<— SIP read from UDP:192.168.1.184:5060 —>
INVITE sip:2000@192.168.1.236 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.184:5060;branch=z9hG4bK-828c194b
From: “Lars Thomsen” sip:user1@192.168.1.236;tag=1ca0c13b5588a3cbo0
To: “user2” sip:2000@192.168.1.236
Call-ID: b8bc61fb-6fd37a0b@192.168.1.184
CSeq: 101 INVITE
Max-Forwards: 70
Contact: “Lars Thomsen” sip:user1@192.168.1.184:5060
Expires: 240
User-Agent: Linksys/SPA942-4.1.12
Content-Length: 397
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 604491 604491 IN IP4 192.168.1.184
s=-
c=IN IP4 192.168.1.184
t=0 0
m=audio 16464 RTP/AVP 8 0 2 4 18 96 97 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
— (13 headers 18 lines) —
== Using SIP RTP CoS mark 5
Sending to 192.168.1.184 : 5060 (no NAT)
Using INVITE request as basis request - b8bc61fb-6fd37a0b@192.168.1.184
Found peer ‘user1’ for ‘user1’ from 192.168.1.184:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format G729a for ID 18
Found audio description format G726-40 for ID 96
Found audio description format G726-24 for ID 97
Found audio description format G726-16 for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.184:16464
Looking for 2000 in default (domain 192.168.1.236)
list_route: hop: sip:user1@192.168.1.184:5060
<— Transmitting (no NAT) to 192.168.1.184:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.184:5060;branch=z9hG4bK-828c194b;received=192.168.1.184
From: “Lars Thomsen” sip:user1@192.168.1.236;tag=1ca0c13b5588a3cbo0
To: “user2” sip:2000@192.168.1.236
Call-ID: b8bc61fb-6fd37a0b@192.168.1.184
CSeq: 101 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:2000@192.168.1.236
Content-Length: 0
<------------>
– Executing Dial(“SIP/user1-0000001a”, “SIP/user2,25,Ttr”)
== Using SIP RTP CoS mark 5
Audio is at 192.168.1.236 port 16580
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.197:5060:
INVITE sip:192.168.1.197:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.236:5060;branch=z9hG4bK220eb3e7;rport
Max-Forwards: 70
From: “user1” sip:1000@192.168.1.236;tag=as43d6b6f3
To: sip:192.168.1.197:5060
Contact: sip:1000@192.168.1.236
Call-ID: 27c03f0825c2f8346959aced164cf116@192.168.1.236
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4
Date: Fri, 01 Jun 2012 11:36:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 511
v=0
o=root 1323039673 1323039673 IN IP4 192.168.1.236
s=Asterisk PBX 1.6.2.5-0ubuntu1.4
c=IN IP4 192.168.1.236
t=0 0
m=audio 16580 RTP/AVP 0 3 8 112 5 10 7 110 2 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
-- Called user2
communica*CLI>
<— Transmitting (no NAT) to 192.168.1.184:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.184:5060;branch=z9hG4bK-828c194b;received=192.168.1.184
From: “Lars Thomsen” sip:user1@192.168.1.236;tag=1ca0c13b5588a3cbo0
To: “user2” sip:2000@192.168.1.236;tag=as29b01e76
Call-ID: b8bc61fb-6fd37a0b@192.168.1.184
CSeq: 101 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:2000@192.168.1.236
Content-Length: 0
<------------>
communica*CLI>
<— SIP read from UDP:192.168.1.197:5060 —>
SIP/2.0 404 Not Found
To: sip:192.168.1.197:5060;tag=fdc91f28e9f7c3di0
From: “user1” sip:1000@192.168.1.236;tag=as43d6b6f3
Call-ID: 27c03f0825c2f8346959aced164cf116@192.168.1.236
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.236:5060;branch=z9hG4bK220eb3e7
Server: Linksys/SPA922-5.1.5
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Transmitting (no NAT) to 192.168.1.197:5060:
ACK sip:192.168.1.197:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.236:5060;branch=z9hG4bK220eb3e7;rport
Max-Forwards: 70
From: “user1” sip:1000@192.168.1.236;tag=as43d6b6f3
To: sip:192.168.1.197:5060;tag=fdc91f28e9f7c3di0
Contact: sip:1000@192.168.1.236
Call-ID: 27c03f0825c2f8346959aced164cf116@192.168.1.236
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4
Content-Length: 0
-- SIP/user2-0000001b is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/user1-0000001a’ status is ‘CONGESTION’
<— Reliably Transmitting (no NAT) to 192.168.1.184:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.184:5060;branch=z9hG4bK-828c194b;received=192.168.1.184
From: “Lars Thomsen” sip:user1@192.168.1.236;tag=1ca0c13b5588a3cbo0
To: “user2” sip:2000@192.168.1.236;tag=as29b01e76
Call-ID: b8bc61fb-6fd37a0b@192.168.1.184
CSeq: 101 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1