nding to 192.168.0.221 : 5060 (NAT)
Using INVITE request as basis request - 4107324395520-71632534025939@192.168.0.221
Found peer '204' for '204' from 192.168.0.221:5060
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.221:10060
Looking for 019 in from-internal (domain 192.168.0.7)
list_route: hop: <sip:204@192.168.0.221:5060>

<--- Transmitting (NAT) to 192.168.0.221:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.221:5060;branch=z9hG4bK28424216501110711173;received=192.168.0.221;rport=5060
From: 204 <sip:204@192.168.0.7>;tag=347327437
To: 019 <sip:019@192.168.0.7>
Call-ID: 4107324395520-71632534025939@192.168.0.221
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:019@192.168.0.7>
Content-Length: 0


<------------>
-- Executing [019@from-internal:1] Macro("SIP/204-0000024c", "exten-vm,novm,019") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/204-0000024c", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/204-0000024c", "AMPUSER=204") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/204-0000024c", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/204-0000024c", "1?Set(REALCALLERIDNUM=204)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/204-0000024c", "AMPUSER=204") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/204-0000024c", "AMPUSERCIDNAME=Afanasiev Serg") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/204-0000024c", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/204-0000024c", "AMPUSERCID=204") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/204-0000024c", "CALLERID(all)="Afanasiev Serg" <204>") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/204-0000024c", "0?continue") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/204-0000024c", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/204-0000024c", "1?continue") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [s@macro-user-callerid:18] NoOp("SIP/204-0000024c", "Using CallerID "Afanasiev Serg" <204>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/204-0000024c", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/204-0000024c", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/204-0000024c", "EXTTOCALL=019") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/204-0000024c", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/204-0000024c", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/204-0000024c", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/204-0000024c", "record-enable,019,IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/204-0000024c", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/204-0000024c", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/204-0000024c", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/204-0000024c", "1?IN") in new stack
-- Goto (macro-record-enable,s,20)
-- Executing [s@macro-record-enable:20] ExecIf("SIP/204-0000024c", "1?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/204-0000024c", "dial,"",tr,019") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/204-0000024c", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/204-0000024c", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Afanasiev Serg' number is '204'
> dialparties.agi: USE_CONFIRMATION: 'FALSE'
> dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 019 to extension map
-- dialparties.agi: Extension 019 cf is disabled
-- dialparties.agi: Extension 019 do not disturb is disabled
> dialparties.agi: extnum 019 has: cw: 1; hascfb: 0 [] hascfu: 0 []
dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
-- dialparties.agi: dbset CALLTRACE/019 to 204
-- dialparties.agi: Filtered ARG3: 019
-- <SIP/204-0000024c>AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/204-0000024c", "SIP/019,"",tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Couldn't call 019
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [s@macro-dial:8] Set("SIP/204-0000024c", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-dial:9] GosubIf("SIP/204-0000024c", "0?CHANUNAVAIL,1") in new stack
-- Executing [s@macro-exten-vm:10] GotoIf("SIP/204-0000024c", "0?exit,return") in new stack
-- Executing [s@macro-exten-vm:11] Set("SIP/204-0000024c", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/204-0000024c", "0?docfu,1") in new stack
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/204-0000024c", "0?docfb,1") in new stack
-- Executing [s@macro-exten-vm:14] Set("SIP/204-0000024c", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:15] NoOp("SIP/204-0000024c", "Voicemail is 'novm'") in new stack
-- Executing [s@macro-exten-vm:16] GotoIf("SIP/204-0000024c", "1?s-CHANUNAVAIL,1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-exten-vm:1] NoOp("SIP/204-0000024c", "IVR_RETVM: IVR_CONTEXT: ") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:2] GotoIf("SIP/204-0000024c", "0?exit,1") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:3] PlayTones("SIP/204-0000024c", "congestion") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:4] Congestion("SIP/204-0000024c", "10") in new stack

<--- Reliably Transmitting (NAT) to 192.168.0.221:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.221:5060;branch=z9hG4bK28424216501110711173;received=192.168.0.221;rport=5060
From: 204 <sip:204@192.168.0.7>;tag=347327437
To: 019 <sip:019@192.168.0.7>;tag=as4a6e8563
Call-ID: 4107324395520-71632534025939@192.168.0.221
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 4) exited non-zero on 'SIP/204-0000024c' in macro 'exten-vm'
== Spawn extension (from-internal, 019, 1) exited non-zero on 'SIP/204-0000024c'
-- Executing [h@from-internal:1] Macro("SIP/204-0000024c", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/204-0000024c", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/204-0000024c", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/204-0000024c", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/204-0000024c", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/204-0000024c' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/204-0000024c'
asterisk*CLI>
<--- SIP read from UDP:192.168.0.221:5060 --->
ACK sip:019@192.168.0.7 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.221:5060;branch=z9hG4bK28424216501110711173;rport
From: 204 <sip:204@192.168.0.7>;tag=347327437
To: 019 <sip:019@192.168.0.7>;tag=as4a6e8563
Call-ID: 4107324395520-71632534025939@192.168.0.221
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4107324395520-71632534025939@192.168.0.221' Method: ACK
asterisk*CLI>
<--- SIP read from UDP:192.168.0.221:5060 --->
REGISTER sip:192.168.0.7:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.221:5060;branch=z9hG4bK830714657319112;rport
From: 204 <sip:204@192.168.0.7>;tag=3221723862
To: 204 <sip:204@192.168.0.7>
Call-ID: 30462775620730-22081900117257@192.168.0.221
CSeq: 20667 REGISTER
Contact: <sip:204@192.168.0.221:5060>
Max-Forwards: 70
Expires: 60
User-Agent: afanasiev
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.0.221 : 5060 (no NAT)

<--- Transmitting (NAT) to 192.168.0.221:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.221:5060;branch=z9hG4bK830714657319112;received=192.168.0.221;rport=5060
From: 204 <sip:204@192.168.0.7>;tag=3221723862
To: 204 <sip:204@192.168.0.7>;tag=as726f95a2
Call-ID: 30462775620730-22081900117257@192.168.0.221
CSeq: 20667 REGISTER
Server: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5fb259eb"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '30462775620730-22081900117257@192.168.0.221' in 32000 ms (Method: REGISTER)
asterisk*CLI>
<--- SIP read from UDP:192.168.0.221:5060 --->
REGISTER sip:192.168.0.7:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.221:5060;branch=z9hG4bK2750680042196715900;rport
From: 204 <sip:204@192.168.0.7>;tag=3221723862
To: 204 <sip:204@192.168.0.7>
Call-ID: 30462775620730-22081900117257@192.168.0.221
CSeq: 20668 REGISTER
Contact: <sip:204@192.168.0.221:5060>
Authorization: Digest username="204", realm="asterisk", nonce="5fb259eb", uri="sip:192.168.0.7:5060", response="90c3846458362d9f7155e3d600d5d205", algorithm=MD5
Max-Forwards: 70
Expires: 60
User-Agent: afanasiev
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.221 : 5060 (NAT)
Reliably Transmitting (NAT) to 192.168.0.221:5060:
OPTIONS sip:204@192.168.0.221:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.7:5060;branch=z9hG4bK0110a13f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.0.7>;tag=as78213e7b
To: <sip:204@192.168.0.221:5060>
Contact: <sip:Unknown@192.168.0.7>
Call-ID: 1c901de939ebcb126c515c94756c93d3@192.168.0.7
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.7
Date: Thu, 21 Oct 2010 06:58:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 192.168.0.221:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.221:5060;branch=z9hG4bK2750680042196715900;received=192.168.0.221;rport=5060
From: 204 <sip:204@192.168.0.7>;tag=3221723862
To: 204 <sip:204@192.168.0.7>;tag=as726f95a2
Call-ID: 30462775620730-22081900117257@192.168.0.221
CSeq: 20668 REGISTER
Server: Asterisk PBX 1.6.2.7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:204@192.168.0.221:5060>;expires=60
Date: Thu, 21 Oct 2010 06:58:49 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '30462775620730-22081900117257@192.168.0.221' in 32000 ms (Method: REGISTER)
asterisk*CLI>
<--- SIP read from UDP:192.168.0.221:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.7:5060;branch=z9hG4bK0110a13f;rport
From: Unknown <sip:Unknown@192.168.0.7>;tag=as78213e7b
To: <sip:204@192.168.0.221:5060>
Call-ID: 1c901de939ebcb126c515c94756c93d3@192.168.0.7
CSeq: 102 OPTIONS
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1c901de939ebcb126c515c94756c93d3@192.168.0.7' Method: OPTIONS
asterisk*CLI>

asterisk*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME DESC
--------------------------------------------------------------------------------
1 (1 << 0) (0x1) audio g723 (G.723.1)
2 (1 << 1) (0x2) audio gsm (GSM)
4 (1 << 2) (0x4) audio ulaw (G.711 u-law)
8 (1 << 3) (0x8) audio alaw (G.711 A-law)
16 (1 << 4) (0x10) audio g726aal2 (G.726 AAL2)
32 (1 << 5) (0x20) audio adpcm (ADPCM)
64 (1 << 6) (0x40) audio slin (16 bit Signed Linear PCM)
128 (1 << 7) (0x80) audio lpc10 (LPC10)
256 (1 << 8) (0x100) audio g729 (G.729A)
512 (1 << 9) (0x200) audio speex (SpeeX)
1024 (1 << 10) (0x400) audio ilbc (iLBC)
2048 (1 << 11) (0x800) audio g726 (G.726 RFC3551)
4096 (1 << 12) (0x1000) audio g722 (G722)
65536 (1 << 16) (0x10000) image jpeg (JPEG image)
131072 (1 << 17) (0x20000) image png (PNG image)
262144 (1 << 18) (0x40000) video h261 (H.261 Video)
524288 (1 << 19) (0x80000) video h263 (H.263 Video)
1048576 (1 << 20) (0x100000) video h263p (H.263+ Video)
2097152 (1 << 21) (0x200000) video h264 (H.264 Video)
asterisk*CLI>

asterisk*CLI> module show like codec
Module |Description Use Count
codec_g726.so |ITU G.726-32kbps G726 Transcoder 0
codec_alaw.so |A-law Coder/Decoder 0
codec_ulaw.so |mu-Law Coder/Decoder 0
codec_lpc10.so |LPC10 2.4kbps Coder/Decoder 0
codec_a_mu.so |A-law and Mulaw direct Coder/Decoder 0
codec_gsm.so |GSM Coder/Decoder 0
codec_g722.so |ITU G.722-64kbps G722 Transcoder 0
codec_adpcm.so |Adaptive Differential PCM Coder/Decoder 0
codec_dahdi.so |Generic DAHDI Transcoder Codec Translato 0
9 modules loaded

[g723]
; 6.3Kbps stream, default
;sendrate=63
; 5.3Kbps
sendrate=53

asterisk*CLI> core show translation
Translation times between formats (in microseconds) for one second of data
Source Format (Rows) Destination Format (Columns)

g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16
g723 - 1001 2 2 1001 2 1 2001 - - - 1001 1001 - - 2001
gsm 5000 - 2 2 1001 2 1 2001 - - - 1001 1001 - - 2001
ulaw 5000 1001 - 1 1001 2 1 2001 - - - 1001 1001 - - 2001
alaw 5000 1001 1 - 1001 2 1 2001 - - - 1001 1001 - - 2001
g726aal2 5999 2000 1001 1001 - 1001 1000 3000 - - - 2000 2000 - - 3000
adpcm 5999 2000 1001 1001 2000 - 1000 3000 - - - 2000 2000 - - 3000
slin 4999 1000 1 1 1000 1 - 2000 - - - 1000 1000 - - 2000
lpc10 5999 2000 1001 1001 2000 1001 1000 - - - - 2000 2000 - - 3000
g729 - - - - - - - - - - - - - - - -
speex - - - - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - - - - -
g726 5999 2000 1001 1001 2000 1001 1000 3000 - - - - 2000 - - 3000
g722 5999 2000 1001 1001 2000 1001 1000 3000 - - - 2000 - - - 1000
siren7 - - - - - - - - - - - - - - - -
siren14 - - - - - - - - - - - - - - - -
slin16 6998 2999 2000 2000 2999 2000 1999 3999 - - - 2999 999 - - -

disallow = all
allow = g723.1

disallow = all
allow = g723.1