="NHhGi0HZL44k6ozZ", nc=00000001, qop=auth
User-Agent: Linphone Desktop/4.4.1 (XS-LT-335) Windows 10 Version 2009, Qt 5.15.2 LinphoneCore/5.1.19-1-g6cdd0918e



2022/06/07 12:13:59.452290 192.168.0.17:5050 -> 10.8.0.2:5060
BYE sip:6000@10.8.0.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:5050;rport;branch=z9hG4bKPj01d9b2ca-aec0-4c14-aade-d4a695e94c98
From: <sip:1@192.168.0.17>;tag=b4e07756-35d6-4483-bf14-e96c28632788
To: "6000" <sip:6000@192.168.0.17>;tag=NS1DLkbUN
Call-ID: JXdZpiSjS0
CSeq: 15606 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 18.10.1
Content-Length:  0



2022/06/07 12:13:59.494842 10.8.0.2:5060 -> 192.168.0.17:5050
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.17:5050;rport;branch=z9hG4bKPj01d9b2ca-aec0-4c14-aade-d4a695e94c98
From: <sip:1@192.168.0.17>;tag=b4e07756-35d6-4483-bf14-e96c28632788
To: "6000" <sip:6000@192.168.0.17>;tag=NS1DLkbUN
Call-ID: JXdZpiSjS0
CSeq: 15606 BYE
User-Agent: Linphone Desktop/4.4.1 (XS-LT-335) Windows 10 Version 2009, Qt 5.15.2 LinphoneCore/5.1.19-1-g6cdd0918e
Supported: replaces, outbound, gruu

Log non working, exact the same, but only receive an Q.850;cause=34

2022/06/07 12:16:54.405019 10.8.0.2:5060 -> 192.168.0.17:5050
INVITE sip:1@192.168.0.17:5050 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.2:5060;branch=z9hG4bK.RompJ1q4d;rport
From: "6000" <sip:6000@192.168.0.17>;tag=aj3ZrAIlR
To: sip:1@192.168.0.17
CSeq: 20 INVITE
Call-ID: UOUzZRe9m9
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 223
Contact: <sip:6000@10.8.0.2;transport=udp>;expires=3599;+sip.instance="<urn:uuid:f95ada1c-90bb-0060-b75e-4f7e6d20508b>";+org.linphone.specs="ephemeral/1.1,groupchat/1.1"
User-Agent: Linphone Desktop/4.4.1 (XS-LT-335) Windows 10 Version 2009, Qt 5.15.2 LinphoneCore/5.1.19-1-g6cdd0918e

v=0
o=6000 3033 3853 IN IP4 10.8.0.2
s=Talk
c=IN IP4 10.8.0.2
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 51320 RTP/AVP 0 8
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr


2022/06/07 12:16:54.405389 192.168.0.17:5050 -> 10.8.0.2:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.8.0.2:5060;rport=5060;received=10.8.0.2;branch=z9hG4bK.RompJ1q4d
Call-ID: UOUzZRe9m9
From: "6000" <sip:6000@192.168.0.17>;tag=aj3ZrAIlR
To: <sip:1@192.168.0.17>;tag=z9hG4bK.RompJ1q4d
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1654597014/c2d97e53b9c2a562a53e833a8a8f2de7",opaque="4dbd03c574adac75",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.10.1
Content-Length:  0



2022/06/07 12:16:54.483264 10.8.0.2:5060 -> 192.168.0.17:5050
ACK sip:1@192.168.0.17:5050 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.2:5060;branch=z9hG4bK.RompJ1q4d;rport
Call-ID: UOUzZRe9m9
From: "6000" <sip:6000@192.168.0.17>;tag=aj3ZrAIlR
To: <sip:1@192.168.0.17>;tag=z9hG4bK.RompJ1q4d
Contact: <sip:6000@10.8.0.2;transport=udp>;expires=3599;+sip.instance="<urn:uuid:f95ada1c-90bb-0060-b75e-4f7e6d20508b>";+org.linphone.specs="ephemeral/1.1,groupchat/1.1"
Max-Forwards: 70
CSeq: 20 ACK



2022/06/07 12:16:54.483540 10.8.0.2:5060 -> 192.168.0.17:5050
INVITE sip:1@192.168.0.17:5050 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.2:5060;branch=z9hG4bK.kjFUtxPVN;rport
From: "6000" <sip:6000@192.168.0.17>;tag=aj3ZrAIlR
To: sip:1@192.168.0.17
CSeq: 21 INVITE
Call-ID: UOUzZRe9m9
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 223
Contact: <sip:6000@10.8.0.2;transport=udp>;expires=3599;+sip.instance="<urn:uuid:f95ada1c-90bb-0060-b75e-4f7e6d20508b>";+org.linphone.specs="ephemeral/1.1,groupchat/1.1"
User-Agent: Linphone Desktop/4.4.1 (XS-LT-335) Windows 10 Version 2009, Qt 5.15.2 LinphoneCore/5.1.19-1-g6cdd0918e
Authorization:  Digest realm="asterisk", nonce="1654597014/c2d97e53b9c2a562a53e833a8a8f2de7", algorithm=md5, opaque="4dbd03c574adac75", username="6000",  uri="sip:1@192.168.0.17:5050", response="f597fc079e568248ad194d6deb42a760", cnonce="z~FTu93sVV1x2RX4", nc=00000001, qop=auth

v=0
o=6000 3033 3853 IN IP4 10.8.0.2
s=Talk
c=IN IP4 10.8.0.2
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 51320 RTP/AVP 0 8
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr


2022/06/07 12:16:54.484046 192.168.0.17:5050 -> 10.8.0.2:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.0.2:5060;rport=5060;received=10.8.0.2;branch=z9hG4bK.kjFUtxPVN
Call-ID: UOUzZRe9m9
From: "6000" <sip:6000@192.168.0.17>;tag=aj3ZrAIlR
To: <sip:1@192.168.0.17>
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.1
Content-Length:  0



2022/06/07 12:16:54.508503 192.168.0.17:5050 -> 10.8.0.2:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.8.0.2:5060;rport=5060;received=10.8.0.2;branch=z9hG4bK.kjFUtxPVN
Call-ID: UOUzZRe9m9
From: "6000" <sip:6000@192.168.0.17>;tag=aj3ZrAIlR
To: <sip:1@192.168.0.17>;tag=dd092536-6d95-42e3-9289-98c5d6f01b95
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.1
Contact: <sip:192.168.0.17:5050>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   193

v=0
o=- 3033 3855 IN IP4 192.168.0.17
s=Asterisk
c=IN IP4 192.168.0.17
t=0 0
m=audio 17154 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:150
a=sendrecv


2022/06/07 12:16:54.619487 192.168.0.17:5050 -> 10.8.0.2:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.8.0.2:5060;rport=5060;received=10.8.0.2;branch=z9hG4bK.kjFUtxPVN
Call-ID: UOUzZRe9m9
From: "6000" <sip:6000@192.168.0.17>;tag=aj3ZrAIlR
To: <sip:1@192.168.0.17>;tag=dd092536-6d95-42e3-9289-98c5d6f01b95
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.0.17:5050>
Content-Type: application/sdp
Content-Length:   193

v=0
o=- 3033 3855 IN IP4 192.168.0.17
s=Asterisk
c=IN IP4 192.168.0.17
t=0 0
m=audio 17154 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:150
a=sendrecv


2022/06/07 12:16:54.931705 192.168.0.17:5050 -> 10.8.0.2:5060
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.8.0.2:5060;rport=5060;received=10.8.0.2;branch=z9hG4bK.kjFUtxPVN
Call-ID: UOUzZRe9m9
From: "6000" <sip:6000@192.168.0.17>;tag=aj3ZrAIlR
To: <sip:1@192.168.0.17>;tag=dd092536-6d95-42e3-9289-98c5d6f01b95
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Reason: Q.850;cause=34
Content-Length:  0



2022/06/07 12:16:55.431721 192.168.0.17:5050 -> 10.8.0.2:5060
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.8.0.2:5060;rport=5060;received=10.8.0.2;branch=z9hG4bK.kjFUtxPVN
Call-ID: UOUzZRe9m9
From: "6000" <sip:6000@192.168.0.17>;tag=aj3ZrAIlR
To: <sip:1@192.168.0.17>;tag=dd092536-6d95-42e3-9289-98c5d6f01b95
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Reason: Q.850;cause=34
Content-Length:  0



2022/06/07 12:16:55.980400 10.8.0.2:5060 -> 192.168.0.17:5050
ACK sip:1@192.168.0.17:5050 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.2:5060;branch=z9hG4bK.kjFUtxPVN;rport
Call-ID: UOUzZRe9m9
From: "6000" <sip:6000@192.168.0.17>;tag=aj3ZrAIlR
To: <sip:1@192.168.0.17>;tag=dd092536-6d95-42e3-9289-98c5d6f01b95
Contact: <sip:6000@10.8.0.2;transport=udp>;expires=3599;+sip.instance="<urn:uuid:f95ada1c-90bb-0060-b75e-4f7e6d20508b>";+org.linphone.specs="ephemeral/1.1,groupchat/1.1"
Max-Forwards: 70
CSeq: 21 ACK



2022/06/07 12:16:56.170468 10.8.0.2:5060 -> 192.168.0.17:5050
ACK sip:1@192.168.0.17:5050 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.2:5060;branch=z9hG4bK.kjFUtxPVN;rport
Call-ID: UOUzZRe9m9
From: "6000" <sip:6000@192.168.0.17>;tag=aj3ZrAIlR
To: <sip:1@192.168.0.17>;tag=dd092536-6d95-42e3-9289-98c5d6f01b95
Contact: <sip:6000@10.8.0.2;transport=udp>;expires=3599;+sip.instance="<urn:uuid:f95ada1c-90bb-0060-b75e-4f7e6d20508b>";+org.linphone.specs="ephemeral/1.1,groupchat/1.1"
Max-Forwards: 70
CSeq: 21 ACK

jcolp · June 7, 2022, 11:29am

There is insufficient information. You’ve provided SIP traces for one side of things. You need to provide all call legs involved as well as the Asterisk console output.

NoFate · June 7, 2022, 11:33am

pjsip set logger on ? what kind of log level do you want me to enable?

jcolp · June 7, 2022, 11:34am

The “pjsip set logger on” will show the SIP traffic. Normal Asterisk console verbose output to show what is executing as well.

The problem with the information you’ve given is that it shows that Asterisk sent a 503 Service Unavailable, however that is generally caused by something the outgoing call leg has done (such as rejecting the call itself with a 503).

NoFate · June 7, 2022, 11:52am

Here is a complete debug, does it help?

debug.txt (127.0 KB)

jcolp · June 7, 2022, 12:24pm

There’s no SIP signaling in this, but I can say that “trunk-linhome” responded with a 488 SIP response, meaning it seemingly didn’t like the INVITE we sent. Why that is I don’t know.

NoFate · June 7, 2022, 12:41pm

could it be audio related? because if i initiate a video call, i can answer, but there is no audio , and the call breaks after 30 sec

but its still strange , if i take the original linhome app, then all works perfect, only an issue with the compiled app

jcolp · June 7, 2022, 12:42pm

I can’t speak for outside applications or code. I can only say what it did from the perspective of Asterisk.

NoFate · June 7, 2022, 12:56pm

is there a way to troubleshoot this?

jcolp · June 7, 2022, 12:59pm

Not from the side of Asterisk. It doesn’t seem to be an Asterisk issue, so any further investigation is outside of Asterisk.

NoFate · June 7, 2022, 1:17pm

ok, thnx for feedback
maybe i can try one thing, because i also see issues in debug log of app , about early media … how can i disable that in in asterisk?

i have now on trunk and endpoint configured this : direct_media=yes

setting it to no, doesnt help

jcolp · June 7, 2022, 1:20pm

There is no such option to disable early media.

system · July 7, 2022, 1:20pm

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.

Topic		Replies	Views
How to link an incoming trunk, now its always "s" Asterisk Dialplan	3	381	July 9, 2022
1 out of 5 calls no two way audio Asterisk SIP	1	162	March 24, 2023
Cannot register to trunk Asterisk Support	1	276	June 22, 2015
SIP trunk unable to connect Asterisk SIP	31	4613	April 2, 2019
SIP trunk cant register Asterisk SIP	3	25	January 18, 2025

Troubleshoot: Reason: Q.850;cause=34

Related topics