[Problem] Anruf als Gast (Loxone, Linphone, Asterisk)

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R

rani22

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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.

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?
 

Anhänge

  • extensions.conf.txt
    2.7 KB · Aufrufe: 4
  • sip.conf.txt
    759 Bytes · Aufrufe: 4
  1. Warum verwendest Du statt dem Channel-Driver chan_pjsip noch den alten chan_sip?
  2. Warum nutzt Du nicht das aktuelle Asterisk 16.7.0 sondern noch 16.2.1?
  3. Warum hantierst Du überhaupt mit „Guests“?
  4. Wo – in der Datei extension.conf – ist Dein Kontext [default], um den Anruf von Guests anzunehmen?
 
Besten Dank für deine Rückmeldung.

sonyKatze schrieb:
Vollzitat von darüber entfernt by stoney
Zum Vergrößern anklicken....

1. Wie kann ich dies beeinflussen?
2. sudo apt-get install asterisk hat mir diese Version auf dem raspberry installiert.
3. Die LoxoneApp hat keinen richtigen SIP cient. Ich kann dort lediglich ip vom sip server und name vom Empfänger hinterlegen. Wenn ich hier zum Beispiel 13 (Notebook mit Linephone) statt 19 (Raspberry mit linephone) hinterlege, klingelt der Notebook. Wenn ich dann aber auf annehmen gehe, passiert grundsätzlich das gleiche. Es wird keine Verbindung afgebaut.
4. In der sip.conf habe ich den context=intern angelegt. Werden dann nicht alle diesem zugeordnet? Brauche ich dann in der extension noch einen [default]? Wie müsste dieser dann aussehen?
 
Zuletzt bearbeitet von einem Moderator:
rani22 schrieb:
Wie kann ich [den SIP Channel-Driver] beeinflussen?
Zum Vergrößern anklicken....
Du arbeitest mit der Konfigurationsdatei pjsip.conf und lädst das Modul chan_sip erst gar nicht:
A) modules.conf und noload oder
B) make menuselect und dort dann chan_sip abhaken.
Das Asterisk-Team hat für die weiteren Schritte eine Anleitung. Allerdings würde ich an Deiner Stelle mit dem Buch zu Asterisk anfangen. Oder Asterisk ganz sausen lassen und gleich auf FreeSWITCH umschwenken. FreeSWITCH ist bei reinen SIP-Umgebungen einfach sinnvoller, weil das nicht mehr diesen SS7/ISDN-Kern mitschleift.
rani22 schrieb:
sudo apt-get install asterisk hat mir diese Version auf dem raspberry installiert.
Zum Vergrößern anklicken....
Asterisk ist in Debian (und seinen Derivaten) nicht bei jenen Paketen, die von Haus aus Sicherheitsupdates bekommen. Daher: Diesen Asterisk-Server nie öffentlich zugänglich machen, weil keine Sicherheitsupdates. Oder besser: Das Debian-Package nur zu Lern- und Probierzwecken nutzen; in allen anderen Fällen Asterisk selbst bauen.
rani22 schrieb:
klingelt der Notebook. Wenn ich dann aber auf annehmen gehe, passiert grundsätzlich das gleiche. Es wird keine Verbindung afgebaut.
Zum Vergrößern anklicken....
Wenn es klingelt, hat das SIP-INVITE geklappt. In dem Fall hast Du ein anderes Problem. Ich sehe in Deinem SIP-Trace „Ignoring this INVITE request“. Was das bedeutet, ist mir unklar. Noch nicht gesehen. Da muss jemand anderes ran.
rani22 schrieb:
Brauche ich dann in der extension noch einen [default]?
Zum Vergrößern anklicken....
Gäste schlagen auch im Default-Context auf. Dort gibst Du die Nebenstelle(n) an, die von Gästen kontaktiert werden dürfen – also bei Dir die „19“. Am Einfachsten machst Du das über eine Umleitung in Deinen üblichen Context – also bei Dir „intern“:
[default]
exten => 19,1,Goto(intern,19,1)
So kann man das allowguests=yes in anderen Contexts vermeiden.

Wo bekommt man mehr Infos über dieses „Loxone“. Das Teil basiert auf PjSIP. Ich verstehe nicht, warum das keine SIP-Authentifizierung unterstützen sollte.
 
Zuletzt bearbeitet:
Besten Dank, werde demnach mal versuchen auf pjsip zu wechseln und den default user hinzufügen.

sonyKatze schrieb:
Wo bekommt man mehr Infos über dieses „Loxone“. Das Teil basiert auf PjSIP. Ich verstehe nicht, warum das keine SIP-Authentifizierung unterstützen sollte.
Zum Vergrößern anklicken....
Mehr Infos für Loxone findest du hier:
https://www.loxone.com/dede/kb/tuersteuerung/
und
Forum Eintrag

Loxone verkauft eine eigene Türgegensprechanlage und verlangt dafür einen premium Preis. Ich denke deshalb ist das ganze so eingeschränkt.
 
Der Call von [email protected] nach [email protected] "funktioniert" scheinbar bis zum Abnehmen seitens 19. Asterisk schickt ein 200 OK an smarthome. Smarthome müsste jetzt mit ack antworten - tut es aber nicht.
Stattdessen kommt das gleiche Invite von smarthome nochmal - was aber von Asterisk wg. der gleichen CSeq abgelehnt wird. Die beiden quatschen danach aneinander vorbei.

Für mich sieht das so aus, als ob die Pakete von Asterisk bei smarthome nicht ankommen oder diese dort verworfen werden. Netz checken! Wieso ist da überhaupt NAT im Spiel? Die NAT-Config von Asterisk ist korrekt?
 
Habe das ding jetzt auf PJSIP umgestellt. Leider hat das phyton ding nicht funktioniert und ich habe das ganze kurz von Hand neu aufgesetzt.
Wenn ich Versuche eine Verbindung aufzubauen, nimmt er auch irgendwie ab aber es wird kein ton übertragen. Nach etwa 30s kommt beim raspberry das Besetzzeichen und auf der App kommt "Audio Connection failed The extension is not available ([email protected]). (408)" dies ist die gleiche Meldung die auch schon vor dem Wechsel auf PJSIP gekommen ist.

Der log sieht nun so aus:
Bash: 
  == Setting global variable 'SIPDOMAIN' to '192.168.2.34'
    -- Executing [19@anonymous:1] Goto("PJSIP/anonymous-00000030", "intern,19,1") in new stack
    -- Goto (intern,19,1)
    -- Executing [19@intern:1] Dial("PJSIP/anonymous-00000030", "PJSIP/19") in new stack
    -- Called PJSIP/19
    -- PJSIP/19-00000031 is ringing
    -- PJSIP/19-00000031 is ringing
    -- PJSIP/19-00000031 answered PJSIP/anonymous-00000030
       > 0x738ed2b0 -- Strict RTP learning after remote address set to: 192.168.2.34:7078
       > 0x738ea850 -- Strict RTP learning after remote address set to: 192.168.2.131:4000
    -- Channel PJSIP/19-00000031 joined 'simple_bridge' basic-bridge <45f580ed-be66-4d21-994d-49475dfcf7db>
    -- Channel PJSIP/anonymous-00000030 joined 'simple_bridge' basic-bridge <45f580ed-be66-4d21-994d-49475dfcf7db>
       > Bridge 45f580ed-be66-4d21-994d-49475dfcf7db: switching from simple_bridge technology to native_rtp
       > Remotely bridged 'PJSIP/anonymous-00000030' and 'PJSIP/19-00000031' - media will flow directly between them
       > 0x738ed2b0 -- Strict RTP switching to RTP target address 192.168.2.34:7078 as source
    -- Channel PJSIP/19-00000031 left 'native_rtp' basic-bridge <45f580ed-be66-4d21-994d-49475dfcf7db>
    -- Channel PJSIP/anonymous-00000030 left 'native_rtp' basic-bridge <45f580ed-be66-4d21-994d-49475dfcf7db>
  == Spawn extension (intern, 19, 1) exited non-zero on 'PJSIP/anonymous-00000030'

Wenn ich von meinem Notebook das Raspberry anrufe funktioniert dies und das log sieht so aus:
Bash: 
  == Setting global variable 'SIPDOMAIN' to '192.168.2.34'
    -- Executing [19@local:1] Dial("PJSIP/13-00000038", "PJSIP/19") in new stack
    -- Called PJSIP/19
    -- PJSIP/19-00000039 is ringing
    -- PJSIP/19-00000039 is ringing
    -- PJSIP/19-00000039 answered PJSIP/13-00000038
       > 0x73a6aa00 -- Strict RTP learning after remote address set to: 192.168.2.34:7078
       > 0x73a549a8 -- Strict RTP learning after remote address set to: 92.104.97.158:7078
    -- Channel PJSIP/19-00000039 joined 'simple_bridge' basic-bridge <0b314a39-bd8f-40e1-9dfb-5de90d329c88>
    -- Channel PJSIP/13-00000038 joined 'simple_bridge' basic-bridge <0b314a39-bd8f-40e1-9dfb-5de90d329c88>
       > Bridge 0b314a39-bd8f-40e1-9dfb-5de90d329c88: switching from simple_bridge technology to native_rtp
       > Remotely bridged 'PJSIP/13-00000038' and 'PJSIP/19-00000039' - media will flow directly between them
       > 0x73a6aa00 -- Strict RTP switching to RTP target address 192.168.2.34:7078 as source
       > 0x73a549a8 -- Strict RTP learning after remote address set to: 192.168.2.123:7078
    -- Channel PJSIP/13-00000038 left 'native_rtp' basic-bridge <0b314a39-bd8f-40e1-9dfb-5de90d329c88>
    -- Channel PJSIP/19-00000039 left 'native_rtp' basic-bridge <0b314a39-bd8f-40e1-9dfb-5de90d329c88>
  == Spawn extension (local, 19, 1) exited non-zero on 'PJSIP/13-00000038'

Was habe ich hier falsch gemacht?
 

Anhänge

  • extensions.conf_200107_2.txt
    2.7 KB · Aufrufe: 0
  • pjsip.conf_200107.txt
    881 Bytes · Aufrufe: 1
Wie ich Dir schon geschrieben habe, sieht das nach einem Netzwerkproblem aus: der Datenfluss von Asterisk -> smarthome scheint nicht zu funktionieren (das wird durch einen Wechsel des sip-Backends nicht gefixt). Geht denn ein ping? Paketfilter korrekt eingestellt? Was soll das NAT an dieser Stelle?

Ansonsten für einen Trace:
Code: 
asterisk -r
pjsip set logger on

Dann call starten
danach mit
Code: 
pjsip set logger off
wieder ausschalten
 
gehtdoch schrieb:
Wie ich Dir schon geschrieben habe, sieht das nach einem Netzwerkproblem aus: der Datenfluss von Asterisk -> smarthome scheint nicht zu funktionieren (das wird durch einen Wechsel des sip-Backends nicht gefixt). Geht denn ein ping? Paketfilter korrekt eingestellt? Was soll das NAT an dieser Stelle?
Zum Vergrößern anklicken....
Was meinst du mit ping und wie kann ich den Paketfilter einstellen? Nach meinem Verständnis brauche ich kein NAT, da alles im gleichen Netz liegt.
Eventuell als Nachtrag: Ich habe auf dem gleichen Handy, wie die Loxone App, auch noch ein linphone client, der kann das Raspi auch ohne Probleme anrufen.
pjsip logger werde ich heute Abend ausprobieren.
 
sonyKatze schrieb:
Fehlermeldung nicht gepostet
Zum Vergrößern anklicken....
Der Skript war erst gar nicht in meiner Installation enthalten. Ich habe diesen dann dazu kopiert. Dieser hat jedoch dann auf Grund von Berechtigungen gar nicht erst gestartet. Da ich relativ wenige Einstellungen gemacht habe, habe ich mich dazu entschlossen nicht nach der Ursache zu suchen und die Einträge einfach neu zu erstellen.


Hier noch das log des aufgezeichneten Trace, wenn die Loxone App am raspi anrufen will. Nach etwa 1min bricht das ganze mit Einer Fehlermeldung der App ab. Gleich wie vor dem Wechsel auf pjsip

Bash: 
loxberry*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Received SIP request (1183 bytes) from UDP:192.168.2.139:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.139:5060;rport;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Max-Forwards: 70
From: sip:[email protected];tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: sip:[email protected]
Contact: <sip:[email protected]:5060;ob>
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
CSeq: 7865 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: quvp3co73havl1r5qns1hk5telp3opb
secHdrB: 9D368D6908B582393DA37B75C80CB698
Content-Type: application/sdp
Content-Length:   476

v=0
o=- 3788455638 3788455638 IN IP4 192.168.2.139
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 192.168.2.139
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.139
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

<--- Transmitting SIP response (366 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length:  0


<--- Transmitting SIP request (916 bytes) to UDP: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;rport;branch=z9hG4bKPjf34459a7-586e-4d22-a4f4-bf555fb1decc
From: <sip:[email protected]>;tag=552bfc74-60e4-4dcc-a615-8f94586a3ffa
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 5e6cb4df-7a0c-46b1-96f3-6c80cf6e1441
CSeq: 3436 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 16.2.1~dfsg-1+deb10u1
Content-Type: application/sdp
Content-Length:   237

v=0
o=- 1941575522 1941575522 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 19318 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

<--- Received SIP response (285 bytes) from UDP:192.168.2.34:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.34:5060;rport;branch=z9hG4bKPjf34459a7-586e-4d22-a4f4-bf555fb1decc
From: <sip:[email protected]>;tag=552bfc74-60e4-4dcc-a615-8f94586a3ffa
To: sip:[email protected]
Call-ID: 5e6cb4df-7a0c-46b1-96f3-6c80cf6e1441
CSeq: 3436 INVITE


<--- Received SIP response (385 bytes) from UDP:192.168.2.34:5065 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.34:5060;rport;branch=z9hG4bKPjf34459a7-586e-4d22-a4f4-bf555fb1decc
From: <sip:[email protected]>;tag=552bfc74-60e4-4dcc-a615-8f94586a3ffa
To: <sip:[email protected]>;tag=Zr28~or
Call-ID: 5e6cb4df-7a0c-46b1-96f3-6c80cf6e1441
CSeq: 3436 INVITE
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)
Supported: replaces, outbound, gruu


<--- Transmitting SIP response (553 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Contact: <sip:192.168.2.34:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


<--- Received SIP response (793 bytes) from UDP:192.168.2.34:5065 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.2.34:5060;rport;branch=z9hG4bKPjf34459a7-586e-4d22-a4f4-bf555fb1decc
From: <sip:[email protected]>;tag=552bfc74-60e4-4dcc-a615-8f94586a3ffa
To: <sip:[email protected]>;tag=Zr28~or
Call-ID: 5e6cb4df-7a0c-46b1-96f3-6c80cf6e1441
CSeq: 3436 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=3599;+sip.instance="<urn:uuid:6f8832a5-7f96-0022-b0b4-6b319593a207>"
Content-Type: application/sdp
Content-Length: 142

v=0
o=19 1309 2098 IN IP4 192.168.2.34
s=Talk
c=IN IP4 192.168.2.34
t=0 0
m=audio 7078 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000

<--- Transmitting SIP request (414 bytes) to UDP: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;rport;branch=z9hG4bKPj60d1f798-da94-47ae-b7c1-fec8a31150b1
From: <sip:[email protected]>;tag=552bfc74-60e4-4dcc-a615-8f94586a3ffa
To: <sip:[email protected]>;tag=Zr28~or
Call-ID: 5e6cb4df-7a0c-46b1-96f3-6c80cf6e1441
CSeq: 3436 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length:  0


<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (928 bytes) to UDP: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;rport;branch=z9hG4bKPja27eac3b-8dcd-49c0-96fb-3f40337f6733
From: <sip:[email protected]>;tag=552bfc74-60e4-4dcc-a615-8f94586a3ffa
To: <sip:[email protected]>;tag=Zr28~or
Contact: <sip:[email protected]:5060>
Call-ID: 5e6cb4df-7a0c-46b1-96f3-6c80cf6e1441
CSeq: 3437 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 16.2.1~dfsg-1+deb10u1
Content-Type: application/sdp
Content-Length:   237

v=0
o=- 1941575522 1941575523 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.139
t=0 0
m=audio 4000 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

<--- Received SIP response (299 bytes) from UDP:192.168.2.34:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.34:5060;rport;branch=z9hG4bKPja27eac3b-8dcd-49c0-96fb-3f40337f6733
From: <sip:[email protected]>;tag=552bfc74-60e4-4dcc-a615-8f94586a3ffa
To: <sip:[email protected]>;tag=Zr28~or
Call-ID: 5e6cb4df-7a0c-46b1-96f3-6c80cf6e1441
CSeq: 3437 INVITE


<--- Received SIP response (793 bytes) from UDP:192.168.2.34:5065 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.2.34:5060;rport;branch=z9hG4bKPja27eac3b-8dcd-49c0-96fb-3f40337f6733
From: <sip:[email protected]>;tag=552bfc74-60e4-4dcc-a615-8f94586a3ffa
To: <sip:[email protected]>;tag=Zr28~or
Call-ID: 5e6cb4df-7a0c-46b1-96f3-6c80cf6e1441
CSeq: 3437 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=3599;+sip.instance="<urn:uuid:6f8832a5-7f96-0022-b0b4-6b319593a207>"
Content-Type: application/sdp
Content-Length: 142

v=0
o=19 1309 2100 IN IP4 192.168.2.34
s=Talk
c=IN IP4 192.168.2.34
t=0 0
m=audio 7078 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000

<--- Transmitting SIP request (414 bytes) to UDP: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;rport;branch=z9hG4bKPjc0b4e53d-b68f-4ad9-be9b-7fcd2782411e
From: <sip:[email protected]>;tag=552bfc74-60e4-4dcc-a615-8f94586a3ffa
To: <sip:[email protected]>;tag=Zr28~or
Call-ID: 5e6cb4df-7a0c-46b1-96f3-6c80cf6e1441
CSeq: 3437 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length:  0


<--- Received SIP request (1183 bytes) from UDP:192.168.2.139:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.139:5060;rport;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Max-Forwards: 70
From: sip:[email protected];tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: sip:[email protected]
Contact: <sip:[email protected]:5060;ob>
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
CSeq: 7865 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: quvp3co73havl1r5qns1hk5telp3opb
secHdrB: 9D368D6908B582393DA37B75C80CB698
Content-Type: application/sdp
Content-Length:   476

v=0
o=- 3788455638 3788455638 IN IP4 192.168.2.139
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 192.168.2.139
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.139
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

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (1183 bytes) from UDP:192.168.2.139:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.139:5060;rport;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Max-Forwards: 70
From: sip:[email protected];tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: sip:[email protected]
Contact: <sip:[email protected]:5060;ob>
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
CSeq: 7865 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: quvp3co73havl1r5qns1hk5telp3opb
secHdrB: 9D368D6908B582393DA37B75C80CB698
Content-Type: application/sdp
Content-Length:   476

v=0
o=- 3788455638 3788455638 IN IP4 192.168.2.139
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 192.168.2.139
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.139
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

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (1183 bytes) from UDP:192.168.2.139:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.139:5060;rport;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Max-Forwards: 70
From: sip:[email protected];tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: sip:[email protected]
Contact: <sip:[email protected]:5060;ob>
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
CSeq: 7865 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: quvp3co73havl1r5qns1hk5telp3opb
secHdrB: 9D368D6908B582393DA37B75C80CB698
Content-Type: application/sdp
Content-Length:   476

v=0
o=- 3788455638 3788455638 IN IP4 192.168.2.139
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 192.168.2.139
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.139
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

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (1183 bytes) from UDP:192.168.2.139:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.139:5060;rport;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Max-Forwards: 70
From: sip:[email protected];tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: sip:[email protected]
Contact: <sip:[email protected]:5060;ob>
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
CSeq: 7865 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: quvp3co73havl1r5qns1hk5telp3opb
secHdrB: 9D368D6908B582393DA37B75C80CB698
Content-Type: application/sdp
Content-Length:   476

v=0
o=- 3788455638 3788455638 IN IP4 192.168.2.139
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 192.168.2.139
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.139
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

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (1183 bytes) from UDP:192.168.2.139:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.139:5060;rport;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Max-Forwards: 70
From: sip:[email protected];tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: sip:[email protected]
Contact: <sip:[email protected]:5060;ob>
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
CSeq: 7865 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: quvp3co73havl1r5qns1hk5telp3opb
secHdrB: 9D368D6908B582393DA37B75C80CB698
Content-Type: application/sdp
Content-Length:   476

v=0
o=- 3788455638 3788455638 IN IP4 192.168.2.139
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 192.168.2.139
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.139
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

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (357 bytes) from UDP:192.168.2.34:5065 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5065;branch=z9hG4bK.adItM7S3q;rport
From: <sip:[email protected]>;tag=Zr28~or
To: <sip:[email protected]>;tag=552bfc74-60e4-4dcc-a615-8f94586a3ffa
CSeq: 111 BYE
Call-ID: 5e6cb4df-7a0c-46b1-96f3-6c80cf6e1441
Max-Forwards: 70
User-Agent: Linphonec/3.12.0 (belle-sip/1.6.3)


<--- Transmitting SIP response (354 bytes) to UDP:192.168.2.34:5065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.34:5065;rport=5065;received=192.168.2.34;branch=z9hG4bK.adItM7S3q
Call-ID: 5e6cb4df-7a0c-46b1-96f3-6c80cf6e1441
From: <sip:[email protected]>;tag=Zr28~or
To: <sip:[email protected]>;tag=552bfc74-60e4-4dcc-a615-8f94586a3ffa
CSeq: 111 BYE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length:  0


<--- Received SIP request (1183 bytes) from UDP:192.168.2.139:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.139:5060;rport;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Max-Forwards: 70
From: sip:[email protected];tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: sip:[email protected]
Contact: <sip:[email protected]:5060;ob>
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
CSeq: 7865 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: quvp3co73havl1r5qns1hk5telp3opb
secHdrB: 9D368D6908B582393DA37B75C80CB698
Content-Type: application/sdp
Content-Length:   476

v=0
o=- 3788455638 3788455638 IN IP4 192.168.2.139
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 192.168.2.139
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.139
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

<--- Transmitting SIP response (917 bytes) to UDP:192.168.2.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.139:5060;rport=5060;received=192.168.2.139;branch=z9hG4bKPjqhSIcUl0DyrbKL0lh3ixQ3GT32O1wPsF
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
From: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
To: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
CSeq: 7865 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.34:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 3788455638 3788455640 IN IP4 192.168.2.34
s=Asterisk
c=IN IP4 192.168.2.34
t=0 0
m=audio 10272 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (430 bytes) to UDP:192.168.2.139:5060 --->
BYE sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;rport;branch=z9hG4bKPj216a8318-8cbf-4656-a58a-5d8e98870b7e
From: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
To: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
CSeq: 7781 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length:  0


<--- Transmitting SIP request (430 bytes) to UDP:192.168.2.139:5060 --->
BYE sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;rport;branch=z9hG4bKPj216a8318-8cbf-4656-a58a-5d8e98870b7e
From: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
To: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
CSeq: 7781 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length:  0


<--- Transmitting SIP request (430 bytes) to UDP:192.168.2.139:5060 --->
BYE sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;rport;branch=z9hG4bKPj216a8318-8cbf-4656-a58a-5d8e98870b7e
From: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
To: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
CSeq: 7781 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length:  0


<--- Transmitting SIP request (430 bytes) to UDP:192.168.2.139:5060 --->
BYE sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;rport;branch=z9hG4bKPj216a8318-8cbf-4656-a58a-5d8e98870b7e
From: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
To: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
CSeq: 7781 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length:  0


<--- Transmitting SIP request (430 bytes) to UDP:192.168.2.139:5060 --->
BYE sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;rport;branch=z9hG4bKPj216a8318-8cbf-4656-a58a-5d8e98870b7e
From: <sip:[email protected]>;tag=b37978d0-e8ed-450f-863c-576d94af2050
To: <sip:[email protected]>;tag=Xy9slzntm583wOxjuCKObzsGaGJ3Pmm4
Call-ID: 42Ym4xZYkX0dJ9Z5oc7yEHqUKHi9x5pL
CSeq: 7781 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length:  0
 
  • Like
Reaktionen: Rosen21
Wenn Du mal einen ordentlichen Trace mit tcpdump (der auch die Zeitschiene mitnimmt) machen würdest und das dann im Wireshark grafisch auswerten würdest, würdest Du mit hoher Wahrscheinlichkeit sehen, dass es so ist, wie ich schonmal geschrieben habe. Entweder kommen die Pakete von Asterisk bei 2.139 nicht an oder der verwirft sie (aus welchen Gründen auch immer). Den Ping hast Du auch noch nicht gemacht. Du musst als aller erstes sicherstellen, dass Dein Netz nachweislich vollständig so funktioniert, wie es soll. Macht keinen Sinn hier, auf dieser Basis weiterzumachen. Was Paketfilter angeht: iptables ist das Stichwort.
 
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