All pastes #353866 Raw Edit

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public text v1 · immutable
·published 2007-02-13 17:04 UTC
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mayamini-lx*CLI>
<--- SIP read from 10.2.110.195:5060 --->
INVITE sip:1234@192.168.9.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.110.195:5060;branch=z9hG4bK4fe9cdf7;rport
From: "foo" <sip:86@10.2.110.195>;tag=as2c33c35f
To: <sip:1234@192.168.9.1>
Contact: <sip:86@10.2.110.195>
Call-ID: 346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195
CSeq: 102 INVITE
User-Agent: 3000
Max-Forwards: 70
Date: Tue, 13 Feb 2007 17:00:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 1845 1845 IN IP4 10.2.119.177
s=session
c=IN IP4 10.2.119.177
t=0 0
m=audio 11388 RTP/AVP 8 111 101
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (14 headers 13 lines) ---
Sending to 10.2.110.195 : 5060 (NAT)
Using INVITE request as basis request - 346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195
Found user '86'
Found RTP audio format 8
Found RTP audio format 111
Found RTP audio format 101
Peer audio RTP is at port 10.2.119.177:11388
Found description format PCMA for ID 8
Found description format G726-32 for ID 111
Found description format telephone-event for ID 101
Capabilities: us - 0x80c0e (gsm|ulaw|alaw|g726|ilbc|h263), peer - audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.2.119.177:11388
Looking for 1234 in default (domain 192.168.9.1)
list_route: hop: <sip:86@10.2.110.195>

<--- Transmitting (no NAT) to 10.2.110.195:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.110.195:5060;branch=z9hG4bK4fe9cdf7;received=10.2.110.195;rport=5060
From: "foo" <sip:86@10.2.110.195>;tag=as2c33c35f
To: <sip:1234@192.168.9.1>
Call-ID: 346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1234@192.168.4.109>
Content-Length: 0


<------------>
    -- Executing [1234@default:1] DumpChan("SIP/86-081b3ee0", "") in new stack

Dumping Info For Channel: SIP/86-081b3ee0:
================================================================================
Info:
Name=               SIP/86-081b3ee0
Type=               SIP
UniqueID=           1171386124.7
CallerID=           86
CallerIDName=       foo
DNIDDigits=         1234
RDNIS=              (N/A)
State=              Ring (4)
Rings=              0
NativeFormat=       0x800 (g726)
WriteFormat=        0x800 (g726)
ReadFormat=         0x800 (g726)
1stFileDescriptor=  20
Framesin=           0
Framesout=          0
TimetoHangup=       0
ElapsedTime=        0h0m0s
Context=            default
Extension=          1234
Priority=           1
CallGroup=
PickupGroup=
Application=        DumpChan
Data=               (Empty)
Blocking_in=        (Not Blocking)

Variables:
SIPCALLID=346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195
SIPUSERAGENT=Mediatrix 3000
SIPDOMAIN=192.168.9.1
SIPURI=sip:86@10.2.110.195
================================================================================
    -- Executing [1234@default:2] MeetMe("SIP/86-081b3ee0", "1234") in new stack
[Feb 13 12:02:04] WARNING[3603]: app_meetme.c:2181 conf_exec: cant transcode chan(SIP/86-081b3ee0) since his nativeformats is 0x800 (g726) (2048)
  == Spawn extension (default, 1234, 2) exited non-zero on 'SIP/86-081b3ee0'
Scheduling destruction of SIP dialog '346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 10.2.110.195:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 10.2.110.195:5060;branch=z9hG4bK4fe9cdf7;received=10.2.110.195;rport=5060
From: "foo" <sip:86@10.2.110.195>;tag=as2c33c35f
To: <sip:1234@192.168.9.1>;tag=as00440db7
Call-ID: 346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1234@192.168.4.109>
Content-Length: 0


<------------>
mayamini-lx*CLI>
<--- SIP read from 10.2.110.195:5060 --->
ACK sip:1234@192.168.9.1 SIP/2.0
Via: SIP/2.0/UDP 10.2.110.195:5060;branch=z9hG4bK4fe9cdf7;rport
From: "foo" <sip:86@10.2.110.195>;tag=as2c33c35f
To: <sip:1234@192.168.9.1>;tag=as00440db7
Contact: <sip:86@10.2.110.195>
Call-ID: 346fa2993c7e09b77ea9e7d41f2e4312@10.2.110.195
CSeq: 102 ACK
User-Agent: 3000
Max-Forwards: 70
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
mayamini-lx*CLI>