Angerufener hört mich ab und an nicht

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Holg

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Hallo,

Selstames Problem. Seit heute hört manchmal derjenige, den ich anrufe mich nicht, aber nicht immer.

Liegt das eher am asterisk, am Telefon oder am Provider (QSC)?

[edit]
Vor allem wenn ich jemanden zweimal hintereinander anrufe passiert das.


Danke für etwaige Tips.
Gruß
Holg
 
Zuletzt bearbeitet:
Also Portforwarding stimmt. DNS wird auch aufgelöst...
Das macht mich echt wahnsinnig. Bei uns kann keiner mehr richtig abgehend telefonieren. Da es wirklich Glückssache ist, ob der Gegenüber einen hört oder nicht...

hier mal noch das sip debugging auf das Telefon, von dem ich telefoniert habe... Erst drei Anrufe, bei denen man mich nicht hört und der letzte ist dann wieder normal...

Code: 
Max-Forwards: 70
Date: Fri, 12 May 2006 11:47:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK03146f79;rport
From: "asterisk" <sip:[email protected]>;tag=as6214f392
To: <sip:[email protected]:50696>;tag=d75cdb7d144d84cc
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Grandstream GXP2000 1.0.2.3
Contact: <sip:[email protected]:50696>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0

--- (11 headers 0 lines)---
Destroying call '[email protected]'
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
INVITE sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK96bc46b4f986633d
From: "Holger Sorg" <sip:[email protected]>;tag=48cb7995d605b4f7
To: <sip:###Handynr###@10.2.1.1>
Contact: <sip:[email protected]:50696>
Supported: replaces, timer
Call-ID: [email protected]
CSeq: 1546 INVITE
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 255

v=0
o=96 8000 8000 IN IP4 10.2.1.96
s=SIP Call
c=IN IP4 10.2.1.96
t=0 0
m=audio 11096 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (13 headers 13 lines)---
Using INVITE request as basis request - [email protected]
Sending to 10.2.1.96 : 50696 (non-NAT)
Reliably Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK96bc46b4f986633d;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=48cb7995d605b4f7
To: <sip:###Handynr###@10.2.1.1>;tag=as594b9a78
Call-ID: [email protected]
CSeq: 1546 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Proxy-Authenticate: Digest realm="asterisk", nonce="0a823cb0"
Content-Length: 0


---
Scheduling destruction of call '[email protected]' in 15000 ms
Found user '96'
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
ACK sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK96bc46b4f986633d
From: "Holger Sorg" <sip:[email protected]>;tag=48cb7995d605b4f7
To: <sip:###Handynr###@10.2.1.1>;tag=as594b9a78
Contact: <sip:[email protected]:50696>
Call-ID: [email protected]
CSeq: 1546 ACK
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

--- (11 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
INVITE sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK4d53d16f5b5cb66c
From: "Holger Sorg" <sip:[email protected]>;tag=48cb7995d605b4f7
To: <sip:###Handynr###@10.2.1.1>
Contact: <sip:[email protected]:50696>
Supported: replaces, timer
Proxy-Authorization: Digest username="96", realm="asterisk", algorithm=MD5, uri="sip:###Handynr###@10.2.1.1", nonce="0a823cb0", response="6ec6e98a073fbb7beb90b27ee6272a90"
Call-ID: [email protected]
CSeq: 1547 INVITE
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 255

v=0
o=96 8000 8001 IN IP4 10.2.1.96
s=SIP Call
c=IN IP4 10.2.1.96
t=0 0
m=audio 11096 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (14 headers 13 lines)---
Using INVITE request as basis request - [email protected]
Sending to 10.2.1.96 : 50696 (NAT)
Found user '96'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.2.1.96:11096
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for ###Handynr### in ext-96 (domain 10.2.1.1)
list_route: hop: <sip:[email protected]:50696>
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK4d53d16f5b5cb66c;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=48cb7995d605b4f7
To: <sip:###Handynr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 1547 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Length: 0


---
    -- Executing Macro("SIP/96-1f40", "dialout|SIP|###Handynr###|@PEER-96||") in new stack
    -- Executing Dial("SIP/96-1f40", "SIP/###Handynr###@PEER-96||") in new stack
    -- Called ###Handynr###@PEER-96
    -- SIP/PEER-96-ee8f is making progress passing it to SIP/96-1f40
We're at 10.2.1.1 port 15384
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK4d53d16f5b5cb66c;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=48cb7995d605b4f7
To: <sip:###Handynr###@10.2.1.1>;tag=as622e74d8
Call-ID: [email protected]
CSeq: 1547 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 6514 6514 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 15384 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:

--- (0 headers 0 lines) Nat keepalive ---
    -- SIP/PEER-96-ee8f is making progress passing it to SIP/96-1f40
    -- SIP/PEER-96-ee8f is ringing
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK4d53d16f5b5cb66c;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=48cb7995d605b4f7
To: <sip:###Handynr###@10.2.1.1>;tag=as622e74d8
Call-ID: [email protected]
CSeq: 1547 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Length: 0


---
    -- SIP/PEER-96-ee8f is making progress passing it to SIP/96-1f40
    -- SIP/PEER-96-ee8f answered SIP/96-1f40
We're at 10.2.1.1 port 15384
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK4d53d16f5b5cb66c;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=48cb7995d605b4f7
To: <sip:###Handynr###@10.2.1.1>;tag=as622e74d8
Call-ID: [email protected]
CSeq: 1547 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 6514 6515 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 15384 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/96-1f40 and SIP/PEER-96-ee8f
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
ACK sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKb0dd73015d932d83
From: "Holger Sorg" <sip:[email protected]>;tag=48cb7995d605b4f7
To: <sip:###Handynr###@10.2.1.1>;tag=as622e74d8
Contact: <sip:[email protected]:50696>
Proxy-Authorization: Digest username="96", realm="asterisk", algorithm=MD5, uri="sip:###Handynr###@10.2.1.1", nonce="0a823cb0", response="d0db97e8db4ebb356cc6c84b02856fc5"
Call-ID: [email protected]
CSeq: 1547 ACK
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

--- (12 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
BYE sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKe9d6b20fe812b3bc
From: "Holger Sorg" <sip:[email protected]>;tag=48cb7995d605b4f7
To: <sip:###Handynr###@10.2.1.1>;tag=as622e74d8
Proxy-Authorization: Digest username="96", realm="asterisk", algorithm=MD5, uri="sip:###Handynr###@10.2.1.1", nonce="0a823cb0", response="252bb475c37195454f9791607d730386"
Call-ID: [email protected]
CSeq: 1548 BYE
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

--- (11 headers 0 lines)---
Sending to 10.2.1.96 : 50696 (NAT)
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKe9d6b20fe812b3bc;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=48cb7995d605b4f7
To: <sip:###Handynr###@10.2.1.1>;tag=as622e74d8
Call-ID: [email protected]
CSeq: 1548 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
  == Spawn extension (macro-dialout, s, 1) exited non-zero on 'SIP/96-1f40' in macro 'dialout'
  == Spawn extension (macro-dialout, s, 1) exited non-zero on 'SIP/96-1f40'
    -- Executing Hangup("SIP/96-1f40", "") in new stack
  == Spawn extension (macro-dialout, h, 1) exited non-zero on 'SIP/96-1f40'
Destroying call '[email protected]'
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:

--- (0 headers 0 lines) Nat keepalive ---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
INVITE sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK2e388640e9136b81
From: "Holger Sorg" <sip:[email protected]>;tag=5bb9dd9d50e152a0
To: <sip:###Handynr###@10.2.1.1>
Contact: <sip:[email protected]:50696>
Supported: replaces, timer
Call-ID: [email protected]
CSeq: 29804 INVITE
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 255

v=0
o=96 8000 8000 IN IP4 10.2.1.96
s=SIP Call
c=IN IP4 10.2.1.96
t=0 0
m=audio 11096 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (13 headers 13 lines)---
Using INVITE request as basis request - [email protected]
Sending to 10.2.1.96 : 50696 (non-NAT)
Reliably Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK2e388640e9136b81;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=5bb9dd9d50e152a0
To: <sip:###Handynr###@10.2.1.1>;tag=as118ff474
Call-ID: [email protected]
CSeq: 29804 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Proxy-Authenticate: Digest realm="asterisk", nonce="24919023"
Content-Length: 0


---
Scheduling destruction of call '[email protected]' in 15000 ms
Found user '96'
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
ACK sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK2e388640e9136b81
From: "Holger Sorg" <sip:[email protected]>;tag=5bb9dd9d50e152a0
To: <sip:###Handynr###@10.2.1.1>;tag=as118ff474
Contact: <sip:[email protected]:50696>
Call-ID: [email protected]
CSeq: 29804 ACK
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

--- (11 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
INVITE sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKdf9620b24d63afd3
From: "Holger Sorg" <sip:[email protected]>;tag=5bb9dd9d50e152a0
To: <sip:###Handynr###@10.2.1.1>
Contact: <sip:[email protected]:50696>
Supported: replaces, timer
Proxy-Authorization: Digest username="96", realm="asterisk", algorithm=MD5, uri="sip:###Handynr###@10.2.1.1", nonce="24919023", response="050f89111f1ce8882c7d380d3b3f3778"
Call-ID: [email protected]
CSeq: 29805 INVITE
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 255

v=0
o=96 8000 8001 IN IP4 10.2.1.96
s=SIP Call
c=IN IP4 10.2.1.96
t=0 0
m=audio 11096 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (14 headers 13 lines)---
Using INVITE request as basis request - [email protected]
Sending to 10.2.1.96 : 50696 (NAT)
Found user '96'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.2.1.96:11096
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for ###Handynr### in ext-96 (domain 10.2.1.1)
list_route: hop: <sip:[email protected]:50696>
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKdf9620b24d63afd3;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=5bb9dd9d50e152a0
To: <sip:###Handynr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 29805 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Length: 0


---
    -- Executing Macro("SIP/96-e99f", "dialout|SIP|###Handynr###|@PEER-96||") in new stack
    -- Executing Dial("SIP/96-e99f", "SIP/###Handynr###@PEER-96||") in new stack
    -- Called ###Handynr###@PEER-96
    -- SIP/PEER-96-c446 is making progress passing it to SIP/96-e99f
We're at 10.2.1.1 port 18674
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKdf9620b24d63afd3;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=5bb9dd9d50e152a0
To: <sip:###Handynr###@10.2.1.1>;tag=as3b1f836f
Call-ID: [email protected]
CSeq: 29805 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 6514 6514 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 18674 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- SIP/PEER-96-c446 is ringing
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKdf9620b24d63afd3;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=5bb9dd9d50e152a0
To: <sip:###Handynr###@10.2.1.1>;tag=as3b1f836f
Call-ID: [email protected]
CSeq: 29805 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Length: 0


---
    -- SIP/PEER-96-c446 is making progress passing it to SIP/96-e99f
    -- SIP/PEER-96-c446 answered SIP/96-e99f
We're at 10.2.1.1 port 18674
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKdf9620b24d63afd3;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=5bb9dd9d50e152a0
To: <sip:###Handynr###@10.2.1.1>;tag=as3b1f836f
Call-ID: [email protected]
CSeq: 29805 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 6514 6515 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 18674 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/96-e99f and SIP/PEER-96-c446
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
ACK sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK243ca5ff9d2e388e
From: "Holger Sorg" <sip:[email protected]>;tag=5bb9dd9d50e152a0
To: <sip:###Handynr###@10.2.1.1>;tag=as3b1f836f
Contact: <sip:[email protected]:50696>
Proxy-Authorization: Digest username="96", realm="asterisk", algorithm=MD5, uri="sip:###Handynr###@10.2.1.1", nonce="24919023", response="4a2d7543da076d81dc3b386948eb4843"
Call-ID: [email protected]
CSeq: 29805 ACK
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

--- (12 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:

--- (0 headers 0 lines) Nat keepalive ---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
BYE sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK65fdab4f287b1240
From: "Holger Sorg" <sip:[email protected]>;tag=5bb9dd9d50e152a0
To: <sip:###Handynr###@10.2.1.1>;tag=as3b1f836f
Proxy-Authorization: Digest username="96", realm="asterisk", algorithm=MD5, uri="sip:###Handynr###@10.2.1.1", nonce="24919023", response="2086b7ee88536153503793afe9ccc147"
Call-ID: [email protected]
CSeq: 29806 BYE
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

--- (11 headers 0 lines)---
Sending to 10.2.1.96 : 50696 (NAT)
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK65fdab4f287b1240;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=5bb9dd9d50e152a0
To: <sip:###Handynr###@10.2.1.1>;tag=as3b1f836f
Call-ID: [email protected]
CSeq: 29806 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
  == Spawn extension (macro-dialout, s, 1) exited non-zero on 'SIP/96-e99f' in macro 'dialout'
  == Spawn extension (macro-dialout, s, 1) exited non-zero on 'SIP/96-e99f'
    -- Executing Hangup("SIP/96-e99f", "") in new stack
  == Spawn extension (macro-dialout, h, 1) exited non-zero on 'SIP/96-e99f'
Destroying call '[email protected]'
12 headers, 0 lines
Reliably Transmitting (NAT) to 10.2.1.96:50696:
OPTIONS sip:[email protected]:50696 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK7e7a4229;rport
From: "asterisk" <sip:[email protected]>;tag=as4f2c750b
To: <sip:[email protected]:50696>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 12 May 2006 11:48:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK7e7a4229;rport
From: "asterisk" <sip:[email protected]>;tag=as4f2c750b
To: <sip:[email protected]:50696>;tag=7cb4369c445b925b
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Grandstream GXP2000 1.0.2.3
Contact: <sip:[email protected]:50696>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0

--- (11 headers 0 lines)---
Destroying call '[email protected]'
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
INVITE sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKebaded03bc3b5d23
From: "Holger Sorg" <sip:[email protected]>;tag=f7b2db730f434ad5
To: <sip:###Handynr###@10.2.1.1>
Contact: <sip:[email protected]:50696>
Supported: replaces, timer
Call-ID: [email protected]
CSeq: 20557 INVITE
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 255

v=0
o=96 8000 8000 IN IP4 10.2.1.96
s=SIP Call
c=IN IP4 10.2.1.96
t=0 0
m=audio 11096 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (13 headers 13 lines)---
Using INVITE request as basis request - [email protected]
Sending to 10.2.1.96 : 50696 (non-NAT)
Reliably Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKebaded03bc3b5d23;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=f7b2db730f434ad5
To: <sip:###Handynr###@10.2.1.1>;tag=as3644b918
Call-ID: [email protected]
CSeq: 20557 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Proxy-Authenticate: Digest realm="asterisk", nonce="7774efa7"
Content-Length: 0


---
Scheduling destruction of call '[email protected]' in 15000 ms
Found user '96'
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
ACK sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKebaded03bc3b5d23
From: "Holger Sorg" <sip:[email protected]>;tag=f7b2db730f434ad5
To: <sip:###Handynr###@10.2.1.1>;tag=as3644b918
Contact: <sip:[email protected]:50696>
Call-ID: [email protected]
CSeq: 20557 ACK
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

--- (11 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
INVITE sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK180d9cb1fdd09000
From: "Holger Sorg" <sip:[email protected]>;tag=f7b2db730f434ad5
To: <sip:###Handynr###@10.2.1.1>
Contact: <sip:[email protected]:50696>
Supported: replaces, timer
Proxy-Authorization: Digest username="96", realm="asterisk", algorithm=MD5, uri="sip:###Handynr###@10.2.1.1", nonce="7774efa7", response="99b5a3f075bb2cb58c1fbd63456efc99"
Call-ID: [email protected]
CSeq: 20558 INVITE
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 255

v=0
o=96 8000 8001 IN IP4 10.2.1.96
s=SIP Call
c=IN IP4 10.2.1.96
t=0 0
m=audio 11096 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (14 headers 13 lines)---
Using INVITE request as basis request - [email protected]
Sending to 10.2.1.96 : 50696 (NAT)
Found user '96'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.2.1.96:11096
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for ###Handynr### in ext-96 (domain 10.2.1.1)
list_route: hop: <sip:[email protected]:50696>
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK180d9cb1fdd09000;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=f7b2db730f434ad5
To: <sip:###Handynr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 20558 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Length: 0


---
    -- Executing Macro("SIP/96-a0ff", "dialout|SIP|###Handynr###|@PEER-96||") in new stack
    -- Executing Dial("SIP/96-a0ff", "SIP/###Handynr###@PEER-96||") in new stack
    -- Called ###Handynr###@PEER-96
    -- SIP/PEER-96-e960 is making progress passing it to SIP/96-a0ff
We're at 10.2.1.1 port 12838
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK180d9cb1fdd09000;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=f7b2db730f434ad5
To: <sip:###Handynr###@10.2.1.1>;tag=as44453e4b
Call-ID: [email protected]
CSeq: 20558 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 6514 6514 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 12838 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- SIP/PEER-96-e960 is ringing
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK180d9cb1fdd09000;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=f7b2db730f434ad5
To: <sip:###Handynr###@10.2.1.1>;tag=as44453e4b
Call-ID: [email protected]
CSeq: 20558 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Length: 0


---
    -- SIP/PEER-96-e960 is making progress passing it to SIP/96-a0ff
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:

--- (0 headers 0 lines) Nat keepalive ---
May 12 13:48:50 NOTICE[6526]: chan_sip.c:5259 sip_reregister:    -- Re-registration for  [email protected]
May 12 13:48:50 NOTICE[6526]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sip.qsc.de is 120 sec (Scheduling reregistration in 105 s)
    -- SIP/PEER-96-e960 answered SIP/96-a0ff
We're at 10.2.1.1 port 12838
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK180d9cb1fdd09000;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=f7b2db730f434ad5
To: <sip:###Handynr###@10.2.1.1>;tag=as44453e4b
Call-ID: [email protected]
CSeq: 20558 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 6514 6515 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 12838 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/96-a0ff and SIP/PEER-96-e960
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
ACK sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bK28c85ca11e330e53
From: "Holger Sorg" <sip:[email protected]>;tag=f7b2db730f434ad5
To: <sip:###Handynr###@10.2.1.1>;tag=as44453e4b
Contact: <sip:[email protected]:50696>
Proxy-Authorization: Digest username="96", realm="asterisk", algorithm=MD5, uri="sip:###Handynr###@10.2.1.1", nonce="7774efa7", response="bcc84df8391c1964bee13b01b94cae92"
Call-ID: [email protected]
CSeq: 20558 ACK
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

--- (12 headers 0 lines)---
May 12 13:48:51 NOTICE[6526]: chan_sip.c:5259 sip_reregister:    -- Re-registration for  [email protected]
May 12 13:48:51 NOTICE[6526]: chan_sip.c:5259 sip_reregister:    -- Re-registration for  [email protected]
May 12 13:48:51 NOTICE[6526]: chan_sip.c:5259 sip_reregister:    -- Re-registration for  [email protected]
May 12 13:48:51 NOTICE[6526]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sip.qsc.de is 120 sec (Scheduling reregistration in 105 s)
May 12 13:48:51 NOTICE[6526]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sip.qsc.de is 120 sec (Scheduling reregistration in 105 s)
May 12 13:48:51 NOTICE[6526]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sip.qsc.de is 120 sec (Scheduling reregistration in 105 s)
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
BYE sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKab21376e90cb5aa9
From: "Holger Sorg" <sip:[email protected]>;tag=f7b2db730f434ad5
To: <sip:###Handynr###@10.2.1.1>;tag=as44453e4b
Proxy-Authorization: Digest username="96", realm="asterisk", algorithm=MD5, uri="sip:###Handynr###@10.2.1.1", nonce="7774efa7", response="11d5bde1c7dacd2a44fe5b89b5be61b4"
Call-ID: [email protected]
CSeq: 20559 BYE
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

--- (11 headers 0 lines)---
Sending to 10.2.1.96 : 50696 (NAT)
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKab21376e90cb5aa9;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=f7b2db730f434ad5
To: <sip:###Handynr###@10.2.1.1>;tag=as44453e4b
Call-ID: [email protected]
CSeq: 20559 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
  == Spawn extension (macro-dialout, s, 1) exited non-zero on 'SIP/96-a0ff' in macro 'dialout'
  == Spawn extension (macro-dialout, s, 1) exited non-zero on 'SIP/96-a0ff'
    -- Executing Hangup("SIP/96-a0ff", "") in new stack
  == Spawn extension (macro-dialout, h, 1) exited non-zero on 'SIP/96-a0ff'
Destroying call '[email protected]'
May 12 13:48:58 NOTICE[6526]: chan_sip.c:5259 sip_reregister:    -- Re-registration for  [email protected]
May 12 13:48:58 NOTICE[6526]: chan_sip.c:5259 sip_reregister:    -- Re-registration for  [email protected]
May 12 13:48:58 NOTICE[6526]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sipgate.de is 120 sec (Scheduling reregistration in 105 s)
May 12 13:48:58 NOTICE[6526]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sipgate.de is 120 sec (Scheduling reregistration in 105 s)
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
INVITE sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKcf8b44516c6feb29
From: "Holger Sorg" <sip:[email protected]>;tag=aee404565f47ce62
To: <sip:###Handynr###@10.2.1.1>
Contact: <sip:[email protected]:50696>
Supported: replaces, timer
Call-ID: [email protected]
CSeq: 48438 INVITE
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 255

v=0
o=96 8000 8000 IN IP4 10.2.1.96
s=SIP Call
c=IN IP4 10.2.1.96
t=0 0
m=audio 11096 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (13 headers 13 lines)---
Using INVITE request as basis request - [email protected]
Sending to 10.2.1.96 : 50696 (non-NAT)
Reliably Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKcf8b44516c6feb29;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=aee404565f47ce62
To: <sip:###Handynr###@10.2.1.1>;tag=as68effc23
Call-ID: [email protected]
CSeq: 48438 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Proxy-Authenticate: Digest realm="asterisk", nonce="561b8afb"
Content-Length: 0


---
Scheduling destruction of call '[email protected]' in 15000 ms
Found user '96'
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
ACK sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKcf8b44516c6feb29
From: "Holger Sorg" <sip:[email protected]>;tag=aee404565f47ce62
To: <sip:###Handynr###@10.2.1.1>;tag=as68effc23
Contact: <sip:[email protected]:50696>
Call-ID: [email protected]
CSeq: 48438 ACK
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

--- (11 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
INVITE sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKcaaa334f51128443
From: "Holger Sorg" <sip:[email protected]>;tag=aee404565f47ce62
To: <sip:###Handynr###@10.2.1.1>
Contact: <sip:[email protected]:50696>
Supported: replaces, timer
Proxy-Authorization: Digest username="96", realm="asterisk", algorithm=MD5, uri="sip:###Handynr###@10.2.1.1", nonce="561b8afb", response="2f749a4f83d481867b4281206653e61e"
Call-ID: [email protected]
CSeq: 48439 INVITE
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 255

v=0
o=96 8000 8001 IN IP4 10.2.1.96
s=SIP Call
c=IN IP4 10.2.1.96
t=0 0
m=audio 11096 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (14 headers 13 lines)---
Using INVITE request as basis request - [email protected]
Sending to 10.2.1.96 : 50696 (NAT)
Found user '96'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.2.1.96:11096
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for ###Handynr### in ext-96 (domain 10.2.1.1)
list_route: hop: <sip:[email protected]:50696>
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKcaaa334f51128443;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=aee404565f47ce62
To: <sip:###Handynr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 48439 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Length: 0


---
    -- Executing Macro("SIP/96-d9b8", "dialout|SIP|###Handynr###|@PEER-96||") in new stack
    -- Executing Dial("SIP/96-d9b8", "SIP/###Handynr###@PEER-96||") in new stack
    -- Called ###Handynr###@PEER-96
    -- SIP/PEER-96-9659 is making progress passing it to SIP/96-d9b8
We're at 10.2.1.1 port 14082
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKcaaa334f51128443;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=aee404565f47ce62
To: <sip:###Handynr###@10.2.1.1>;tag=as1ccd2af8
Call-ID: [email protected]
CSeq: 48439 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 6514 6514 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 14082 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- SIP/PEER-96-9659 is ringing
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKcaaa334f51128443;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=aee404565f47ce62
To: <sip:###Handynr###@10.2.1.1>;tag=as1ccd2af8
Call-ID: [email protected]
CSeq: 48439 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Length: 0


---
    -- SIP/PEER-96-9659 is making progress passing it to SIP/96-d9b8
    -- SIP/PEER-96-9659 answered SIP/96-d9b8
We're at 10.2.1.1 port 14082
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKcaaa334f51128443;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=aee404565f47ce62
To: <sip:###Handynr###@10.2.1.1>;tag=as1ccd2af8
Call-ID: [email protected]
CSeq: 48439 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 6514 6515 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 14082 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/96-d9b8 and SIP/PEER-96-9659
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
ACK sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKe26eb5ca916389de
From: "Holger Sorg" <sip:[email protected]>;tag=aee404565f47ce62
To: <sip:###Handynr###@10.2.1.1>;tag=as1ccd2af8
Contact: <sip:[email protected]:50696>
Proxy-Authorization: Digest username="96", realm="asterisk", algorithm=MD5, uri="sip:###Handynr###@10.2.1.1", nonce="561b8afb", response="85313ba846c4802f94ebb02cdb38c9bb"
Call-ID: [email protected]
CSeq: 48439 ACK
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

--- (12 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:

--- (0 headers 0 lines) Nat keepalive ---
tkanlage*CLI>
<-- SIP read from 10.2.1.96:50696:
BYE sip:###Handynr###@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKf7ad5f529c8e8d2f
From: "Holger Sorg" <sip:[email protected]>;tag=aee404565f47ce62
To: <sip:###Handynr###@10.2.1.1>;tag=as1ccd2af8
Proxy-Authorization: Digest username="96", realm="asterisk", algorithm=MD5, uri="sip:###Handynr###@10.2.1.1", nonce="561b8afb", response="f1fda898a23c765b9ee682e62374c140"
Call-ID: [email protected]
CSeq: 48440 BYE
User-Agent: Grandstream GXP2000 1.0.2.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

--- (11 headers 0 lines)---
Sending to 10.2.1.96 : 50696 (NAT)
Transmitting (NAT) to 10.2.1.96:50696:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.96:50696;branch=z9hG4bKf7ad5f529c8e8d2f;received=10.2.1.96
From: "Holger Sorg" <sip:[email protected]>;tag=aee404565f47ce62
To: <sip:###Handynr###@10.2.1.1>;tag=as1ccd2af8
Call-ID: [email protected]
CSeq: 48440 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###Handynr###@10.2.1.1>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
  == Spawn extension (macro-dialout, s, 1) exited non-zero on 'SIP/96-d9b8' in macro 'dialout'
  == Spawn extension (macro-dialout, s, 1) exited non-zero on 'SIP/96-d9b8'
    -- Executing Hangup("SIP/96-d9b8", "") in new stack
  == Spawn extension (macro-dialout, h, 1) exited non-zero on 'SIP/96-d9b8'
Destroying call '[email protected]'
tkanlage*CLI>


und hier mal das debugging von dem peer über den ich anrufe.

Code: 
Connected to Asterisk 1.2.7.1 currently running on tkanlage (pid = 5375)
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
We're at 10.2.1.1 port 18918
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
13 headers, 9 lines
Reliably Transmitting (no NAT) to 213.148.136.10:5060:
INVITE sip:###HandyNr###@sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK39dfcace;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as6d956053
To: <sip:###HandyNr###@sip.qsc.de>
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 12 May 2006 12:33:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 174

v=0
o=root 5376 5376 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 18918 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 100 Trying
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as6d956053
To: <sip:###HandyNr###@sip.qsc.de>
CSeq: 102 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK39dfcace;received=212.202.198.225;rport=5060
Content-Length: 0

--- (7 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 407 Proxy Authentication Required
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as6d956053
To: <sip:###HandyNr###@sip.qsc.de>;tag=ce67d6b4
CSeq: 102 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK39dfcace;received=212.202.198.225;rport=5060
Proxy-Authenticate: Digest realm="qsc.de",nonce="446481c6a483f4d8ca2545484c0f43ab127df8e7",qop="auth"
Server: QSC SIP Router
Content-Length: 0

--- (9 headers 0 lines)---
Transmitting (no NAT) to 213.148.136.10:5060:
ACK sip:###HandyNr###@sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK39dfcace;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as6d956053
To: <sip:###HandyNr###@sip.qsc.de>;tag=ce67d6b4
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
We're at 10.2.1.1 port 18918
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 213.148.136.10:5060:
INVITE sip:###HandyNr###@sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK7cef9810;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as6d956053
To: <sip:###HandyNr###@sip.qsc.de>
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="###QSCNr###", realm="qsc.de", algorithm=MD5, uri="sip:###HandyNr###@sip.qsc.de", nonce="446481c6a483f4d8ca2545484c0f43ab127df8e7", response="a7e3a1a38d298c820ede77dbd241ddb7", opaque="", qop=auth, cnonce="18d72644", nc=00000001
Date: Fri, 12 May 2006 12:33:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 174

v=0
o=root 5376 5377 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 18918 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 100 Trying
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as6d956053
To: <sip:###HandyNr###@sip.qsc.de>
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK7cef9810;received=212.202.198.225;rport=5060
Content-Length: 0

--- (7 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 183 Session Progress
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as6d956053
To: <sip:###HandyNr###@sip.qsc.de>;tag=42789519
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK7cef9810;received=212.202.198.225;rport=5060
Contact: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
Content-Length: 150
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 539888 539888 IN IP4 213.148.135.2
s=Sip Call
c=IN IP4 213.148.136.10
t=0 0
m=audio 34272 RTP/AVP 8
a=rtpmap:8 PCMA/8000
--- (9 headers 7 lines)---
Found RTP audio format 8
Peer audio RTP is at port 213.148.136.10:34272
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 180 Ringing
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as6d956053
To: <sip:###HandyNr###@sip.qsc.de>;tag=42789519
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK7cef9810;received=212.202.198.225;rport=5060
Contact: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
Content-Length: 150
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 539888 539889 IN IP4 213.148.135.2
s=Sip Call
c=IN IP4 213.148.136.10
t=0 0
m=audio 34272 RTP/AVP 8
a=rtpmap:8 PCMA/8000
--- (9 headers 7 lines)---
Found RTP audio format 8
Peer audio RTP is at port 213.148.136.10:34272
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 200 OK
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as6d956053
To: <sip:###HandyNr###@sip.qsc.de>;tag=42789519
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK7cef9810;received=212.202.198.225;rport=5060
Contact: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
Content-Length: 150
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 539888 539890 IN IP4 213.148.135.2
s=Sip Call
c=IN IP4 213.148.136.10
t=0 0
m=audio 34272 RTP/AVP 8
a=rtpmap:8 PCMA/8000
--- (9 headers 7 lines)---
Found RTP audio format 8
Peer audio RTP is at port 213.148.136.10:34272
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
set_destination: Parsing <sip:###HandyNr###@213.148.136.10:5060;user=phone> for address/port to send to
set_destination: set destination to 213.148.136.10, port 5060
Transmitting (no NAT) to 213.148.136.10:5060:
ACK sip:###HandyNr###@213.148.136.10:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK11fee61d;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as6d956053
To: <sip:###HandyNr###@sip.qsc.de>;tag=42789519
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:###HandyNr###@213.148.136.10:5060;user=phone> for address/port to send to
set_destination: set destination to 213.148.136.10, port 5060
Reliably Transmitting (no NAT) to 213.148.136.10:5060:
BYE sip:###HandyNr###@213.148.136.10:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK26a381e2;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as6d956053
To: <sip:###HandyNr###@sip.qsc.de>;tag=42789519
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="###QSCNr###", realm="qsc.de", algorithm=MD5, uri="sip:###HandyNr###@213.148.136.10:5060", nonce="446481c6a483f4d8ca2545484c0f43ab127df8e7", response="344add028224a59884e08f5181372873", opaque="", qop=auth, cnonce="7f80ed00", nc=00000002
Content-Length: 0


---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 200 OK
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as6d956053
To: <sip:###HandyNr###@sip.qsc.de>;tag=42789519
CSeq: 104 BYE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK26a381e2;received=212.202.198.225;rport=5060
Content-Length: 0

--- (7 headers 0 lines)---
Destroying call '[email protected]'
We're at 10.2.1.1 port 14024
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
13 headers, 9 lines
Reliably Transmitting (no NAT) to 213.148.136.10:5060:
INVITE sip:###HandyNr###@sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK22dac777;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
To: <sip:###HandyNr###@sip.qsc.de>
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 12 May 2006 12:33:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 174

v=0
o=root 5376 5376 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 14024 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 100 Trying
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
To: <sip:###HandyNr###@sip.qsc.de>
CSeq: 102 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK22dac777;received=212.202.198.225;rport=5060
Content-Length: 0

--- (7 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 407 Proxy Authentication Required
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
To: <sip:###HandyNr###@sip.qsc.de>;tag=c6eeaf73
CSeq: 102 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK22dac777;received=212.202.198.225;rport=5060
Proxy-Authenticate: Digest realm="qsc.de",nonce="446481d455046ce30f226ff783c947d89a3097a3",qop="auth"
Server: QSC SIP Router
Content-Length: 0

--- (9 headers 0 lines)---
Transmitting (no NAT) to 213.148.136.10:5060:
ACK sip:###HandyNr###@sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK22dac777;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
To: <sip:###HandyNr###@sip.qsc.de>;tag=c6eeaf73
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
We're at 10.2.1.1 port 14024
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 213.148.136.10:5060:
INVITE sip:###HandyNr###@sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK359b543d;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
To: <sip:###HandyNr###@sip.qsc.de>
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="###QSCNr###", realm="qsc.de", algorithm=MD5, uri="sip:###HandyNr###@sip.qsc.de", nonce="446481d455046ce30f226ff783c947d89a3097a3", response="46b947aaad8bf9e6d5314da7eb2b88c1", opaque="", qop=auth, cnonce="7d0899b3", nc=00000001
Date: Fri, 12 May 2006 12:33:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 174

v=0
o=root 5376 5377 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 14024 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 100 Trying
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
To: <sip:###HandyNr###@sip.qsc.de>
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK359b543d;received=212.202.198.225;rport=5060
Content-Length: 0

--- (7 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 183 Session Progress
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
To: <sip:###HandyNr###@sip.qsc.de>;tag=62c7bf67
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK359b543d;received=212.202.198.225;rport=5060
Contact: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
Content-Length: 150
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 540004 540004 IN IP4 213.148.135.2
s=Sip Call
c=IN IP4 213.148.136.10
t=0 0
m=audio 34320 RTP/AVP 8
a=rtpmap:8 PCMA/8000
--- (9 headers 7 lines)---
Found RTP audio format 8
Peer audio RTP is at port 213.148.136.10:34320
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 183 Session Progress
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
To: <sip:###HandyNr###@sip.qsc.de>;tag=62c7bf67
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK359b543d;received=212.202.198.225;rport=5060
Contact: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
Content-Length: 150
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 540004 540005 IN IP4 213.148.135.2
s=Sip Call
c=IN IP4 213.148.136.10
t=0 0
m=audio 34320 RTP/AVP 8
a=rtpmap:8 PCMA/8000
--- (9 headers 7 lines)---
Found RTP audio format 8
Peer audio RTP is at port 213.148.136.10:34320
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 180 Ringing
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
To: <sip:###HandyNr###@sip.qsc.de>;tag=62c7bf67
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK359b543d;received=212.202.198.225;rport=5060
Contact: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
Content-Length: 150
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 540004 540006 IN IP4 213.148.135.2
s=Sip Call
c=IN IP4 213.148.136.10
t=0 0
m=audio 34320 RTP/AVP 8
a=rtpmap:8 PCMA/8000
--- (9 headers 7 lines)---
Found RTP audio format 8
Peer audio RTP is at port 213.148.136.10:34320
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 200 OK
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
To: <sip:###HandyNr###@sip.qsc.de>;tag=62c7bf67
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK359b543d;received=212.202.198.225;rport=5060
Contact: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
Content-Length: 150
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 540004 540007 IN IP4 213.148.135.2
s=Sip Call
c=IN IP4 213.148.136.10
t=0 0
m=audio 34320 RTP/AVP 8
a=rtpmap:8 PCMA/8000
--- (9 headers 7 lines)---
Found RTP audio format 8
Peer audio RTP is at port 213.148.136.10:34320
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
set_destination: Parsing <sip:###HandyNr###@213.148.136.10:5060;user=phone> for address/port to send to
set_destination: set destination to 213.148.136.10, port 5060
Transmitting (no NAT) to 213.148.136.10:5060:
ACK sip:###HandyNr###@213.148.136.10:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK38ebdc42;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
To: <sip:###HandyNr###@sip.qsc.de>;tag=62c7bf67
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
BYE sip:###QSCNr###@10.2.1.1:5060;user=phone SIP/2.0
From: <sip:###HandyNr###@sip.qsc.de>;tag=62c7bf67
To: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
CSeq: 1 BYE
Call-ID: [email protected]
Via: SIP/2.0/UDP 213.148.136.10:5060;branch=z9hG4bK6521e96b4
Max-Forwards: 16
Reason: Q.850;cause=16;text="normal call clearing"
Content-Length: 0

--- (9 headers 0 lines)---
Sending to 213.148.136.10 : 5060 (non-NAT)
Transmitting (no NAT) to 213.148.136.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.148.136.10:5060;branch=z9hG4bK6521e96b4;received=213.148.136.10
From: <sip:###HandyNr###@sip.qsc.de>;tag=62c7bf67
To: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as3b928357
Call-ID: [email protected]
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:###QSCNr###@10.2.1.1>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
Destroying call '[email protected]'
May 12 14:33:59 NOTICE[5604]: chan_sip.c:5259 sip_reregister:    -- Re-registration for  [email protected]
May 12 14:33:59 NOTICE[5604]: chan_sip.c:5259 sip_reregister:    -- Re-registration for  [email protected]
May 12 14:33:59 NOTICE[5604]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sipgate.de is 120 sec (Scheduling reregistration in 105 s)
May 12 14:33:59 NOTICE[5604]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sipgate.de is 120 sec (Scheduling reregistration in 105 s)
May 12 14:33:59 NOTICE[5604]: chan_sip.c:5259 sip_reregister:    -- Re-registration for  [email protected]
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 213.148.136.10:5060:
REGISTER sip:sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK2c3fff3d;rport
From: <sip:[email protected]>;tag=as5a14761d
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="03081450698", realm="qsc.de", algorithm=MD5, uri="sip:sip.qsc.de", nonce="44648111886cc3bfcbb22d56b35cace45458759d", response="c82ad8d1ff43a58faa9a6d74ea7d067b", opaque="", qop=auth, cnonce="589b8670", nc=00000002
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 200 OK
From: <sip:[email protected]>;tag=as5a14761d
To: <sip:[email protected]>;tag=17bbb0c4
CSeq: 105 REGISTER
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK2c3fff3d;received=212.202.198.225;rport=5060
Server: QSC SIP Router
Contact: <sip:[email protected]:5060;user=phone>;expires=300
Content-Length: 0

--- (9 headers 0 lines)---
Scheduling destruction of call '[email protected]' in 32000 ms
May 12 14:33:59 NOTICE[5604]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sip.qsc.de is 120 sec (Scheduling reregistration in 105 s)
May 12 14:33:59 NOTICE[5604]: chan_sip.c:5259 sip_reregister:    -- Re-registration for  [email protected]
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 213.148.136.10:5060:
REGISTER sip:sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK1d2b3fe0;rport
From: <sip:[email protected]>;tag=as644e4d59
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="03081450697", realm="qsc.de", algorithm=MD5, uri="sip:sip.qsc.de", nonce="44648111886cc3bfcbb22d56b35cace45458759d", response="18ed3e90ec6d4f5d619e5190c2ac845e", opaque="", qop=auth, cnonce="0d99b238", nc=00000002
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---
May 12 14:33:59 NOTICE[5604]: chan_sip.c:5259 sip_reregister:    -- Re-registration for  ###QSCNr###@sip.qsc.de
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 213.148.136.10:5060:
REGISTER sip:sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK7021181f;rport
From: <sip:###QSCNr###@sip.qsc.de>;tag=as474fb881
To: <sip:###QSCNr###@sip.qsc.de>
Call-ID: [email protected]
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="###QSCNr###", realm="qsc.de", algorithm=MD5, uri="sip:sip.qsc.de", nonce="44648111886cc3bfcbb22d56b35cace45458759d", response="0790a24558c10f8bb53cd2d97f33f97e", opaque="", qop=auth, cnonce="166d2138", nc=00000002
Expires: 120
Contact: <sip:###QSCNr###@10.2.1.1>
Event: registration
Content-Length: 0


---
May 12 14:33:59 NOTICE[5604]: chan_sip.c:5259 sip_reregister:    -- Re-registration for  [email protected]
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 213.148.136.10:5060:
REGISTER sip:sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK6bc6fca5;rport
From: <sip:[email protected]>;tag=as4a38845e
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="03081450695", realm="qsc.de", algorithm=MD5, uri="sip:sip.qsc.de", nonce="44648111886cc3bfcbb22d56b35cace45458759d", response="88ca82a27f57b5c474f816c3a77fd150", opaque="", qop=auth, cnonce="047ebc89", nc=00000002
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---

<-- SIP read from 213.148.136.10:5060:
SIP/2.0 200 OK
From: <sip:[email protected]>;tag=as644e4d59
To: <sip:[email protected]>;tag=3a0b5aba
CSeq: 105 REGISTER
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK1d2b3fe0;received=212.202.198.225;rport=5060
Server: QSC SIP Router
Contact: <sip:[email protected]:5060;user=phone>;expires=300
Content-Length: 0

--- (9 headers 0 lines)---
Scheduling destruction of call '[email protected]' in 32000 ms
May 12 14:33:59 NOTICE[5604]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sip.qsc.de is 120 sec (Scheduling reregistration in 105 s)

<-- SIP read from 213.148.136.10:5060:
SIP/2.0 200 OK
From: <sip:###QSCNr###@sip.qsc.de>;tag=as474fb881
To: <sip:###QSCNr###@sip.qsc.de>;tag=1310b6d0
CSeq: 105 REGISTER
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK7021181f;received=212.202.198.225;rport=5060
Server: QSC SIP Router
Contact: <sip:###QSCNr###@10.2.1.1:5060;user=phone>;expires=300
Content-Length: 0

--- (9 headers 0 lines)---
Scheduling destruction of call '[email protected]' in 32000 ms
May 12 14:33:59 NOTICE[5604]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sip.qsc.de is 120 sec (Scheduling reregistration in 105 s)
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 200 OK
From: <sip:[email protected]>;tag=as4a38845e
To: <sip:[email protected]>;tag=e7f10f94
CSeq: 105 REGISTER
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK6bc6fca5;received=212.202.198.225;rport=5060
Server: QSC SIP Router
Contact: <sip:[email protected]:5060;user=phone>;expires=300
Content-Length: 0

--- (9 headers 0 lines)---
Scheduling destruction of call '[email protected]' in 32000 ms
May 12 14:33:59 NOTICE[5604]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sip.qsc.de is 120 sec (Scheduling reregistration in 105 s)
We're at 10.2.1.1 port 18122
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
13 headers, 9 lines
Reliably Transmitting (no NAT) to 213.148.136.10:5060:
INVITE sip:###HandyNr###@sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK0022f866;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 12 May 2006 12:34:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 174

v=0
o=root 5376 5376 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 18122 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 100 Trying
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>
CSeq: 102 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK0022f866;received=212.202.198.225;rport=5060
Content-Length: 0

--- (7 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 407 Proxy Authentication Required
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>;tag=711940e7
CSeq: 102 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK0022f866;received=212.202.198.225;rport=5060
Proxy-Authenticate: Digest realm="qsc.de",nonce="446481e7fe247949f54815a1bbe84c101b14427b",qop="auth"
Server: QSC SIP Router
Content-Length: 0

--- (9 headers 0 lines)---
Transmitting (no NAT) to 213.148.136.10:5060:
ACK sip:###HandyNr###@sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK0022f866;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>;tag=711940e7
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
We're at 10.2.1.1 port 18122
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 213.148.136.10:5060:
INVITE sip:###HandyNr###@sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK4fbed055;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="###QSCNr###", realm="qsc.de", algorithm=MD5, uri="sip:###HandyNr###@sip.qsc.de", nonce="446481e7fe247949f54815a1bbe84c101b14427b", response="ac497a21636be9971f9765fe57d909b0", opaque="", qop=auth, cnonce="74e53cf1", nc=00000001
Date: Fri, 12 May 2006 12:34:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 174

v=0
o=root 5376 5377 IN IP4 10.2.1.1
s=session
c=IN IP4 10.2.1.1
t=0 0
m=audio 18122 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 100 Trying
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK4fbed055;received=212.202.198.225;rport=5060
Content-Length: 0

--- (7 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 183 Session Progress
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>;tag=2204f9b4
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK4fbed055;received=212.202.198.225;rport=5060
Contact: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
Content-Length: 150
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 540202 540202 IN IP4 213.148.135.2
s=Sip Call
c=IN IP4 213.148.136.10
t=0 0
m=audio 34416 RTP/AVP 8
a=rtpmap:8 PCMA/8000
--- (9 headers 7 lines)---
Found RTP audio format 8
Peer audio RTP is at port 213.148.136.10:34416
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 183 Session Progress
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>;tag=2204f9b4
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK4fbed055;received=212.202.198.225;rport=5060
Contact: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
Content-Length: 150
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 540202 540203 IN IP4 213.148.135.2
s=Sip Call
c=IN IP4 213.148.136.10
t=0 0
m=audio 34416 RTP/AVP 8
a=rtpmap:8 PCMA/8000
--- (9 headers 7 lines)---
Found RTP audio format 8
Peer audio RTP is at port 213.148.136.10:34416
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 180 Ringing
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>;tag=2204f9b4
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK4fbed055;received=212.202.198.225;rport=5060
Contact: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
Content-Length: 150
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 540202 540204 IN IP4 213.148.135.2
s=Sip Call
c=IN IP4 213.148.136.10
t=0 0
m=audio 34416 RTP/AVP 8
a=rtpmap:8 PCMA/8000
--- (9 headers 7 lines)---
Found RTP audio format 8
Peer audio RTP is at port 213.148.136.10:34416
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 200 OK
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>;tag=2204f9b4
CSeq: 103 INVITE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK4fbed055;received=212.202.198.225;rport=5060
Contact: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
Content-Length: 150
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 540202 540205 IN IP4 213.148.135.2
s=Sip Call
c=IN IP4 213.148.136.10
t=0 0
m=audio 34416 RTP/AVP 8
a=rtpmap:8 PCMA/8000
--- (9 headers 7 lines)---
Found RTP audio format 8
Peer audio RTP is at port 213.148.136.10:34416
Found description format PCMA
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:###HandyNr###@213.148.136.10:5060;user=phone>
set_destination: Parsing <sip:###HandyNr###@213.148.136.10:5060;user=phone> for address/port to send to
set_destination: set destination to 213.148.136.10, port 5060
Transmitting (no NAT) to 213.148.136.10:5060:
ACK sip:###HandyNr###@213.148.136.10:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK55896305;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>;tag=2204f9b4
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:###HandyNr###@213.148.136.10:5060;user=phone> for address/port to send to
set_destination: set destination to 213.148.136.10, port 5060
Reliably Transmitting (no NAT) to 213.148.136.10:5060:
BYE sip:###HandyNr###@213.148.136.10:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK31e23306;rport
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>;tag=2204f9b4
Contact: <sip:###QSCNr###@10.2.1.1>
Call-ID: [email protected]
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="###QSCNr###", realm="qsc.de", algorithm=MD5, uri="sip:###HandyNr###@213.148.136.10:5060", nonce="446481e7fe247949f54815a1bbe84c101b14427b", response="5cf8c1c7f88326b584ee6dd95661fee9", opaque="", qop=auth, cnonce="6bac0625", nc=00000002
Content-Length: 0


---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
SIP/2.0 200 OK
From: "Holger Sorg" <sip:###QSCNr###@10.2.1.1>;tag=as129a9dee
To: <sip:###HandyNr###@sip.qsc.de>;tag=2204f9b4
CSeq: 104 BYE
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK31e23306;received=212.202.198.225;rport=5060
Content-Length: 0

--- (7 headers 0 lines)---
Destroying call '[email protected]'
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI>
<-- SIP read from 213.148.136.10:5060:
hello
--- (1 headers 0 lines)---
tkanlage*CLI> sip no debug
SIP Debugging Disabled
tkanlage*CLI>

hier ging übrigens der erste ohne Probleme und die restlichen hörte man mich nicht.

Hier noch meine sip.conf

Code: 
[general]
context=sip-incomming
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
language=de
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
pedantic=yes


[96]
callerid=Holger Sorg <96>
defaultip=10.2.1.96
host=dynamic
port=50696
user=96
secret=*****
type=friend
mailbox=96
nat=no
canreinvite=no
language=de
qualify=yes
context=ext-96
dtmfmode=rfc2833

[PEER-96]
username=###QSCNr###
type=peer
secret=*****
nat=no
insecure=very
host=sip.qsc.de
fromuser=###QSCNr###
from=###QSCNr###
dtmfmode=inband
canreinvite=no
authname=###QSCNr###@sip.qsc.de

[edit]
Ich habe das gerade mal mit meinem sipagte Account versucht, da hat ich das Problem nicht. Liegt das dann wieder an dem SER von QSC oder gibt es wieder Einstellungen, die man machen muss, von denen man aber nichts weiß.



Hilfe ;o)
 
Zuletzt bearbeitet:
Hallo Holg,

ich habe fast das selbe Problem aber als Dauerzustand auf einem V-Server. Es sind nur SIP-Gespräche betroffen.
Allem Anschein nach ist die Strecke Asterisk internes Sip-Endgerät betroffen.
NAT Probleme sind es keine, da auch Telefone mit öffentlicher ip die direkt am * hängen betroffen sind.
 
Um antworten zu können musst du eingeloggt sein.

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