*CLI> sip set debug on 
SIP Debugging enabled
*CLI> [Apr 20 08:46:18] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SUBSCRIBE sip:85@asterisk.volker-sauer.de;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-gxtc4fsi638c;rport
From: <sip:snomvs@asterisk.volker-sauer.de>;tag=8t8yucyvon
To: <sip:85@asterisk.volker-sauer.de;user=phone>
Call-ID: 3cdeacb7bb7d-bq69dm3pa2p4
CSeq: 64437 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:snomvs@192.168.0.14:5060>;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
User-Agent: snom370/8.4.31
Expires: 3600
Content-Length: 0

<------------->
[Apr 20 08:46:18] --- (13 headers 0 lines) ---
[Apr 20 08:46:18] Creating new subscription
[Apr 20 08:46:18] Sending to 192.168.0.14:5060 (NAT)
[Apr 20 08:46:18] list_route: hop: <sip:snomvs@192.168.0.14:5060>
[Apr 20 08:46:18] Found peer 'snomvs' for 'snomvs' from 192.168.0.14:5060
[Apr 20 08:46:18] Looking for 85 in incoming (domain asterisk.volker-sauer.de)
[Apr 20 08:46:18] Scheduling destruction of SIP dialog '3cdeacb7bb7d-bq69dm3pa2p4' in 310000 ms (Method: SUBSCRIBE)
[Apr 20 08:46:18] 
<--- Transmitting (NAT) to 192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-gxtc4fsi638c;received=192.168.0.14;rport=5060
From: <sip:snomvs@asterisk.volker-sauer.de>;tag=8t8yucyvon
To: <sip:85@asterisk.volker-sauer.de;user=phone>;tag=as49871c93
Call-ID: 3cdeacb7bb7d-bq69dm3pa2p4
CSeq: 64437 SUBSCRIBE
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:85@192.168.0.1:5060>;expires=300
Content-Length: 0


<------------>
[Apr 20 08:46:18] set_destination: Parsing <sip:snomvs@192.168.0.14:5060> for address/port to send to
[Apr 20 08:46:18] set_destination: set destination to 192.168.0.14:5060
[Apr 20 08:46:18] Reliably Transmitting (NAT) to 192.168.0.14:5060:
NOTIFY sip:snomvs@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK51ce4b94;rport
Max-Forwards: 70
From: <sip:85@asterisk.volker-sauer.de;user=phone>;tag=as49871c93
To: <sip:snomvs@asterisk.volker-sauer.de>;tag=8t8yucyvon
Contact: <sip:85@192.168.0.1:5060>
Call-ID: 3cdeacb7bb7d-bq69dm3pa2p4
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 10.3.0
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 213

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:85@asterisk.volker-sauer.de">
<dialog id="85">
<state>terminated</state>
</dialog>
</dialog-info>

---
[Apr 20 08:46:18] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK51ce4b94;rport=5060
From: <sip:85@asterisk.volker-sauer.de;user=phone>;tag=as49871c93
To: <sip:snomvs@asterisk.volker-sauer.de>;tag=8t8yucyvon
Call-ID: 3cdeacb7bb7d-bq69dm3pa2p4
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
[Apr 20 08:46:18] --- (7 headers 0 lines) ---
[Apr 20 08:46:18] 
<--- SIP read from UDP:192.168.0.13:5060 --->
SUBSCRIBE sip:58@asterisk.volker-sauer.de;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK-9cq5lexoui4e;rport
From: <sip:snomhmk@asterisk.volker-sauer.de>;tag=2b5r6j6322
To: <sip:58@asterisk.volker-sauer.de;user=phone>
Call-ID: 3cf0bcded8b5-1wzi4l6d7rku
CSeq: 70767 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:snomhmk@192.168.0.13:5060>;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
User-Agent: snom320/8.4.31
Expires: 3600
Content-Length: 0

<------------->
[Apr 20 08:46:18] --- (13 headers 0 lines) ---
[Apr 20 08:46:18] Creating new subscription
[Apr 20 08:46:18] Sending to 192.168.0.13:5060 (NAT)
[Apr 20 08:46:18] list_route: hop: <sip:snomhmk@192.168.0.13:5060>
[Apr 20 08:46:18] Found peer 'snomhmk' for 'snomhmk' from 192.168.0.13:5060
[Apr 20 08:46:18] Looking for 58 in incoming (domain asterisk.volker-sauer.de)
[Apr 20 08:46:18] Scheduling destruction of SIP dialog '3cf0bcded8b5-1wzi4l6d7rku' in 310000 ms (Method: SUBSCRIBE)
[Apr 20 08:46:18] 
<--- Transmitting (NAT) to 192.168.0.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK-9cq5lexoui4e;received=192.168.0.13;rport=5060
From: <sip:snomhmk@asterisk.volker-sauer.de>;tag=2b5r6j6322
To: <sip:58@asterisk.volker-sauer.de;user=phone>;tag=as68203e0b
Call-ID: 3cf0bcded8b5-1wzi4l6d7rku
CSeq: 70767 SUBSCRIBE
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:58@192.168.0.1:5060>;expires=300
Content-Length: 0


<------------>
[Apr 20 08:46:18] set_destination: Parsing <sip:snomhmk@192.168.0.13:5060> for address/port to send to
[Apr 20 08:46:18] set_destination: set destination to 192.168.0.13:5060
[Apr 20 08:46:18] Reliably Transmitting (NAT) to 192.168.0.13:5060:
NOTIFY sip:snomhmk@192.168.0.13:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK26faac4b;rport
Max-Forwards: 70
From: <sip:58@asterisk.volker-sauer.de;user=phone>;tag=as68203e0b
To: <sip:snomhmk@asterisk.volker-sauer.de>;tag=2b5r6j6322
Contact: <sip:58@192.168.0.1:5060>
Call-ID: 3cf0bcded8b5-1wzi4l6d7rku
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 10.3.0
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 213

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:58@asterisk.volker-sauer.de">
<dialog id="58">
<state>terminated</state>
</dialog>
</dialog-info>

---
[Apr 20 08:46:18] 
<--- SIP read from UDP:192.168.0.13:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK26faac4b;rport=5060
From: <sip:58@asterisk.volker-sauer.de;user=phone>;tag=as68203e0b
To: <sip:snomhmk@asterisk.volker-sauer.de>;tag=2b5r6j6322
Call-ID: 3cf0bcded8b5-1wzi4l6d7rku
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
[Apr 20 08:46:18] --- (7 headers 0 lines) ---
[Apr 20 08:46:20] 
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:asterisk.volker-sauer.de SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-re61x4;rport
From: "SauerKraut (81)" <sip:snomm9-1@asterisk.volker-sauer.de>;tag=0voien
To: "SauerKraut (81)" <sip:snomm9-1@asterisk.volker-sauer.de>
Call-ID: c0phb047@snom
CSeq: 32469 REGISTER
Max-Forwards: 70
Contact: <sip:snomm9-1@192.168.0.10:5060;transport=udp;line=h8x4r1>;reg-id=1;+sip.instance="<urn:uuid:572d1aa1-bfd5-4b8a-ab1e-f9095df386e5>"
Supported: path, outbound, gruu
User-Agent: snom-m9/9.5.14-a
Expires: 3600
Content-Length: 0

<------------->
[Apr 20 08:46:20] --- (12 headers 0 lines) ---
[Apr 20 08:46:20] Sending to 192.168.0.10:5060 (NAT)
[Apr 20 08:46:20] Reliably Transmitting (NAT) to 192.168.0.10:5060:
OPTIONS sip:snomm9-1@192.168.0.10:5060;transport=udp;line=h8x4r1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7a19e40e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as667ba5db
To: <sip:snomm9-1@192.168.0.10:5060;transport=udp;line=h8x4r1>
Contact: <sip:asterisk@192.168.0.1:5060>
Call-ID: 0cd11148631404dc2c1355520f66662b@volker-sauer.de
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 20 Apr 2012 06:46:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Apr 20 08:46:20] 
<--- Transmitting (NAT) to 192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-re61x4;received=192.168.0.10;rport=5060
From: "SauerKraut (81)" <sip:snomm9-1@asterisk.volker-sauer.de>;tag=0voien
To: "SauerKraut (81)" <sip:snomm9-1@asterisk.volker-sauer.de>;tag=as6e8e606a
Call-ID: c0phb047@snom
CSeq: 32469 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:snomm9-1@192.168.0.10:5060;transport=udp;line=h8x4r1>;expires=300
Date: Fri, 20 Apr 2012 06:46:20 GMT
Content-Length: 0


<------------>
[Apr 20 08:46:20] Scheduling destruction of SIP dialog 'c0phb047@snom' in 32000 ms (Method: REGISTER)
[Apr 20 08:46:20] 
<--- SIP read from UDP:192.168.0.10:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7a19e40e;rport=5060
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as667ba5db
To: <sip:snomm9-1@192.168.0.10:5060;transport=udp;line=h8x4r1>
Call-ID: 0cd11148631404dc2c1355520f66662b@volker-sauer.de
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Supported: 100rel, replaces
Content-Length: 0

<------------->
[Apr 20 08:46:20] --- (9 headers 0 lines) ---
[Apr 20 08:46:20] Really destroying SIP dialog '0cd11148631404dc2c1355520f66662b@volker-sauer.de' Method: OPTIONS
[Apr 20 08:46:20] 
<--- SIP read from UDP:192.168.0.10:5060 --->
REGISTER sip:asterisk.volker-sauer.de SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-n4mqbg;rport
From: "Firma DECT2 (85)" <sip:snomm9-2@asterisk.volker-sauer.de>;tag=e6xz68
To: "Firma DECT2 (85)" <sip:snomm9-2@asterisk.volker-sauer.de>
Call-ID: pxl6qlv7@snom
CSeq: 31108 REGISTER
Max-Forwards: 70
Contact: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>;reg-id=1;+sip.instance="<urn:uuid:f889131d-819f-4c2f-a7a3-4a0c5f5d15e8>"
Supported: path, outbound, gruu
User-Agent: snom-m9/9.5.14-a
Expires: 3600
Content-Length: 0

<------------->
[Apr 20 08:46:20] --- (12 headers 0 lines) ---
[Apr 20 08:46:20] Sending to 192.168.0.10:5060 (NAT)
[Apr 20 08:46:20] Reliably Transmitting (NAT) to 192.168.0.10:5060:
OPTIONS sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7352f255;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as417cf300
To: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>
Contact: <sip:asterisk@192.168.0.1:5060>
Call-ID: 172de1603cf2b38445ad4b720c0055d0@volker-sauer.de
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 20 Apr 2012 06:46:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Apr 20 08:46:20] 
<--- Transmitting (NAT) to 192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-n4mqbg;received=192.168.0.10;rport=5060
From: "Firma DECT2 (85)" <sip:snomm9-2@asterisk.volker-sauer.de>;tag=e6xz68
To: "Firma DECT2 (85)" <sip:snomm9-2@asterisk.volker-sauer.de>;tag=as309c88c8
Call-ID: pxl6qlv7@snom
CSeq: 31108 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>;expires=300
Date: Fri, 20 Apr 2012 06:46:20 GMT
Content-Length: 0


<------------>
[Apr 20 08:46:20] Scheduling destruction of SIP dialog 'pxl6qlv7@snom' in 32000 ms (Method: REGISTER)
[Apr 20 08:46:20] 
<--- SIP read from UDP:192.168.0.10:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7352f255;rport=5060
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as417cf300
To: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>
Call-ID: 172de1603cf2b38445ad4b720c0055d0@volker-sauer.de
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Supported: 100rel, replaces
Content-Length: 0

<------------->
[Apr 20 08:46:20] --- (9 headers 0 lines) ---
[Apr 20 08:46:20] Really destroying SIP dialog '172de1603cf2b38445ad4b720c0055d0@volker-sauer.de' Method: OPTIONS
[Apr 20 08:46:22] 
<--- SIP read from UDP:217.10.79.9:5060 --->
INVITE sip:2216950e1@130.83.208.238:5060 SIP/2.0
Record-Route: <sip:217.10.79.9;lr;ftag=as026b057e>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr;ftag=as026b057e>
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK68c3.6651ffd1.0
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK68c3.6651ffd1.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK0ebe7890
Via: SIP/2.0/UDP 217.10.67.13:5060;received=217.10.67.13;branch=z9hG4bK0ebe7890;rport=5060
Max-Forwards: 67
From: "01796901475" <sip:01796901475@sipgate.de>;tag=as026b057e
To: <sip:00496151154260@sipgate.de>
Contact: <sip:01796901475@217.10.67.13>
Call-ID: 2aa296b60bf3ebb91dea752762df8a63@sipgate.de
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 426

v=0
o=root 833857671 833857671 IN IP4 217.10.67.13
s=sipgate VoIP GW
c=IN IP4 217.10.67.13
t=0 0
m=audio 18476 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
[Apr 20 08:46:22] --- (18 headers 19 lines) ---
[Apr 20 08:46:22] Sending to 217.10.79.9:5060 (NAT)
[Apr 20 08:46:22] Using INVITE request as basis request - 2aa296b60bf3ebb91dea752762df8a63@sipgate.de
[Apr 20 08:46:22] Found peer 'sipgate-gmbh-e0' for '01796901475' from 217.10.79.9:5060
[Apr 20 08:46:22]   == Using SIP RTP CoS mark 5
[Apr 20 08:46:22] Found RTP audio format 8
[Apr 20 08:46:22] Found RTP audio format 0
[Apr 20 08:46:22] Found RTP audio format 3
[Apr 20 08:46:22] Found RTP audio format 97
[Apr 20 08:46:22] Found RTP audio format 18
[Apr 20 08:46:22] Found RTP audio format 112
[Apr 20 08:46:22] Found RTP audio format 101
[Apr 20 08:46:22] Found audio description format PCMA for ID 8
[Apr 20 08:46:22] Found audio description format PCMU for ID 0
[Apr 20 08:46:22] Found audio description format GSM for ID 3
[Apr 20 08:46:22] Found audio description format iLBC for ID 97
[Apr 20 08:46:22] Found audio description format G729 for ID 18
[Apr 20 08:46:22] Found audio description format G726-32 for ID 112
[Apr 20 08:46:22] Found audio description format telephone-event for ID 101
[Apr 20 08:46:22] Capabilities: us - (gsm|ulaw|alaw|g726|adpcm|g722), peer - audio=(gsm|ulaw|alaw|g726|g729|ilbc)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g726)
[Apr 20 08:46:22] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Apr 20 08:46:22] Peer audio RTP is at port 217.10.67.13:18476
[Apr 20 08:46:22] Looking for 2216950e1 in incoming (domain 130.83.208.238)
[Apr 20 08:46:22] list_route: hop: <sip:217.10.79.9;lr;ftag=as026b057e>
[Apr 20 08:46:22] list_route: hop: <sip:172.20.40.2;lr=on>
[Apr 20 08:46:22] list_route: hop: <sip:217.10.79.9;lr;ftag=as026b057e>
[Apr 20 08:46:22] 
<--- Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK68c3.6651ffd1.0;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK68c3.6651ffd1.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK0ebe7890
Via: SIP/2.0/UDP 217.10.67.13:5060;received=217.10.67.13;branch=z9hG4bK0ebe7890;rport=5060
Record-Route: <sip:217.10.79.9;lr;ftag=as026b057e>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr;ftag=as026b057e>
From: "01796901475" <sip:01796901475@sipgate.de>;tag=as026b057e
To: <sip:00496151154260@sipgate.de>
Call-ID: 2aa296b60bf3ebb91dea752762df8a63@sipgate.de
CSeq: 102 INVITE
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2216950e1@130.83.208.238:5060>
Content-Length: 0


<------------>
[Apr 20 08:46:22]     -- Executing [2216950e1@incoming:1] Goto("SIP/sipgate-gmbh-e0-00000000", "83,510") in new stack
[Apr 20 08:46:22]     -- Goto (incoming,83,510)
[Apr 20 08:46:22]     -- Executing [83@incoming:510] Macro("SIP/sipgate-gmbh-e0-00000000", "CallExt2Int,SIP/snomvs831&IAX2/vsauer,V1:,,volker@volker-sauer.de") in new stack
[Apr 20 08:46:22]     -- Executing [s@macro-CallExt2Int:1] AGI("SIP/sipgate-gmbh-e0-00000000", "telefonbuch_reverselookup.php,01796901475") in new stack
[Apr 20 08:46:22]     -- Launched AGI Script /opt/asterisk/var/lib/asterisk/agi-bin/telefonbuch_reverselookup.php
[Apr 20 08:46:22] ERROR[19435]: utils.c:1170 ast_carefulwrite: write() returned error: Broken pipe
[Apr 20 08:46:22]     -- <SIP/sipgate-gmbh-e0-00000000>AGI Script telefonbuch_reverselookup.php completed, returning 0
[Apr 20 08:46:22]     -- Executing [s@macro-CallExt2Int:2] Set("SIP/sipgate-gmbh-e0-00000000", "PICKUPMARK=83") in new stack
[Apr 20 08:46:22]     -- Executing [s@macro-CallExt2Int:3] NoOp("SIP/sipgate-gmbh-e0-00000000", "83") in new stack
[Apr 20 08:46:22]     -- Executing [s@macro-CallExt2Int:4] Set("SIP/sipgate-gmbh-e0-00000000", "VOLUME(TX)=") in new stack
[Apr 20 08:46:22]     -- Executing [s@macro-CallExt2Int:5] Set("SIP/sipgate-gmbh-e0-00000000", "VOLUME(TX)=") in new stack
[Apr 20 08:46:22]     -- Executing [s@macro-CallExt2Int:6] Set("SIP/sipgate-gmbh-e0-00000000", "CALLERID(name)=V1:Sauer,Volker(cell)") in new stack
[Apr 20 08:46:22]     -- Executing [s@macro-CallExt2Int:7] JabberSend("SIP/sipgate-gmbh-e0-00000000", "asterisk,volker@volker-sauer.de,Sie wurden angerufen von "V1:Sauer,Volker(cell)" <01796901475> um Fri Apr 20 08:46:22 2012") in new stack
[Apr 20 08:46:22]     -- Executing [s@macro-CallExt2Int:8] Dial("SIP/sipgate-gmbh-e0-00000000", "SIP/snomvs831&IAX2/vsauer,,kKtTwW") in new stack
[Apr 20 08:46:22]   == Using SIP RTP CoS mark 5
[Apr 20 08:46:22] Audio is at 10068
[Apr 20 08:46:22] Adding codec 100003 (ulaw) to SDP
[Apr 20 08:46:22] Adding codec 100012 (g722) to SDP
[Apr 20 08:46:22] Adding codec 100004 (alaw) to SDP
[Apr 20 08:46:22] Adding codec 100002 (gsm) to SDP
[Apr 20 08:46:22] Adding codec 100011 (g726) to SDP
[Apr 20 08:46:22] Adding codec 100006 (adpcm) to SDP
[Apr 20 08:46:22] Adding non-codec 0x1 (telephone-event) to SDP
[Apr 20 08:46:22] Reliably Transmitting (NAT) to 192.168.0.14:5060:
INVITE sip:snomvs831@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK3aff5344;rport
Max-Forwards: 70
From: "V1:Sauer,Volker(cell)" <sip:01796901475@volker-sauer.de>;tag=as45bbf9d9
To: <sip:snomvs831@192.168.0.14:5060>
Contact: <sip:01796901475@192.168.0.1:5060>
Call-ID: 0c958bd1483264e6128f691b77598256@volker-sauer.de
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 20 Apr 2012 06:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 359

v=0
o=root 2064002393 2064002393 IN IP4 192.168.0.1
s=Asterisk PBX 10.3.0
c=IN IP4 192.168.0.1
t=0 0
m=audio 10068 RTP/AVP 0 9 8 3 111 5 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Apr 20 08:46:22]     -- Called SIP/snomvs831
[Apr 20 08:46:22] WARNING[19435]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
[Apr 20 08:46:22] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK3aff5344;rport=5060
From: "V1:Sauer,Volker(cell)" <sip:01796901475@volker-sauer.de>;tag=as45bbf9d9
To: "V.Sauer Genion" <sip:snomvs831@192.168.0.14:5060>;tag=4edds85r39
Call-ID: 0c958bd1483264e6128f691b77598256@volker-sauer.de
CSeq: 102 INVITE
Contact: <sip:85@asterisk.volker-sauer.de;user=phone>
Diversion: <sip:snomvs831@192.168.0.14:5060>;reason="unconditional"
Content-Length: 0

<------------->
[Apr 20 08:46:22] --- (9 headers 0 lines) ---
[Apr 20 08:46:22]     -- Got SIP response 302 "Moved Temporarily" back from 192.168.0.14:5060
[Apr 20 08:46:22] RDNIS for this call is snomvs831 (reason "unconditional)
[Apr 20 08:46:22] Transmitting (NAT) to 192.168.0.14:5060:
ACK sip:snomvs831@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK3aff5344;rport
Max-Forwards: 70
From: "V1:Sauer,Volker(cell)" <sip:01796901475@volker-sauer.de>;tag=as45bbf9d9
To: <sip:snomvs831@192.168.0.14:5060>;tag=4edds85r39
Contact: <sip:01796901475@192.168.0.1:5060>
Call-ID: 0c958bd1483264e6128f691b77598256@volker-sauer.de
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.3.0
Content-Length: 0


---
[Apr 20 08:46:22]     -- Now forwarding SIP/sipgate-gmbh-e0-00000000 to 'Local/85@doLocalCallsPrivat' (thanks to SIP/snomvs831-00000001)
[Apr 20 08:46:22] NOTICE[19435]: app_dial.c:883 do_forward: Not accepting call completion offers from call-forward recipient Local/85@doLocalCallsPrivat-99d3;1
[Apr 20 08:46:22] 
<--- Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 181 Call is being forwarded
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK68c3.6651ffd1.0;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK68c3.6651ffd1.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK0ebe7890
Via: SIP/2.0/UDP 217.10.67.13:5060;received=217.10.67.13;branch=z9hG4bK0ebe7890;rport=5060
Record-Route: <sip:217.10.79.9;lr;ftag=as026b057e>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr;ftag=as026b057e>
From: "01796901475" <sip:01796901475@sipgate.de>;tag=as026b057e
To: <sip:00496151154260@sipgate.de>;tag=as5a1d63af
Call-ID: 2aa296b60bf3ebb91dea752762df8a63@sipgate.de
CSeq: 102 INVITE
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2216950e1@130.83.208.238:5060>
Diversion: <sip:snomvs831@130.83.208.238>;reason=unknown
Content-Length: 0


<------------>
[Apr 20 08:46:22]     -- Executing [85@doLocalCallsPrivat:1] NoOp("Local/85@doLocalCallsPrivat-99d3;2", "Local/85@d") in new stack
[Apr 20 08:46:22]     -- Executing [85@doLocalCallsPrivat:2] Set("Local/85@doLocalCallsPrivat-99d3;2", "VOLUME(TX)=") in new stack
[Apr 20 08:46:22]     -- Executing [85@doLocalCallsPrivat:3] Goto("Local/85@doLocalCallsPrivat-99d3;2", "doLocalCallsGeneric,85,1") in new stack
[Apr 20 08:46:22]     -- Goto (doLocalCallsGeneric,85,1)
[Apr 20 08:46:22] NOTICE[19438]: ast_expr2.y:760 compose_func_args: argbuf allocated 12 bytes;
[Apr 20 08:46:22] NOTICE[19438]: ast_expr2.y:779 compose_func_args: argbuf uses 11 bytes;
[Apr 20 08:46:22]     -- Executing [85@doLocalCallsGeneric:1] Set("Local/85@doLocalCallsPrivat-99d3;2", "CALLERID(num)=01796901475") in new stack
[Apr 20 08:46:22]     -- Executing [85@doLocalCallsGeneric:2] Goto("Local/85@doLocalCallsPrivat-99d3;2", "incoming,85,100") in new stack
[Apr 20 08:46:22]     -- Goto (incoming,85,100)
[Apr 20 08:46:22]     -- Executing [85@incoming:100] Macro("Local/85@doLocalCallsPrivat-99d3;2", "CallInt2Int,SIP/snomm9-2") in new stack
[Apr 20 08:46:22]     -- Executing [s@macro-CallInt2Int:1] Set("Local/85@doLocalCallsPrivat-99d3;2", "PICKUPMARK=85") in new stack
[Apr 20 08:46:22]     -- Executing [s@macro-CallInt2Int:2] NoOp("Local/85@doLocalCallsPrivat-99d3;2", "85") in new stack
[Apr 20 08:46:22]     -- Executing [s@macro-CallInt2Int:3] Set("Local/85@doLocalCallsPrivat-99d3;2", "VOLUME(TX)=") in new stack
[Apr 20 08:46:22]     -- Executing [s@macro-CallInt2Int:4] Set("Local/85@doLocalCallsPrivat-99d3;2", "VOLUME(TX)=-4") in new stack
[Apr 20 08:46:22]     -- Executing [s@macro-CallInt2Int:5] Dial("Local/85@doLocalCallsPrivat-99d3;2", "SIP/snomm9-2,,kKwWtTj") in new stack
[Apr 20 08:46:22]   == Using SIP RTP CoS mark 5
[Apr 20 08:46:22] Really destroying SIP dialog '0c958bd1483264e6128f691b77598256@volker-sauer.de' Method: INVITE
[Apr 20 08:46:22] set_destination: Parsing <sip:snomvs@192.168.0.14:5060> for address/port to send to
[Apr 20 08:46:22] set_destination: set destination to 192.168.0.14:5060
[Apr 20 08:46:22] Audio is at 10060
[Apr 20 08:46:22] Adding codec 100003 (ulaw) to SDP
[Apr 20 08:46:22] Adding codec 100012 (g722) to SDP
[Apr 20 08:46:22] Adding codec 100004 (alaw) to SDP
[Apr 20 08:46:22] Adding codec 100002 (gsm) to SDP
[Apr 20 08:46:22] Adding codec 100011 (g726) to SDP
[Apr 20 08:46:22] Adding codec 100006 (adpcm) to SDP
[Apr 20 08:46:22] Adding non-codec 0x1 (telephone-event) to SDP
[Apr 20 08:46:22] Reliably Transmitting (NAT) to 192.168.0.10:5060:
INVITE sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7ae207d7;rport
Max-Forwards: 70
From: "V1:Sauer,Volker(cell)" <sip:01796901475@volker-sauer.de>;tag=as44e1ac96
To: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>
Contact: <sip:01796901475@192.168.0.1:5060>
Call-ID: 4ea097e456e2746a6fec07ef20c5c1b8@volker-sauer.de
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 20 Apr 2012 06:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Diversion: <sip:snomvs831@192.168.0.1>;reason=unknown
Content-Type: application/sdp
Content-Length: 359

v=0
o=root 1459069130 1459069130 IN IP4 192.168.0.1
s=Asterisk PBX 10.3.0
c=IN IP4 192.168.0.1
t=0 0
m=audio 10060 RTP/AVP 0 9 8 3 111 5 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Apr 20 08:46:22]     -- Called SIP/snomm9-2
[Apr 20 08:46:22] Reliably Transmitting (NAT) to 192.168.0.14:5060:
NOTIFY sip:snomvs@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK35a6c3a2;rport
Max-Forwards: 70
From: <sip:85@asterisk.volker-sauer.de;user=phone>;tag=as49871c93
To: <sip:snomvs@asterisk.volker-sauer.de>;tag=8t8yucyvon
Contact: <sip:85@192.168.0.1:5060>
Call-ID: 3cdeacb7bb7d-bq69dm3pa2p4
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 10.3.0
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 608

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" state="full" entity="sip:85@asterisk.volker-sauer.de">
<dialog id="85" call-id="pickup-3cdeacb7bb7d-bq69dm3pa2p4" local-tag="8t8yucyvon" remote-tag="as49871c93" direction="recipient">
<remote>
<identity display="V1:Sauer,Volker(cell)">sip:01796901475@asterisk.volker-sauer.de</identity>
<target uri="sip:01796901475@asterisk.volker-sauer.de"/>
</remote>
<local>
<identity>sip:85@asterisk.volker-sauer.de</identity>
<target uri="sip:85@asterisk.volker-sauer.de"/>
</local>
<state>early</state>
</dialog>
</dialog-info>

---
[Apr 20 08:46:22]   == Extension Changed 85[incoming] new state Ringing for Notify User snomvs 
[Apr 20 08:46:22] 
<--- SIP read from UDP:192.168.0.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7ae207d7;rport=5060
From: "V1:Sauer,Volker(cell)" <sip:01796901475@volker-sauer.de>;tag=as44e1ac96
To: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>;tag=xde12r
Call-ID: 4ea097e456e2746a6fec07ef20c5c1b8@volker-sauer.de
CSeq: 102 INVITE
Contact: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>
Supported: 100rel, replaces, norefersub
Content-Length: 0

<------------->
[Apr 20 08:46:22] --- (9 headers 0 lines) ---
[Apr 20 08:46:22] 
<--- SIP read from UDP:192.168.0.10:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7ae207d7;rport=5060
From: "V1:Sauer,Volker(cell)" <sip:01796901475@volker-sauer.de>;tag=as44e1ac96
To: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>;tag=xde12r
Call-ID: 4ea097e456e2746a6fec07ef20c5c1b8@volker-sauer.de
CSeq: 102 INVITE
Contact: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>
Supported: 100rel, replaces, norefersub
Content-Length: 0

<------------->
[Apr 20 08:46:22] --- (9 headers 0 lines) ---
[Apr 20 08:46:22] list_route: hop: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>
[Apr 20 08:46:22]     -- SIP/snomm9-2-00000002 is ringing
[Apr 20 08:46:22]     -- Local/85@doLocalCallsPrivat-99d3;1 is ringing
[Apr 20 08:46:22] 
<--- Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK68c3.6651ffd1.0;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK68c3.6651ffd1.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK0ebe7890
Via: SIP/2.0/UDP 217.10.67.13:5060;received=217.10.67.13;branch=z9hG4bK0ebe7890;rport=5060
Record-Route: <sip:217.10.79.9;lr;ftag=as026b057e>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr;ftag=as026b057e>
From: "01796901475" <sip:01796901475@sipgate.de>;tag=as026b057e
To: <sip:00496151154260@sipgate.de>;tag=as5a1d63af
Call-ID: 2aa296b60bf3ebb91dea752762df8a63@sipgate.de
CSeq: 102 INVITE
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2216950e1@130.83.208.238:5060>
Content-Length: 0


<------------>
[Apr 20 08:46:22] Retransmitting #1 (NAT) to 192.168.0.14:5060:
NOTIFY sip:snomvs@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK35a6c3a2;rport
Max-Forwards: 70
From: <sip:85@asterisk.volker-sauer.de;user=phone>;tag=as49871c93
To: <sip:snomvs@asterisk.volker-sauer.de>;tag=8t8yucyvon
Contact: <sip:85@192.168.0.1:5060>
Call-ID: 3cdeacb7bb7d-bq69dm3pa2p4
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 10.3.0
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 608

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" state="full" entity="sip:85@asterisk.volker-sauer.de">
<dialog id="85" call-id="pickup-3cdeacb7bb7d-bq69dm3pa2p4" local-tag="8t8yucyvon" remote-tag="as49871c93" direction="recipient">
<remote>
<identity display="V1:Sauer,Volker(cell)">sip:01796901475@asterisk.volker-sauer.de</identity>
<target uri="sip:01796901475@asterisk.volker-sauer.de"/>
</remote>
<local>
<identity>sip:85@asterisk.volker-sauer.de</identity>
<target uri="sip:85@asterisk.volker-sauer.de"/>
</local>
<state>early</state>
</dialog>
</dialog-info>

---
[Apr 20 08:46:22] Retransmitting #2 (NAT) to 192.168.0.14:5060:
NOTIFY sip:snomvs@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK35a6c3a2;rport
Max-Forwards: 70
From: <sip:85@asterisk.volker-sauer.de;user=phone>;tag=as49871c93
To: <sip:snomvs@asterisk.volker-sauer.de>;tag=8t8yucyvon
Contact: <sip:85@192.168.0.1:5060>
Call-ID: 3cdeacb7bb7d-bq69dm3pa2p4
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 10.3.0
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 608

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" state="full" entity="sip:85@asterisk.volker-sauer.de">
<dialog id="85" call-id="pickup-3cdeacb7bb7d-bq69dm3pa2p4" local-tag="8t8yucyvon" remote-tag="as49871c93" direction="recipient">
<remote>
<identity display="V1:Sauer,Volker(cell)">sip:01796901475@asterisk.volker-sauer.de</identity>
<target uri="sip:01796901475@asterisk.volker-sauer.de"/>
</remote>
<local>
<identity>sip:85@asterisk.volker-sauer.de</identity>
<target uri="sip:85@asterisk.volker-sauer.de"/>
</local>
<state>early</state>
</dialog>
</dialog-info>

---
[Apr 20 08:46:23] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK35a6c3a2;rport=5060
From: <sip:85@asterisk.volker-sauer.de;user=phone>;tag=as49871c93
To: <sip:snomvs@asterisk.volker-sauer.de>;tag=8t8yucyvon
Call-ID: 3cdeacb7bb7d-bq69dm3pa2p4
CSeq: 103 NOTIFY
Content-Length: 0

<------------->
[Apr 20 08:46:23] --- (7 headers 0 lines) ---
[Apr 20 08:46:23] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK35a6c3a2;rport=5060
From: <sip:85@asterisk.volker-sauer.de;user=phone>;tag=as49871c93
To: <sip:snomvs@asterisk.volker-sauer.de>;tag=8t8yucyvon
Call-ID: 3cdeacb7bb7d-bq69dm3pa2p4
CSeq: 103 NOTIFY
Content-Length: 0

<------------->
[Apr 20 08:46:23] --- (7 headers 0 lines) ---
[Apr 20 08:46:23] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK35a6c3a2;rport=5060
From: <sip:85@asterisk.volker-sauer.de;user=phone>;tag=as49871c93
To: <sip:snomvs@asterisk.volker-sauer.de>;tag=8t8yucyvon
Call-ID: 3cdeacb7bb7d-bq69dm3pa2p4
CSeq: 103 NOTIFY
Content-Length: 0

<------------->
[Apr 20 08:46:23] --- (7 headers 0 lines) ---
[Apr 20 08:46:23] 
<--- SIP read from UDP:192.168.0.14:5060 --->
REGISTER sip:asterisk.volker-sauer.de SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-bt17zj7ew9iw;rport
From: "V.Sauer IT-0" <sip:snomvs832@asterisk.volker-sauer.de>;tag=ujnb7nfrh5
To: "V.Sauer IT-0" <sip:snomvs832@asterisk.volker-sauer.de>
Call-ID: 3c26702cb829-kb3ma5hrdj8n
CSeq: 161664 REGISTER
Max-Forwards: 70
Contact: <sip:snomvs832@192.168.0.14:5060>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:af47ae8c-91d3-44bc-9808-e3333ce6cdb8>";audio;mobility="fixed";duplex="full";description="snom370";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom370/8.4.31
Allow-Events: dialog
X-Real-IP: 192.168.0.14
Supported: path, gruu
Expires: 3600
Content-Length: 0

<------------->
[Apr 20 08:46:23] --- (14 headers 0 lines) ---
[Apr 20 08:46:23] Sending to 192.168.0.14:5060 (NAT)
[Apr 20 08:46:23] Reliably Transmitting (NAT) to 192.168.0.14:5060:
OPTIONS sip:snomvs832@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK2a509927;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as54fa3f10
To: <sip:snomvs832@192.168.0.14:5060>
Contact: <sip:asterisk@192.168.0.1:5060>
Call-ID: 37f0f54a26338c42783ff4890e45663b@volker-sauer.de
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 20 Apr 2012 06:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Apr 20 08:46:23] 
<--- Transmitting (NAT) to 192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-bt17zj7ew9iw;received=192.168.0.14;rport=5060
From: "V.Sauer IT-0" <sip:snomvs832@asterisk.volker-sauer.de>;tag=ujnb7nfrh5
To: "V.Sauer IT-0" <sip:snomvs832@asterisk.volker-sauer.de>;tag=as6aaf0554
Call-ID: 3c26702cb829-kb3ma5hrdj8n
CSeq: 161664 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:snomvs832@192.168.0.14:5060>;expires=300
Date: Fri, 20 Apr 2012 06:46:23 GMT
Content-Length: 0


<------------>
[Apr 20 08:46:23] Scheduling destruction of SIP dialog '3c26702cb829-kb3ma5hrdj8n' in 32000 ms (Method: REGISTER)
[Apr 20 08:46:23] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK2a509927;rport=5060
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as54fa3f10
To: <sip:snomvs832@192.168.0.14:5060>
Call-ID: 37f0f54a26338c42783ff4890e45663b@volker-sauer.de
CSeq: 102 OPTIONS
Contact: <sip:snomvs831@192.168.0.14:5060>;reg-id=1
User-Agent: snom370/8.4.31
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

<------------->
[Apr 20 08:46:23] --- (14 headers 0 lines) ---
[Apr 20 08:46:23] Really destroying SIP dialog '37f0f54a26338c42783ff4890e45663b@volker-sauer.de' Method: OPTIONS
[Apr 20 08:46:23] 
<--- SIP read from UDP:192.168.0.14:5060 --->
REGISTER sip:asterisk.volker-sauer.de SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-fglmehfgxy4z;rport
From: "V.Sauer CG" <sip:snomvs837@asterisk.volker-sauer.de>;tag=gtub0rxrh1
To: "V.Sauer CG" <sip:snomvs837@asterisk.volker-sauer.de>
Call-ID: 3c26702d8885-3rdu8px65a9f
CSeq: 161772 REGISTER
Max-Forwards: 70
Contact: <sip:snomvs837@192.168.0.14:5060>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:e8be6891-c8c8-4cc7-87f9-e1bd4e6ba74a>";audio;mobility="fixed";duplex="full";description="snom370";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom370/8.4.31
Allow-Events: dialog
X-Real-IP: 192.168.0.14
Supported: path, gruu
Expires: 3600
Content-Length: 0

<------------->
[Apr 20 08:46:23] --- (14 headers 0 lines) ---
[Apr 20 08:46:23] Sending to 192.168.0.14:5060 (NAT)
[Apr 20 08:46:23] Reliably Transmitting (NAT) to 192.168.0.14:5060:
OPTIONS sip:snomvs837@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK2fc2a88b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as663f6695
To: <sip:snomvs837@192.168.0.14:5060>
Contact: <sip:asterisk@192.168.0.1:5060>
Call-ID: 34bfaa2f75fb748460d21cd116f34c82@volker-sauer.de
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 20 Apr 2012 06:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Apr 20 08:46:23] 
<--- Transmitting (NAT) to 192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-fglmehfgxy4z;received=192.168.0.14;rport=5060
From: "V.Sauer CG" <sip:snomvs837@asterisk.volker-sauer.de>;tag=gtub0rxrh1
To: "V.Sauer CG" <sip:snomvs837@asterisk.volker-sauer.de>;tag=as29019b18
Call-ID: 3c26702d8885-3rdu8px65a9f
CSeq: 161772 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:snomvs837@192.168.0.14:5060>;expires=300
Date: Fri, 20 Apr 2012 06:46:23 GMT
Content-Length: 0


<------------>
[Apr 20 08:46:23] Scheduling destruction of SIP dialog '3c26702d8885-3rdu8px65a9f' in 32000 ms (Method: REGISTER)
[Apr 20 08:46:23] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK2fc2a88b;rport=5060
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as663f6695
To: <sip:snomvs837@192.168.0.14:5060>
Call-ID: 34bfaa2f75fb748460d21cd116f34c82@volker-sauer.de
CSeq: 102 OPTIONS
Contact: <sip:snomvs831@192.168.0.14:5060>;reg-id=1
User-Agent: snom370/8.4.31
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

<------------->
[Apr 20 08:46:23] --- (14 headers 0 lines) ---
[Apr 20 08:46:23] Really destroying SIP dialog '34bfaa2f75fb748460d21cd116f34c82@volker-sauer.de' Method: OPTIONS
[Apr 20 08:46:24] 
<--- SIP read from UDP:192.168.0.14:5060 --->
REGISTER sip:asterisk.volker-sauer.de SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-7vdmvu40tioz;rport
From: "V.Sauer Privat" <sip:snomvs@asterisk.volker-sauer.de>;tag=p3slu93tp4
To: "V.Sauer Privat" <sip:snomvs@asterisk.volker-sauer.de>
Call-ID: 3c26702ca38e-5nyybeqvaqtm
CSeq: 161728 REGISTER
Max-Forwards: 70
Contact: <sip:snomvs@192.168.0.14:5060>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:270905d9-94c3-41db-beac-5502ecec38c7>";audio;mobility="fixed";duplex="full";description="snom370";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom370/8.4.31
Allow-Events: dialog
X-Real-IP: 192.168.0.14
Supported: path, gruu
Expires: 3600
Content-Length: 0

<------------->
[Apr 20 08:46:24] --- (14 headers 0 lines) ---
[Apr 20 08:46:24] Sending to 192.168.0.14:5060 (NAT)
[Apr 20 08:46:24] Reliably Transmitting (NAT) to 192.168.0.14:5060:
OPTIONS sip:snomvs@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK3cbf267c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as751b74df
To: <sip:snomvs@192.168.0.14:5060>
Contact: <sip:asterisk@192.168.0.1:5060>
Call-ID: 29fefa2422250d207319404a75fea06c@volker-sauer.de
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 20 Apr 2012 06:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Apr 20 08:46:24] 
<--- Transmitting (NAT) to 192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-7vdmvu40tioz;received=192.168.0.14;rport=5060
From: "V.Sauer Privat" <sip:snomvs@asterisk.volker-sauer.de>;tag=p3slu93tp4
To: "V.Sauer Privat" <sip:snomvs@asterisk.volker-sauer.de>;tag=as42a9d12a
Call-ID: 3c26702ca38e-5nyybeqvaqtm
CSeq: 161728 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:snomvs@192.168.0.14:5060>;expires=300
Date: Fri, 20 Apr 2012 06:46:24 GMT
Content-Length: 0


<------------>
[Apr 20 08:46:24] Scheduling destruction of SIP dialog '3c26702ca38e-5nyybeqvaqtm' in 32000 ms (Method: REGISTER)
[Apr 20 08:46:24] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK3cbf267c;rport=5060
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as751b74df
To: <sip:snomvs@192.168.0.14:5060>
Call-ID: 29fefa2422250d207319404a75fea06c@volker-sauer.de
CSeq: 102 OPTIONS
Contact: <sip:snomvs831@192.168.0.14:5060>;reg-id=1
User-Agent: snom370/8.4.31
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

<------------->
[Apr 20 08:46:24] --- (14 headers 0 lines) ---
[Apr 20 08:46:24] Really destroying SIP dialog '29fefa2422250d207319404a75fea06c@volker-sauer.de' Method: OPTIONS
[Apr 20 08:46:24] 
<--- SIP read from UDP:192.168.1.219:5060 --->
REGISTER sip:volker-sauer.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bKa13960c4ee1edce5145b1b5f888bc139;rport
From: "E. Sauer (58)" <sip:gigasetbbo@volker-sauer.de>;tag=1992373079
To: "E. Sauer (58)" <sip:gigasetbbo@volker-sauer.de>
Call-ID: 1230343168@192_168_1_219
CSeq: 6543 REGISTER
Contact: <sip:gigasetbbo@192.168.1.219:5060>
Max-Forwards: 70
User-Agent: S685IP/022270000000
Expires: 300
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
[Apr 20 08:46:24] --- (12 headers 0 lines) ---
[Apr 20 08:46:24] Sending to 192.168.1.219:5060 (NAT)
[Apr 20 08:46:24] Reliably Transmitting (NAT) to 192.168.1.219:5060:
OPTIONS sip:gigasetbbo@192.168.1.219:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.204.253:5060;branch=z9hG4bK59a867ab;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as6c1913ed
To: <sip:gigasetbbo@192.168.1.219:5060>
Contact: <sip:asterisk@192.168.204.253:5060>
Call-ID: 4afb11da0cd800684e70632f7aaaa3b7@volker-sauer.de
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 20 Apr 2012 06:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Apr 20 08:46:24] 
<--- Transmitting (NAT) to 192.168.1.219:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.219:5060;branch=z9hG4bKa13960c4ee1edce5145b1b5f888bc139;received=192.168.1.219;rport=5060
From: "E. Sauer (58)" <sip:gigasetbbo@volker-sauer.de>;tag=1992373079
To: "E. Sauer (58)" <sip:gigasetbbo@volker-sauer.de>;tag=as04b33235
Call-ID: 1230343168@192_168_1_219
CSeq: 6543 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:gigasetbbo@192.168.1.219:5060>;expires=300
Date: Fri, 20 Apr 2012 06:46:24 GMT
Content-Length: 0


<------------>
[Apr 20 08:46:24] Scheduling destruction of SIP dialog '1230343168@192_168_1_219' in 32000 ms (Method: REGISTER)
[Apr 20 08:46:24] 
<--- SIP read from UDP:192.168.0.14:5060 --->
REGISTER sip:asterisk.volker-sauer.de SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-5hh9hvbpup54;rport
From: "V.Sauer IT-1" <sip:snomvs833@asterisk.volker-sauer.de>;tag=r9pu3ue6hv
To: "V.Sauer IT-1" <sip:snomvs833@asterisk.volker-sauer.de>
Call-ID: 3c26702cbacf-kmwpty565g7z
CSeq: 161740 REGISTER
Max-Forwards: 70
Contact: <sip:snomvs833@192.168.0.14:5060>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:281047ed-2165-4e19-8592-077571b1c070>";audio;mobility="fixed";duplex="full";description="snom370";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom370/8.4.31
Allow-Events: dialog
X-Real-IP: 192.168.0.14
Supported: path, gruu
Expires: 3600
Content-Length: 0

<------------->
[Apr 20 08:46:24] --- (14 headers 0 lines) ---
[Apr 20 08:46:24] Sending to 192.168.0.14:5060 (NAT)
[Apr 20 08:46:24] Reliably Transmitting (NAT) to 192.168.0.14:5060:
OPTIONS sip:snomvs833@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0f73ec65;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as0b1235ca
To: <sip:snomvs833@192.168.0.14:5060>
Contact: <sip:asterisk@192.168.0.1:5060>
Call-ID: 7f4dd82d60fcf1283d27faed68d5bca0@volker-sauer.de
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 20 Apr 2012 06:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Apr 20 08:46:24] 
<--- Transmitting (NAT) to 192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-5hh9hvbpup54;received=192.168.0.14;rport=5060
From: "V.Sauer IT-1" <sip:snomvs833@asterisk.volker-sauer.de>;tag=r9pu3ue6hv
To: "V.Sauer IT-1" <sip:snomvs833@asterisk.volker-sauer.de>;tag=as16f1d9ef
Call-ID: 3c26702cbacf-kmwpty565g7z
CSeq: 161740 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:snomvs833@192.168.0.14:5060>;expires=300
Date: Fri, 20 Apr 2012 06:46:24 GMT
Content-Length: 0


<------------>
[Apr 20 08:46:24] Scheduling destruction of SIP dialog '3c26702cbacf-kmwpty565g7z' in 32000 ms (Method: REGISTER)
[Apr 20 08:46:24] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0f73ec65;rport=5060
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as0b1235ca
To: <sip:snomvs833@192.168.0.14:5060>
Call-ID: 7f4dd82d60fcf1283d27faed68d5bca0@volker-sauer.de
CSeq: 102 OPTIONS
Contact: <sip:snomvs831@192.168.0.14:5060>;reg-id=1
User-Agent: snom370/8.4.31
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

<------------->
[Apr 20 08:46:24] --- (14 headers 0 lines) ---
[Apr 20 08:46:24] Really destroying SIP dialog '7f4dd82d60fcf1283d27faed68d5bca0@volker-sauer.de' Method: OPTIONS
[Apr 20 08:46:24] 
<--- SIP read from UDP:192.168.1.219:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.204.253:5060;branch=z9hG4bK59a867ab;rport=5060
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as6c1913ed
To: <sip:gigasetbbo@192.168.1.219:5060>;tag=3547415798
Call-ID: 4afb11da0cd800684e70632f7aaaa3b7@volker-sauer.de
CSeq: 102 OPTIONS
Contact: <sip:gigasetbbo@192.168.1.219:5060>
Supported: replaces
Allow-Events: message-summary, refer
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0

<------------->
[Apr 20 08:46:24] --- (14 headers 0 lines) ---
[Apr 20 08:46:24] Really destroying SIP dialog '4afb11da0cd800684e70632f7aaaa3b7@volker-sauer.de' Method: OPTIONS
[Apr 20 08:46:24] 
<--- SIP read from UDP:192.168.0.14:5060 --->
REGISTER sip:asterisk.volker-sauer.de SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-3h9f6nowc1j0;rport
From: "V.Sauer Genion" <sip:snomvs831@asterisk.volker-sauer.de>;tag=3alwwjs2ki
To: "V.Sauer Genion" <sip:snomvs831@asterisk.volker-sauer.de>
Call-ID: 3c26702cb57c-l713jspk79zg
CSeq: 161773 REGISTER
Max-Forwards: 70
Contact: <sip:snomvs831@192.168.0.14:5060>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:08cb3362-66bb-4508-898e-aac75dce86c2>";audio;mobility="fixed";duplex="full";description="snom370";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom370/8.4.31
Allow-Events: dialog
X-Real-IP: 192.168.0.14
Supported: path, gruu
Expires: 3600
Content-Length: 0

<------------->
[Apr 20 08:46:24] --- (14 headers 0 lines) ---
[Apr 20 08:46:24] Sending to 192.168.0.14:5060 (NAT)
[Apr 20 08:46:24] Reliably Transmitting (NAT) to 192.168.0.14:5060:
OPTIONS sip:snomvs831@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK401b14ec;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as715ae3bd
To: <sip:snomvs831@192.168.0.14:5060>
Contact: <sip:asterisk@192.168.0.1:5060>
Call-ID: 12feda522d44edc8573bfdbe06d1c597@volker-sauer.de
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 20 Apr 2012 06:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Apr 20 08:46:24] 
<--- Transmitting (NAT) to 192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-3h9f6nowc1j0;received=192.168.0.14;rport=5060
From: "V.Sauer Genion" <sip:snomvs831@asterisk.volker-sauer.de>;tag=3alwwjs2ki
To: "V.Sauer Genion" <sip:snomvs831@asterisk.volker-sauer.de>;tag=as0754f443
Call-ID: 3c26702cb57c-l713jspk79zg
CSeq: 161773 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:snomvs831@192.168.0.14:5060>;expires=300
Date: Fri, 20 Apr 2012 06:46:24 GMT
Content-Length: 0


<------------>
[Apr 20 08:46:24] Scheduling destruction of SIP dialog '3c26702cb57c-l713jspk79zg' in 32000 ms (Method: REGISTER)
[Apr 20 08:46:24] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK401b14ec;rport=5060
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as715ae3bd
To: <sip:snomvs831@192.168.0.14:5060>
Call-ID: 12feda522d44edc8573bfdbe06d1c597@volker-sauer.de
CSeq: 102 OPTIONS
Contact: <sip:snomvs831@192.168.0.14:5060>;reg-id=1
User-Agent: snom370/8.4.31
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

<------------->
[Apr 20 08:46:24] --- (14 headers 0 lines) ---
[Apr 20 08:46:24] Really destroying SIP dialog '12feda522d44edc8573bfdbe06d1c597@volker-sauer.de' Method: OPTIONS
[Apr 20 08:46:24] 
<--- SIP read from UDP:192.168.0.14:5060 --->
REGISTER sip:asterisk.volker-sauer.de SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-jy28uetxyf7k;rport
From: "V.Sauer BBO" <sip:snomvs838@asterisk.volker-sauer.de>;tag=r36jl6honi
To: "V.Sauer BBO" <sip:snomvs838@asterisk.volker-sauer.de>
Call-ID: 3c26702d85dd-jqpvtnn0cehm
CSeq: 161714 REGISTER
Max-Forwards: 70
Contact: <sip:snomvs838@192.168.0.14:5060>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:db342480-ba09-4d29-9247-a389fc40504f>";audio;mobility="fixed";duplex="full";description="snom370";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom370/8.4.31
Allow-Events: dialog
X-Real-IP: 192.168.0.14
Supported: path, gruu
Expires: 3600
Content-Length: 0

<------------->
[Apr 20 08:46:24] --- (14 headers 0 lines) ---
[Apr 20 08:46:24] Sending to 192.168.0.14:5060 (NAT)
[Apr 20 08:46:24] Reliably Transmitting (NAT) to 192.168.0.14:5060:
OPTIONS sip:snomvs838@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK4e8002b9;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as7a564765
To: <sip:snomvs838@192.168.0.14:5060>
Contact: <sip:asterisk@192.168.0.1:5060>
Call-ID: 2b8309d56abcda74056366252f5deb3c@volker-sauer.de
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 20 Apr 2012 06:46:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Apr 20 08:46:24] 
<--- Transmitting (NAT) to 192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-jy28uetxyf7k;received=192.168.0.14;rport=5060
From: "V.Sauer BBO" <sip:snomvs838@asterisk.volker-sauer.de>;tag=r36jl6honi
To: "V.Sauer BBO" <sip:snomvs838@asterisk.volker-sauer.de>;tag=as6a7d0226
Call-ID: 3c26702d85dd-jqpvtnn0cehm
CSeq: 161714 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:snomvs838@192.168.0.14:5060>;expires=300
Date: Fri, 20 Apr 2012 06:46:24 GMT
Content-Length: 0


<------------>
[Apr 20 08:46:24] Scheduling destruction of SIP dialog '3c26702d85dd-jqpvtnn0cehm' in 32000 ms (Method: REGISTER)
[Apr 20 08:46:24] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK4e8002b9;rport=5060
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as7a564765
To: <sip:snomvs838@192.168.0.14:5060>
Call-ID: 2b8309d56abcda74056366252f5deb3c@volker-sauer.de
CSeq: 102 OPTIONS
Contact: <sip:snomvs831@192.168.0.14:5060>;reg-id=1
User-Agent: snom370/8.4.31
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

<------------->
[Apr 20 08:46:24] --- (14 headers 0 lines) ---
[Apr 20 08:46:24] Really destroying SIP dialog '2b8309d56abcda74056366252f5deb3c@volker-sauer.de' Method: OPTIONS
[Apr 20 08:46:25] 
<--- SIP read from UDP:192.168.0.14:5060 --->
REGISTER sip:asterisk.volker-sauer.de SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-kpdx6ccxg8of;rport
From: "Anonym" <sip:snomvs839@asterisk.volker-sauer.de>;tag=9s1rh5p0z3
To: "Anonym" <sip:snomvs839@asterisk.volker-sauer.de>
Call-ID: 3c26702d60a2-7y910b11d8i8
CSeq: 161738 REGISTER
Max-Forwards: 70
Contact: <sip:snomvs839@192.168.0.14:5060>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:ea447901-5d12-4c6c-9bc3-175b6eaae36c>";audio;mobility="fixed";duplex="full";description="snom370";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom370/8.4.31
Allow-Events: dialog
X-Real-IP: 192.168.0.14
Supported: path, gruu
Expires: 3600
Content-Length: 0

<------------->
[Apr 20 08:46:25] --- (14 headers 0 lines) ---
[Apr 20 08:46:25] Sending to 192.168.0.14:5060 (NAT)
[Apr 20 08:46:25] Reliably Transmitting (NAT) to 192.168.0.14:5060:
OPTIONS sip:snomvs839@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK6d91eeb7;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as3eb8b370
To: <sip:snomvs839@192.168.0.14:5060>
Contact: <sip:asterisk@192.168.0.1:5060>
Call-ID: 44f4357c76f4537d36b58e4b769fe140@volker-sauer.de
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 20 Apr 2012 06:46:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Apr 20 08:46:25] 
<--- Transmitting (NAT) to 192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-kpdx6ccxg8of;received=192.168.0.14;rport=5060
From: "Anonym" <sip:snomvs839@asterisk.volker-sauer.de>;tag=9s1rh5p0z3
To: "Anonym" <sip:snomvs839@asterisk.volker-sauer.de>;tag=as6b02e6c3
Call-ID: 3c26702d60a2-7y910b11d8i8
CSeq: 161738 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: <sip:snomvs839@192.168.0.14:5060>;expires=300
Date: Fri, 20 Apr 2012 06:46:25 GMT
Content-Length: 0


<------------>
[Apr 20 08:46:25] Scheduling destruction of SIP dialog '3c26702d60a2-7y910b11d8i8' in 32000 ms (Method: REGISTER)
[Apr 20 08:46:25] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK6d91eeb7;rport=5060
From: "asterisk" <sip:asterisk@volker-sauer.de>;tag=as3eb8b370
To: <sip:snomvs839@192.168.0.14:5060>
Call-ID: 44f4357c76f4537d36b58e4b769fe140@volker-sauer.de
CSeq: 102 OPTIONS
Contact: <sip:snomvs831@192.168.0.14:5060>;reg-id=1
User-Agent: snom370/8.4.31
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

<------------->
[Apr 20 08:46:25] --- (14 headers 0 lines) ---
[Apr 20 08:46:25] Really destroying SIP dialog '44f4357c76f4537d36b58e4b769fe140@volker-sauer.de' Method: OPTIONS
[Apr 20 08:46:26] 
<--- SIP read from UDP:192.168.0.10:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7ae207d7;rport=5060
From: "V1:Sauer,Volker(cell)" <sip:01796901475@volker-sauer.de>;tag=as44e1ac96
To: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>;tag=xde12r
Call-ID: 4ea097e456e2746a6fec07ef20c5c1b8@volker-sauer.de
CSeq: 102 INVITE
Contact: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>
Supported: 100rel, replaces, norefersub
User-Agent: snom-m9/9.5.14-a
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 1476000769 1476000770 IN IP4 192.168.0.10
s=-
c=IN IP4 192.168.0.10
t=0 0
m=audio 52872 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
[Apr 20 08:46:26] --- (11 headers 10 lines) ---
[Apr 20 08:46:26] Found RTP audio format 0
[Apr 20 08:46:26] Found RTP audio format 101
[Apr 20 08:46:26] Found audio description format pcmu for ID 0
[Apr 20 08:46:26] Found audio description format telephone-event for ID 101
[Apr 20 08:46:26] Capabilities: us - (gsm|ulaw|alaw|g726|adpcm|g722), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
[Apr 20 08:46:26] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Apr 20 08:46:26] Peer audio RTP is at port 192.168.0.10:52872
[Apr 20 08:46:26] list_route: hop: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>
[Apr 20 08:46:26] set_destination: Parsing <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz> for address/port to send to
[Apr 20 08:46:26] set_destination: set destination to 192.168.0.10:5060
[Apr 20 08:46:26] set_destination: Parsing <sip:snomvs@192.168.0.14:5060> for address/port to send to
[Apr 20 08:46:26] Transmitting (NAT) to 192.168.0.10:5060:
ACK sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK6c18bbd8;rport
Max-Forwards: 70
From: "V1:Sauer,Volker(cell)" <sip:01796901475@volker-sauer.de>;tag=as44e1ac96
To: <sip:snomm9-2@192.168.0.10:5060;transport=udp;line=ccrjtz>;tag=xde12r
Contact: <sip:01796901475@192.168.0.1:5060>
Call-ID: 4ea097e456e2746a6fec07ef20c5c1b8@volker-sauer.de
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.3.0
Content-Length: 0


---
[Apr 20 08:46:26] set_destination: set destination to 192.168.0.14:5060
[Apr 20 08:46:26] Reliably Transmitting (NAT) to 192.168.0.14:5060:
NOTIFY sip:snomvs@192.168.0.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK61799f9a;rport
Max-Forwards: 70
From: <sip:85@asterisk.volker-sauer.de;user=phone>;tag=as49871c93
To: <sip:snomvs@asterisk.volker-sauer.de>;tag=8t8yucyvon
Contact: <sip:85@192.168.0.1:5060>
Call-ID: 3cdeacb7bb7d-bq69dm3pa2p4
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX 10.3.0
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 212

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="2" state="full" entity="sip:85@asterisk.volker-sauer.de">
<dialog id="85">
<state>confirmed</state>
</dialog>
</dialog-info>

---
[Apr 20 08:46:26]   == Extension Changed 85[incoming] new state InUse for Notify User snomvs 
[Apr 20 08:46:26]     -- SIP/snomm9-2-00000002 answered Local/85@doLocalCallsPrivat-99d3;2
[Apr 20 08:46:26]     -- Local/85@doLocalCallsPrivat-99d3;1 answered SIP/sipgate-gmbh-e0-00000000
[Apr 20 08:46:26] Audio is at 10020
[Apr 20 08:46:26] Adding codec 100003 (ulaw) to SDP
[Apr 20 08:46:26] Adding codec 100004 (alaw) to SDP
[Apr 20 08:46:26] Adding codec 100002 (gsm) to SDP
[Apr 20 08:46:26] Adding codec 100011 (g726) to SDP
[Apr 20 08:46:26] Adding non-codec 0x1 (telephone-event) to SDP
[Apr 20 08:46:26] 
<--- Reliably Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK68c3.6651ffd1.0;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK68c3.6651ffd1.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK0ebe7890
Via: SIP/2.0/UDP 217.10.67.13:5060;received=217.10.67.13;branch=z9hG4bK0ebe7890;rport=5060
Record-Route: <sip:217.10.79.9;lr;ftag=as026b057e>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr;ftag=as026b057e>
From: "01796901475" <sip:01796901475@sipgate.de>;tag=as026b057e
To: <sip:00496151154260@sipgate.de>;tag=as5a1d63af
Call-ID: 2aa296b60bf3ebb91dea752762df8a63@sipgate.de
CSeq: 102 INVITE
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2216950e1@130.83.208.238:5060>
Content-Type: application/sdp
Content-Length: 317

v=0
o=root 1924178378 1924178378 IN IP4 130.83.208.238
s=Asterisk PBX 10.3.0
c=IN IP4 130.83.208.238
t=0 0
m=audio 10020 RTP/AVP 0 8 3 112 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Apr 20 08:46:26] 
<--- SIP read from UDP:217.10.79.9:5060 --->
ACK sip:2216950e1@130.83.208.238:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK68c3.6651ffd1.2
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK68c3.6651ffd1.2
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK554caccc
Via: SIP/2.0/UDP 217.10.67.13:5060;received=217.10.67.13;branch=z9hG4bK554caccc;rport=5060
Max-Forwards: 67
From: "01796901475" <sip:01796901475@sipgate.de>;tag=as026b057e
To: <sip:00496151154260@sipgate.de>;tag=as5a1d63af
Contact: <sip:01796901475@217.10.67.13>
Call-ID: 2aa296b60bf3ebb91dea752762df8a63@sipgate.de
CSeq: 102 ACK
Content-Length: 0
X-hint: rr-enforced

<------------->
[Apr 20 08:46:26] --- (13 headers 0 lines) ---
[Apr 20 08:46:26] 
<--- SIP read from UDP:192.168.0.14:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK61799f9a;rport=5060
From: <sip:85@asterisk.volker-sauer.de;user=phone>;tag=as49871c93
To: <sip:snomvs@asterisk.volker-sauer.de>;tag=8t8yucyvon
Call-ID: 3cdeacb7bb7d-bq69dm3pa2p4
CSeq: 104 NOTIFY
Content-Length: 0

<------------->
[Apr 20 08:46:26] --- (7 headers 0 lines) ---
[Apr 20 08:46:26]   == Spawn extension (macro-CallInt2Int, s, 5) exited non-zero on 'Local/85@doLocalCallsPrivat-99d3;2' in macro 'CallInt2Int'
[Apr 20 08:46:26]   == Spawn extension (incoming, 85, 100) exited non-zero on 'Local/85@doLocalCallsPrivat-99d3;2'
[Apr 20 08:46:26] ERROR[19435]: lock.c:280 __ast_pthread_mutex_lock: audiohook.c line 842 (audio_audiohook_write_list): Error obtaining mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:407 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): mutex '&(audiohook)->lock' freed more times than we've locked!
[Apr 20 08:46:26] ERROR[19435]: lock.c:438 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): Error releasing mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:280 __ast_pthread_mutex_lock: audiohook.c line 842 (audio_audiohook_write_list): Error obtaining mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:407 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): mutex '&(audiohook)->lock' freed more times than we've locked!
[Apr 20 08:46:26] ERROR[19435]: lock.c:438 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): Error releasing mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:280 __ast_pthread_mutex_lock: audiohook.c line 842 (audio_audiohook_write_list): Error obtaining mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:407 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): mutex '&(audiohook)->lock' freed more times than we've locked!
[Apr 20 08:46:26] ERROR[19435]: lock.c:438 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): Error releasing mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:280 __ast_pthread_mutex_lock: audiohook.c line 842 (audio_audiohook_write_list): Error obtaining mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:407 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): mutex '&(audiohook)->lock' freed more times than we've locked!
[Apr 20 08:46:26] ERROR[19435]: lock.c:438 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): Error releasing mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:280 __ast_pthread_mutex_lock: audiohook.c line 842 (audio_audiohook_write_list): Error obtaining mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:407 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): mutex '&(audiohook)->lock' freed more times than we've locked!
[Apr 20 08:46:26] ERROR[19435]: lock.c:438 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): Error releasing mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:280 __ast_pthread_mutex_lock: audiohook.c line 842 (audio_audiohook_write_list): Error obtaining mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:407 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): mutex '&(audiohook)->lock' freed more times than we've locked!
[Apr 20 08:46:26] ERROR[19435]: lock.c:438 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): Error releasing mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:280 __ast_pthread_mutex_lock: audiohook.c line 842 (audio_audiohook_write_list): Error obtaining mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:407 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): mutex '&(audiohook)->lock' freed more times than we've locked!
[Apr 20 08:46:26] ERROR[19435]: lock.c:438 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): Error releasing mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:280 __ast_pthread_mutex_lock: audiohook.c line 842 (audio_audiohook_write_list): Error obtaining mutex: Invalid argument
[Apr 20 08:46:26] ERROR[19435]: lock.c:407 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): mutex '&(audiohook)->lock' freed more times than we've locked!


[repeated....]

[Apr 20 08:46:30] ERROR[19435]: lock.c:280 __ast_pthread_mutex_lock: audiohook.c line 842 (audio_audiohook_write_list): Error obtaining mutex: Invalid argument
[Apr 20 08:46:30] ERROR[19435]: lock.c:407 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): mutex '&(audiohook)->lock' freed more times than we've locked!
[Apr 20 08:46:30] ERROR[19435]: lock.c:438 __ast_pthread_mutex_unlock: audiohook.c line 861 (audio_audiohook_write_list): Error releasing mutex: Invalid argument
[Apr 20 08:46:30] 
<--- SIP read from UDP:217.10.79.9:5060 --->
BYE sip:2216950e1@130.83.208.238:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK78c3.269f1871.0
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK78c3.269f1871.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK78279070
Via: SIP/2.0/UDP 217.10.67.13:5060;received=217.10.67.13;branch=z9hG4bK78279070;rport=5060
Max-Forwards: 67
From: "01796901475" <sip:01796901475@sipgate.de>;tag=as026b057e
To: <sip:00496151154260@sipgate.de>;tag=as5a1d63af
Call-ID: 2aa296b60bf3ebb91dea752762df8a63@sipgate.de
CSeq: 103 BYE
Content-Length: 0
X-hint: rr-enforced

<------------->
[Apr 20 08:46:30] --- (12 headers 0 lines) ---
[Apr 20 08:46:30] Sending to 217.10.79.9:5060 (NAT)
[Apr 20 08:46:30] Scheduling destruction of SIP dialog '2aa296b60bf3ebb91dea752762df8a63@sipgate.de' in 6400 ms (Method: BYE)
[Apr 20 08:46:30] 
<--- Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK78c3.269f1871.0;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK78c3.269f1871.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK78279070
Via: SIP/2.0/UDP 217.10.67.13:5060;received=217.10.67.13;branch=z9hG4bK78279070;rport=5060
From: "01796901475" <sip:01796901475@sipgate.de>;tag=as026b057e
To: <sip:00496151154260@sipgate.de>;tag=as5a1d63af
Call-ID: 2aa296b60bf3ebb91dea752762df8a63@sipgate.de
CSeq: 103 BYE
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


[crash]