-- Attempting call on SIP/030613xxxxx@gmx_out for s@playall:1 (Retry 1)
We're at 85.25.xx.xx port 4690
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 212.227.15.197:5060:
INVITE sip:
[email protected] SIP/2.0
Via: SIP/2.0/UDP 85.25.xx.xx:5600;branch=z9hG4bK4c0e86ef;rport
From: "asterisk" <sip:
[email protected]>;tag=as6abb1249
To: <sip:
[email protected]>
Contact: <sip:
[email protected]:5600>
Call-ID:
[email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Date: Sun, 23 Jul 2006 18:48:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 7486 7486 IN IP4 85.25.xx.xx
s=session
c=IN IP4 85.25.xx.xx
t=0 0
m=audio 4690 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp
ff - - - -
---
vs2053031*CLI>
<-- SIP read from 212.227.15.197:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 85.25.xx.xx:5600;branch=z9hG4bK4c0e86ef;rport=5600
From: "asterisk" <sip:
[email protected]>;tag=as6abb1249
To: <sip:
[email protected]>;tag=329cfeaa6ded039da25ff8cbb8668bd2.bd5e
Call-ID:
[email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip-gmx.net", nonce="44c3c5b98e19c87402b43cd147c90e255cf33af3"
Server: UI OpenSer
Content-Length: 0
--- (9 headers 0 lines)---
Transmitting (no NAT) to 212.227.15.197:5060:
ACK sip:
[email protected] SIP/2.0
Via: SIP/2.0/UDP 85.25.xx.xx:5600;branch=z9hG4bK4c0e86ef;rport
From: "asterisk" <sip:
[email protected]>;tag=as6abb1249
To: <sip:
[email protected]>;tag=329cfeaa6ded039da25ff8cbb8668bd2.bd5e
Contact: <sip:
[email protected]:5600>
Call-ID:
[email protected]
CSeq: 102 ACK
Max-Forwards: 70
Content-Length: 0
---
We're at 85.25.xx.xx port 4690
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 212.227.15.197:5060:
INVITE sip:
[email protected] SIP/2.0
Via: SIP/2.0/UDP 85.25.xx.xx:5600;branch=z9hG4bK430d291f;rport
From: "asterisk" <sip:
[email protected]>;tag=as6abb1249
To: <sip:
[email protected]>
Contact: <sip:
[email protected]:5600>
Call-ID:
[email protected]
CSeq: 103 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="4930381xxxxx", realm="sip-gmx.net", algorithm=MD5, uri="sip:
[email protected]", nonce="44c3c5b98e19c87402b43cd147c90e255cf33af3", response="84705ecf6b2f642c5397f78d203a60c9", opaque=""
Date: Sun, 23 Jul 2006 18:48:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 7486 7487 IN IP4 85.25.xx.xx
s=session
c=IN IP4 85.25.xx.xx
t=0 0
m=audio 4690 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp
ff - - - -
---
vs2053031*CLI>
<-- SIP read from 212.227.15.197:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 85.25.xx.xx:5600;branch=z9hG4bK430d291f;rport=5600
From: "asterisk" <sip:
[email protected]>;tag=as6abb1249
To: <sip:
[email protected]>
Call-ID:
[email protected]
CSeq: 103 INVITE
Server: UI OpenSer
Content-Length: 0
--- (8 headers 0 lines)---
vs2053031*CLI>
<-- SIP read from 212.227.15.197:5060:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 85.25.xx.xx:5600;branch=z9hG4bK430d291f;rport=5600
From: "asterisk" <sip:
[email protected]>;tag=as6abb1249
To: <sip:
[email protected]>;tag=B177BA3232DA6CD977D7746E73F2C
Call-ID:
[email protected]
CSeq: 103 INVITE
User-Agent: AVM FRITZ!Box Fon 5010 (UI) 23.04.01 (Jan 25 2006)
Content-Type: application/sdp
Content-Length: 375
X-Route-Info: IP
v=0
o=user 2532002 2532002 IN IP4 84.190.9.215
s=call
c=IN IP4 84.190.9.215
t=1153680557 1153684157
m=audio 7082 RTP/AVP 8 0 2 102 100 99 97 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtcp:7083
--- (10 headers 16 lines)---
-- Got SIP response 488 "Not Acceptable Here" back from 212.227.15.197
Transmitting (no NAT) to 212.227.15.197:5060:
ACK sip:
[email protected] SIP/2.0
Via: SIP/2.0/UDP 85.25.xx.xx:5600;branch=z9hG4bK430d291f;rport
From: "asterisk" <sip:
[email protected]>;tag=as6abb1249
To: <sip:
[email protected]>;tag=B177BA3232DA6CD977D7746E73F2C
Contact: <sip:
[email protected]:5600>
Call-ID:
[email protected]
CSeq: 103 ACK
Max-Forwards: 70
Content-Length: 0
---
> Channel SIP/gmx_out-ed9d was never answered.
Jul 23 20:48:48 WARNING[30152]: cdr.c:550 ast_cdr_disposition: Cause not handled
Destroying call '
[email protected]'
vs2053031*CLI>