Audio is at 217.69.212.124 port 24952
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 95.128.224.36:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 217.69.212.124:5060;branch=z9hG4bK1c24a568;rport
Max-Forwards: 70
From: "test" <sip:[email protected]>;tag=as4bfd4deb
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Thu, 15 Dec 2011 13:18:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 2013526211 2013526211 IN IP4 217.69.212.124
s=Asterisk PBX 1.6.0.10-FONCORE-r40
c=IN IP4 217.69.212.124
t=0 0
m=audio 24952 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called mcm/89265308358
trixbox*CLI>
<--- SIP read from UDP://95.128.224.36:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.69.212.124:5060;branch=z9hG4bK1c24a568;rport=5060;received=217.69.212.124
From: "test" <sip:[email protected]>;tag=as4bfd4deb
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: MERA MVTS3G v.4.3.0-22
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
trixbox*CLI>
<--- SIP read from UDP://95.128.224.36:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.69.212.124:5060;branch=z9hG4bK1c24a568;rport=5060;received=217.69.212.124
From: "test" <sip:[email protected]>;tag=as4bfd4deb
To: <sip:[email protected]>;tag=1208231724-3775995687-620760967-3806017680
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: MERA MVTS3G v.4.3.0-22
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/mcm-b7419fe0 is ringing
trixbox*CLI>
<--- SIP read from UDP://95.128.224.36:5060 --->
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 95.128.224.36:5060;rport;branch=z9hG4bK-1849250864-3775995687-620761735-38060176801
Via: SIP/2.0/UDP 95.128.224.36:5061;rport=5061;branch=z9hG4bK-1849250864-3775995687-620761735-3806017680;received=95.128.224.36
From: <sip:[email protected]>;tag=1208231724-3775995687-620760967-3806017680
To: "test" <sip:[email protected]>;tag=as4bfd4deb
Call-ID: [email protected]
CSeq: 103 OPTIONS
Contact: <sip:[email protected]:5060>
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE
Accept: application/dtmf-relay
Accept: application/isup
Accept: application/sdp
Supported: 100rel
Max-Forwards: 70
User-Agent: MERA MVTS3G v.4.3.0-22
Content-Length: 0


<------------->
--- (16 headers 0 lines) ---
trixbox*CLI>
<--- Transmitting (no NAT) to 95.128.224.36:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 95.128.224.36:5060;branch=z9hG4bK-1849250864-3775995687-620761735-38060176801;received=95.128.224.36;rport=5060
Via: SIP/2.0/UDP 95.128.224.36:5061;rport=5061;branch=z9hG4bK-1849250864-3775995687-620761735-3806017680;received=95.128.224.36
From: <sip:[email protected]>;tag=1208231724-3775995687-620760967-3806017680
To: "test" <sip:[email protected]>;tag=as4bfd4deb
Call-ID: [email protected]
CSeq: 103 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:[email protected]>
Accept: application/sdp
Content-Length: 0


<------------>
trixbox*CLI>
<--- SIP read from UDP://95.128.224.36:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.69.212.124:5060;branch=z9hG4bK1c24a568;rport=5060;received=217.69.212.124
From: "test" <sip:[email protected]>;tag=as4bfd4deb
To: <sip:[email protected]>;tag=1208231724-3775995687-620760967-3806017680
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE
Server: MERA MVTS3G v.4.3.0-22
X-mera-expires: 3660
Content-Length:   242

v=0
o=- 1323955072 1323955072 IN IP4 95.128.224.36
s=-
c=IN IP4 95.128.224.36
t=0 0
m=audio 5380 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 95.128.224.36:5380
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 95.128.224.36:5380
list_route: hop: <sip:[email protected]:5060>
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 95.128.224.36, port 5060
Transmitting (no NAT) to 95.128.224.36:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.69.212.124:5060;branch=z9hG4bK2ccfb71a;rport
Max-Forwards: 70
From: "test" <sip:[email protected]>;tag=as4bfd4deb
To: <sip:[email protected]>;tag=1208231724-3775995687-620760967-3806017680
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Content-Length: 0


---
    -- SIP/mcm-b7419fe0 answered SIP/1478-09943388
[Dec 15 16:18:46] NOTICE[4582]: chan_sip.c:9496 sip_reg_timeout:    -- Registration for '[email protected]' timed out, trying again (Attempt #1)
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/1478-09943388", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1478-09943388", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/1478-09943388", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1478-09943388", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/1478-09943388", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1478-09943388' in macro 'hangupcall'
  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/1478-09943388'
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: OPTIONS)
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 95.128.224.36, port 5060
Reliably Transmitting (no NAT) to 95.128.224.36:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.69.212.124:5060;branch=z9hG4bK5b21c0ae;rport
Max-Forwards: 70
From: "test" <sip:[email protected]>;tag=as4bfd4deb
To: <sip:[email protected]>;tag=1208231724-3775995687-620760967-3806017680
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/1478-09943388' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 889265308358, 4) exited non-zero on 'SIP/1478-09943388'
[Dec 15 16:18:48] WARNING[4567]: pbx.c:3163 ast_hint_extension: FONALITY: This thread has already held the conlock, skip locking
trixbox*CLI>
<--- SIP read from UDP://95.128.224.36:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.69.212.124:5060;branch=z9hG4bK5b21c0ae;rport=5060;received=217.69.212.124
From: "test" <sip:[email protected]>;tag=as4bfd4deb
To: <sip:[email protected]>;tag=1208231724-3775995687-620760967-3806017680
Call-ID: [email protected]
CSeq: 103 BYE
Contact: <sip:[email protected]:5060>
Server: MERA MVTS3G v.4.3.0-22
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: OPTIONS
trixbox*CLI>
Disconnected from Asterisk server