Ich habe bei mir auf einem Raspberry 3 ein Asterisk 192.168.2.34 und ein Linphone client (sip:[email protected]) am Laufen. Wenn ich diesen von meinem Handy mit der Linephone app anrufe funktioniert alles einwandfrei (Bild und Ton). Ruf ich diesen aber als Gast über den Türbaustein der beschränkte Loxone app an (sip:[email protected]) schafft er es nicht eine Verbindung aufzubauen.
Kann mir jemand einen Tip geben, wieso dies nicht klappt?
C:
root@loxberry:/etc/asterisk# asterisk -r
Asterisk 16.2.1~dfsg-1+deb10u1, Copyright (C) 1999 - 2018, Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 16.2.1~dfsg-1+deb10u1 currently running on loxberry (pid = 657)
<--- SIP read from UDP:192.168.2.34:5065 --->
<------------->
<--- SIP read from UDP:192.168.2.128:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.128:5060;rport;branch=z9hG4bKPjVUZrtC6RDqJ-nn5UgesJU-W8XR2pur0z
Max-Forwards: 70
From: sip:[email protected];tag=fQqJNQjffzZpjrJhFxqHkhx7cGw4QgJD
To: sip:[email protected]
Contact: <sip:[email protected]:5060;ob>
Call-ID: RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
CSeq: 14806 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Loxone Pjsua2 Wrapper
secHdrA: 3qomarv4737c0vankncs5ulgqb9t5i7
secHdrB: 5FA3E61EC1F66DB55878EC23F717A874
Content-Type: application/sdp
Content-Length: 476
v=0
o=- 3787248588 3787248588 IN IP4 192.168.2.128
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 8 0 104 3 98 97 99 9 96
c=IN IP4 192.168.2.128
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.128
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
<------------->
--- (17 headers 22 lines) ---
Sending to 192.168.2.128:5060 (NAT)
Sending to 192.168.2.128:5060 (NAT)
Using INVITE request as basis request - RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
No matching peer for 'smarthome' from '192.168.2.128:5060'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 104
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 99
Found RTP audio format 9
Found RTP audio format 96
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format iLBC for ID 104
Found audio description format GSM for ID 3
Found audio description format speex for ID 98
Found audio description format speex for ID 97
Found audio description format speex for ID 99
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|g722|speex|speex16|speex32|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.128:4000
Looking for 19 in intern (domain 192.168.2.34)
sip_route_dump: route/path hop: <sip:[email protected]:5060;ob>
<--- Transmitting (NAT) to 192.168.2.128:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.128:5060;branch=z9hG4bKPjVUZrtC6RDqJ-nn5UgesJU-W8XR2pur0z;received=192.168.2.128;rport=5060
From: sip:[email protected];tag=fQqJNQjffzZpjrJhFxqHkhx7cGw4QgJD
To: sip:[email protected]
Call-ID: RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
CSeq: 14806 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
Audio is at 14986
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.2.34:5065:
INVITE sip:[email protected]:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK26ecef35;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as21b02264
To: <sip:[email protected]:5065;transport=udp>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Date: Sun, 05 Jan 2020 21:29:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 294
v=0
o=root 59255726 59255726 IN IP4 192.168.2.34
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 192.168.2.34
t=0 0
m=audio 14986 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:192.168.2.34:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK26ecef35;rport
From: <sip:[email protected]>;tag=as21b02264
To: <sip:[email protected]:5065;transport=udp>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:192.168.2.34:5065 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK26ecef35;rport
From: <sip:[email protected]>;tag=as21b02264
To: <sip:[email protected]:5065;transport=udp>;tag=a3d1-c1
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Supported: replaces, outbound, gruu
<------------->
--- (8 headers 0 lines) ---
sip_route_dump: no route/path
<--- Transmitting (NAT) to 192.168.2.128:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.128:5060;branch=z9hG4bKPjVUZrtC6RDqJ-nn5UgesJU-W8XR2pur0z;received=192.168.2.128;rport=5060
From: sip:[email protected];tag=fQqJNQjffzZpjrJhFxqHkhx7cGw4QgJD
To: sip:[email protected];tag=as28c6af14
Call-ID: RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
CSeq: 14806 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.2.34:5065 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK26ecef35;rport
From: <sip:[email protected]>;tag=as21b02264
To: <sip:[email protected]:5065;transport=udp>;tag=a3d1-c1
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: <sip:[email protected]:5065;transport=udp>;expires=3600;+sip.instance="<urn:uuid:6f8832a5-7f96-0022-b0b4-6b319593a207>"
Content-Type: application/sdp
Content-Length: 142
v=0
o=19 1118 2306 IN IP4 192.168.2.34
s=Talk
c=IN IP4 192.168.2.34
t=0 0
m=audio 7078 RTP/AVP 0 8 96
a=rtpmap:96 telephone-event/8000
<------------->
--- (12 headers 7 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|gsm|h263|vp8), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.34:7078
sip_route_dump: route/path hop: <sip:[email protected]:5065;transport=udp>
Transmitting (NAT) to 192.168.2.34:5065:
ACK sip:[email protected]:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK678aa3cb;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as21b02264
To: <sip:[email protected]:5065;transport=udp>;tag=a3d1-c1
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length: 0
---
Audio is at 12420
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 192.168.2.128:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.128:5060;branch=z9hG4bKPjVUZrtC6RDqJ-nn5UgesJU-W8XR2pur0z;received=192.168.2.128;rport=5060
From: sip:[email protected];tag=fQqJNQjffzZpjrJhFxqHkhx7cGw4QgJD
To: sip:[email protected];tag=as28c6af14
Call-ID: RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
CSeq: 14806 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 296
v=0
o=root 224192528 224192528 IN IP4 192.168.2.34
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 192.168.2.34
t=0 0
m=audio 12420 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv
<------------>
<--- SIP read from UDP:192.168.2.128:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.128:5060;rport;branch=z9hG4bKPjVUZrtC6RDqJ-nn5UgesJU-W8XR2pur0z
Max-Forwards: 70
From: sip:[email protected];tag=fQqJNQjffzZpjrJhFxqHkhx7cGw4QgJD
To: sip:[email protected]
Contact: <sip:[email protected]:5060;ob>
Call-ID: RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
CSeq: 14806 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Loxone Pjsua2 Wrapper
secHdrA: 3qomarv4737c0vankncs5ulgqb9t5i7
secHdrB: 5FA3E61EC1F66DB55878EC23F717A874
Content-Type: application/sdp
Content-Length: 476
v=0
o=- 3787248588 3787248588 IN IP4 192.168.2.128
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 8 0 104 3 98 97 99 9 96
c=IN IP4 192.168.2.128
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.128
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
<------------->
--- (17 headers 22 lines) ---
Ignoring this INVITE request
<--- Transmitting (NAT) to 192.168.2.128:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.128:5060;branch=z9hG4bKPjVUZrtC6RDqJ-nn5UgesJU-W8XR2pur0z;received=192.168.2.128;rport=5060
From: sip:[email protected];tag=fQqJNQjffzZpjrJhFxqHkhx7cGw4QgJD
To: sip:[email protected]
Call-ID: RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
CSeq: 14806 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
Audio is at 12420
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 192.168.2.128:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.128:5060;branch=z9hG4bKPjVUZrtC6RDqJ-nn5UgesJU-W8XR2pur0z;received=192.168.2.128;rport=5060
From: sip:[email protected];tag=fQqJNQjffzZpjrJhFxqHkhx7cGw4QgJD
To: sip:[email protected];tag=as28c6af14
Call-ID: RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
CSeq: 14806 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 296
v=0
o=root 224192528 224192529 IN IP4 192.168.2.34
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 192.168.2.34
t=0 0
m=audio 12420 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv
<------------>
Retransmitting #1 (NAT) to 192.168.2.128:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.128:5060;branch=z9hG4bKPjVUZrtC6RDqJ-nn5UgesJU-W8XR2pur0z;received=192.168.2.128;rport=5060
From: sip:[email protected];tag=fQqJNQjffzZpjrJhFxqHkhx7cGw4QgJD
To: sip:[email protected];tag=as28c6af14
Call-ID: RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
CSeq: 14806 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 296
v=0
o=root 224192528 224192528 IN IP4 192.168.2.34
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 192.168.2.34
t=0 0
m=audio 12420 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:192.168.2.128:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.128:5060;rport;branch=z9hG4bKPjVUZrtC6RDqJ-nn5UgesJU-W8XR2pur0z
Max-Forwards: 70
From: sip:[email protected];tag=fQqJNQjffzZpjrJhFxqHkhx7cGw4QgJD
To: sip:[email protected]
Contact: <sip:[email protected]:5060;ob>
Call-ID: RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
CSeq: 14806 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Loxone Pjsua2 Wrapper
secHdrA: 3qomarv4737c0vankncs5ulgqb9t5i7
secHdrB: 5FA3E61EC1F66DB55878EC23F717A874
Content-Type: application/sdp
Content-Length: 476
v=0
o=- 3787248588 3787248588 IN IP4 192.168.2.128
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 8 0 104 3 98 97 99 9 96
c=IN IP4 192.168.2.128
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.128
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
<------------->
--- (17 headers 22 lines) ---
Ignoring this INVITE request
<--- Transmitting (NAT) to 192.168.2.128:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.128:5060;branch=z9hG4bKPjVUZrtC6RDqJ-nn5UgesJU-W8XR2pur0z;received=192.168.2.128;rport=5060
From: sip:[email protected];tag=fQqJNQjffzZpjrJhFxqHkhx7cGw4QgJD
To: sip:[email protected]
Call-ID: RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
CSeq: 14806 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
Audio is at 12420
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 192.168.2.128:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.128:5060;branch=z9hG4bKPjVUZrtC6RDqJ-nn5UgesJU-W8XR2pur0z;received=192.168.2.128;rport=5060
From: sip:[email protected];tag=fQqJNQjffzZpjrJhFxqHkhx7cGw4QgJD
To: sip:[email protected];tag=as28c6af14
Call-ID: RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
CSeq: 14806 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 296
v=0
o=root 224192528 224192530 IN IP4 192.168.2.34
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 192.168.2.34
t=0 0
m=audio 12420 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv
<------------>
Retransmitting #2 (NAT) to 192.168.2.128:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.128:5060;branch=z9hG4bKPjVUZrtC6RDqJ-nn5UgesJU-W8XR2pur0z;received=192.168.2.128;rport=5060
From: sip:[email protected];tag=fQqJNQjffzZpjrJhFxqHkhx7cGw4QgJD
To: sip:[email protected];tag=as28c6af14
Call-ID: RjSMbVFRaWe0K1u5JH0h3CDmtAjDdynS
CSeq: 14806 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 296
v=0
o=root 224192528 224192528 IN IP4 192.168.2.34
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 192.168.2.34
t=0 0
m=audio 12420 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv
---
<------------->
<--- SIP read from UDP:192.168.2.128:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.128:5060;rport;branch=z9hG4bKPjT9WByz5cq0aKiUe7vlYWoYGJB0bbWO8M
Max-Forwards: 70
From: sip:[email protected];tag=-KZb12.Gdpcl0hDYfsmOg7NEFmyTa.j5
To: sip:[email protected]
Contact: <sip:[email protected]:5060;ob>
Call-ID: jD7.Qe-H.LIpGqlwNcVN-7U5YPSy8R-3
CSeq: 6710 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Loxone Pjsua2 Wrapper
secHdrA: ccr0qmme60re7sv666m3ai601ul000c
secHdrB: B871864FE4A27320F06390755927BC99
Content-Type: application/sdp
Content-Length: 476
v=0
o=- 3787248331 3787248331 IN IP4 192.168.2.128
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 8 0 104 3 98 97 99 9 96
c=IN IP4 192.168.2.128
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.128
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
<------------->
--- (17 headers 22 lines) ---
Ignoring this INVITE request
<--- Transmitting (NAT) to 192.168.2.128:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.128:5060;branch=z9hG4bKPjT9WByz5cq0aKiUe7vlYWoYGJB0bbWO8M;received=192.168.2.128;rport=5060
From: sip:[email protected];tag=-KZb12.Gdpcl0hDYfsmOg7NEFmyTa.j5
To: sip:[email protected]
Call-ID: jD7.Qe-H.LIpGqlwNcVN-7U5YPSy8R-3
CSeq: 6710 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
Audio is at 12604
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 192.168.2.128:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.128:5060;branch=z9hG4bKPjT9WByz5cq0aKiUe7vlYWoYGJB0bbWO8M;received=192.168.2.128;rport=5060
From: sip:[email protected];tag=-KZb12.Gdpcl0hDYfsmOg7NEFmyTa.j5
To: sip:[email protected];tag=as3b98ee49
Call-ID: jD7.Qe-H.LIpGqlwNcVN-7U5YPSy8R-3
CSeq: 6710 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 296
v=0
o=root 357710664 357710670 IN IP4 192.168.2.34
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 192.168.2.34
t=0 0
m=audio 12604 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv
<------------>
Retransmitting #10 (NAT) to 192.168.2.128:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.128:5060;branch=z9hG4bKPjT9WByz5cq0aKiUe7vlYWoYGJB0bbWO8M;received=192.168.2.128;rport=5060
From: sip:[email protected];tag=-KZb12.Gdpcl0hDYfsmOg7NEFmyTa.j5
To: sip:[email protected];tag=as3b98ee49
Call-ID: jD7.Qe-H.LIpGqlwNcVN-7U5YPSy8R-3
CSeq: 6710 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 296
v=0
o=root 357710664 357710664 IN IP4 192.168.2.34
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 192.168.2.34
t=0 0
m=audio 12604 RTP/AVP 0 8 3 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:150
a=sendrecv
---
[Jan 5 22:26:04] WARNING[816]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission jD7.Qe-H.LIpGqlwNcVN-7U5YPSy8R-3 for seqno 6710 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Jan 5 22:26:04] WARNING[816]: chan_sip.c:4143 retrans_pkt: Hanging up call jD7.Qe-H.LIpGqlwNcVN-7U5YPSy8R-3 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Scheduling destruction of SIP dialog 'jD7.Qe-H.LIpGqlwNcVN-7U5YPSy8R-3' in 32000 ms (Method: INVITE)
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 192.168.2.128:5060:
BYE sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK0cffa759;rport
Max-Forwards: 70
From: sip:[email protected];tag=as3b98ee49
To: sip:[email protected];tag=-KZb12.Gdpcl0hDYfsmOg7NEFmyTa.j5
Call-ID: jD7.Qe-H.LIpGqlwNcVN-7U5YPSy8R-3
CSeq: 102 BYE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
---
Reliably Transmitting (NAT) to 192.168.2.34:5065:
BYE sip:[email protected]:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK03591d50;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as568ecc91
To: <sip:[email protected]:5065;transport=udp>;tag=6zRKISp
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
---
<--- SIP read from UDP:192.168.2.34:5065 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK03591d50;rport
From: <sip:[email protected]>;tag=as568ecc91
To: <sip:[email protected]:5065;transport=udp>;tag=6zRKISp
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Supported: replaces, outbound, gruu
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: INVITE
<--- SIP read from UDP:192.168.2.34:5065 --->
<------------->
Retransmitting #1 (NAT) to 192.168.2.128:5060:
BYE sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK0cffa759;rport
Max-Forwards: 70
From: sip:[email protected];tag=as3b98ee49
To: sip:[email protected];tag=-KZb12.Gdpcl0hDYfsmOg7NEFmyTa.j5
Call-ID: jD7.Qe-H.LIpGqlwNcVN-7U5YPSy8R-3
CSeq: 102 BYE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
---
Retransmitting #2 (NAT) to 192.168.2.128:5060:
BYE sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK0cffa759;rport
Max-Forwards: 70
From: sip:[email protected];tag=as3b98ee49
To: sip:[email protected];tag=-KZb12.Gdpcl0hDYfsmOg7NEFmyTa.j5
Call-ID: jD7.Qe-H.LIpGqlwNcVN-7U5YPSy8R-3
CSeq: 102 BYE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
---
Retransmitting #3 (NAT) to 192.168.2.128:5060:
BYE sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK0cffa759;rport
Max-Forwards: 70
From: sip:[email protected];tag=as3b98ee49
To: sip:[email protected];tag=-KZb12.Gdpcl0hDYfsmOg7NEFmyTa.j5
Call-ID: jD7.Qe-H.LIpGqlwNcVN-7U5YPSy8R-3
CSeq: 102 BYE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
Kann mir jemand einen Tip geben, wieso dies nicht klappt?