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Messages posted by: skunky
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Salut.

Am implementata urmatoarea interconectare pe voce pe mai multe sucursale ( ma voi rezuma la doar 3 puncte - HQ & 2 sucursale)

Elastix / Trixbox


SipPhone/X-Lite/Fxs <-----LAN-----> Asterisk HQ <-----IAX2 (OpenVpn-UDP) ----> Asterisk Sucursala 1 <---LAN---> SipPhone/X-Lite/Fxs.
<-----IAX2 (OpenVpn-UDP - legat tot la Asterisk HQ) ----> Asterisk Sucursala 2 <---LAN---> SipPhone/X-Lite/Fxs.


Fiecare dintre Astersik-uri are si trunk-uri SIP de la Provideri de telefonie. Acum... Apelurile prin trunchiurile sip din locatii sunt ok (dupa ce am activat pe SIP jitterbuffer adaptive pe centralele din locatii) . Traficul intre ip-urile ce ridica vpn-ul de voce este prioritizat de catre provider, iar in Lan este priorizat cu Queues pe dscp 26 /46 /port 5060



Pe centrala ce le leaga pe toate (Trixbox) nu am configurat jitter.
1. Ar trebui activat jitter pe SIP si pe centrala principala, sau doar in centralale din locatii ?
2. Ar trebui activat jitter si pe IAX ( apelul intre locatii este SIP-IAX-SIP ) ?

Multumesc anticipat.
Salut.
Am 2 bucati Linksys PAP2T (2 porturi FXS). Exista vreo sansa in a configura aceste echipamente astfel incat daca se formeaza de pe FXS1-Tel1 sa sune FXS2-Tel1 ?
Adica sa le interconectez fara a le inregistra la o centrala ip.
Am mai facut un apel din 165 (nec) , desi CID apelant imi apare 100, si se pare ca To si Contact nu sunt empty:

<------------>
-- SIP/3801-00000090 is ringing

<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100"<sip:100@192.168.1.11>;tag=2A6132463135364100165109
To: <sip:3801@192.168.1.16:5060>
Contact: <sip:100@192.168.1.11:5060>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C2C0028140000000002A@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK00532995F794FB02
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10020 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
--- (14 headers 12 lines) ---
Ignoring this INVITE request
Multumesc mult,
192.168.1.20 apartine tot nec, pe acesta adresa se face traficul de voce. Log pe nec se pare ca nu avem. Cei de la nec au spus: ", NEC-ul nu da drumul la voce pentru ca nu primeste OK la SDP". Iata un ngrep pentru un apel 3801 -> 104 (care functioneaza ok):

U 192.168.1.16:11456 -> 192.168.1.20:10028
...://.. ."........UT.....U....TWU.UTVVT..UVWTU.....UTU.UT..UTWWU....U.........UUU.....UWWT.....UUUTVQQQWWWWVVVVUUUTTU.UTUTUUTWTWUTTTWTU....TU...TU...........UU.....
....U..
#
U 192.168.1.20:10028 -> 192.168.1.16:11456
...f..U...JT.........................................................................................................................................................
.......
#
U 192.168.1.16:11456 -> 192.168.1.20:10028
...;//.. ."..UTU......................UTU...U........UUTTTTWWWVWUTWWWTUTUUU....WWU..............................UU.......UTT....U.......UWVWU..............UU.UTTTTWV
VWU....
#
U 192.168.1.20:10028 -> 192.168.1.16:11456
...g..V@..JT.........................................................................................................................................................
.......
#

Deci trafic in ambele sensuri. Pe cand invers:

U 192.168.1.16:15184 -> 192.168.1.20:10020
......qH5../WUU.....UUUTWTUTU....UWU......TTTWT.........TWTT..U...UUTTTWTU...UWWUW.U..U..TUQW.....U.TT.T..WV.WQ.UTU....UVQT...VW.TV..T..UT.WQ.Q.R.Z.].........5...**.
.**..3.
#
U 192.168.1.16:15184 -> 192.168.1.20:10020
......q.5../**..*...."(...*(..3**,..#'..T..x<;2..>..7..".........>9..l.....q.............6.HO.....h......\...z....f...S........a.......}....m..d6....^.....W...1.....
.P.`...
#
U 192.168.1.16:15184 -> 192.168.1.20:10020
......r.5../zPr....~..Y.....`l.f.....J....CJ.......|.y.p..{G.............~.........w....7.6....f....RJJ......ca.iC...C..R..f.K....v..j..?...'...d...j'f..d.....0.T.
].....f
#
U 192.168.1.16:15184 -> 192.168.1.20:10020
......s(5../57....l...2......sn..dz..@....6m......n.1..c..eTX..........**..1.**..,....*.....)4..**..9&...%...1..6..*7....*...%...!......;&.7**=.......-**7)**..**....
.***,!.
#

De la mine pleaca voce, insa nu primesc nimic. Cineva pe un forum la asterisk mi-a spus urmatoarea treaba:
"There is no ACK on what I assume to be a re-invite from the NEC.

There also seem to be protocol violations: To, Contact, etc., containing empty values."

O idee, ceva ?

Salut !
Am o MARE problema, ce trebuie rezolvata sau macar identificata.
Centrala NEC-Phillips cu licenta de trunchi SIP instalata si configurata, activat g711a.
Elastix - Asterisk 1.8.7 cu un trunchi sip creat, avand urmatoarele setari:

host=192.168.1.11
type=peer
disallow=all
allow=alaw

la incoming am lasat blank.

Are pus si "allow anonymous inbound calls".

(192.168.1.11 - NEC, 192.168.1.16 - Asterisk)

Apelez de pe un Aastra inregistrat in asterisk cu extensie 3801 catre 104 - extensie NEC. Suna, se aude, totul ok.

Apelez de pe 104 catre 3801 - 3801 suna - raspund - nu se aude nimic.
Apelez de pe 104 catre 3801 - 3801 suna - inchid 104 - 3801 continua sa sune la nesfarsit.


La un moment dat, in log apare "Ignoring this invite request - De ce oare, asta o fi cauza?"

Deja sunt in faza in care nu mai stiu ce sa-i fac, domnii de la NEC m-au ajutat pana la un punct in care mi-au precizat ca Asterisk nu figureaza pe lista de echipamente compatibile.
Totusi cred ca mie imi scapa ceva in cofigurarea Asterisk.

Pun si o bucata (cam mare) de log - apel 104 -> 3801 si raspuns pe 3801 (nu aud voce) - imi pare rau daca am postat prea multa informatie:

<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>
Contact: <sip:100@192.168.1.11:5060>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Sending to 192.168.1.11:5060 (no NAT)
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Using INVITE request as basis request - 0201C1A90C81400000000010@192.168.1.11
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found peer 'nec-silf' for '100' from 192.168.1.11:5060
[May 8 13:17:27] VERBOSE[10572] netsock2.c: == Using SIP RTP TOS bits 184
[May 8 13:17:27] VERBOSE[10572] netsock2.c: == Using SIP RTP CoS mark 5
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found RTP audio format 8
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found RTP audio format 2
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found RTP audio format 18
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found RTP audio format 9
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found audio description format PCMA for ID 8
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found audio description format G726-32 for ID 2
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found audio description format G729 for ID 18
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Found audio description format G722 for ID 9
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Capabilities: us - 0x8 (alaw), peer - audio=0x1908 (alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Peer audio RTP is at port 192.168.1.20:10026
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: Looking for 3801 in from-trunk-sip-nec-silf (domain 192.168.1.16)
[May 8 13:17:27] VERBOSE[10572] chan_sip.c: list_route: hop: <sip:100@192.168.1.11:5060>
[May 8 13:17:27] VERBOSE[10572] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Length: 0


<------------>
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [3801@from-trunk-sip-nec-silf:1] Set("SIP/nec-silf-00000004", "GROUP()=OUT_4") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [3801@from-trunk-sip-nec-silf:2] Goto("SIP/nec-silf-00000004", "from-trunk,3801,1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (from-trunk,3801,1)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [3801@from-trunk:1] Macro("SIP/nec-silf-00000004", "exten-vm,novm,3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/nec-silf-00000004", "user-callerid,") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/nec-silf-00000004", "AMPUSER=100") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/nec-silf-00000004", "0?report") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/nec-silf-00000004", "1?Set(REALCALLERIDNUM=100)") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:4] Set("SIP/nec-silf-00000004", "AMPUSER=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:5] Set("SIP/nec-silf-00000004", "AMPUSERCIDNAME=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/nec-silf-00000004", "1?report") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-user-callerid,s,10)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/nec-silf-00000004", "0?continue") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:11] Set("SIP/nec-silf-00000004", "__TTL=64") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:12] GotoIf("SIP/nec-silf-00000004", "1?continue") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-user-callerid,s,19)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:19] Set("SIP/nec-silf-00000004", "CALLERID(number)=100") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:20] Set("SIP/nec-silf-00000004", "CALLERID(name)=100") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-user-callerid:21] NoOp("SIP/nec-silf-00000004", "Using CallerID "100" <100>") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:2] Set("SIP/nec-silf-00000004", "RingGroupMethod=none") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:3] Set("SIP/nec-silf-00000004", "VMBOX=novm") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:4] Set("SIP/nec-silf-00000004", "__EXTTOCALL=3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:5] Set("SIP/nec-silf-00000004", "CFUEXT=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:6] Set("SIP/nec-silf-00000004", "CFBEXT=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:7] Set("SIP/nec-silf-00000004", "RT=""") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:8] Macro("SIP/nec-silf-00000004", "record-enable,3801,IN") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/nec-silf-00000004", "1?check") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-record-enable,s,4)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("SIP/nec-silf-00000004", "0?MacroExit()") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("SIP/nec-silf-00000004", "0?Group:OUT") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-record-enable,s,15)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-record-enable:15] GotoIf("SIP/nec-silf-00000004", "1?IN") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-record-enable,s,20)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-record-enable:20] ExecIf("SIP/nec-silf-00000004", "1?MacroExit()") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-exten-vm:9] Macro("SIP/nec-silf-00000004", "dial-one,"",tr,3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:1] Set("SIP/nec-silf-00000004", "DEXTEN=3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:2] Set("SIP/nec-silf-00000004", "DIALSTATUS_CW=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:3] GosubIf("SIP/nec-silf-00000004", "0?screen,1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:4] GosubIf("SIP/nec-silf-00000004", "0?cf,1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:5] GotoIf("SIP/nec-silf-00000004", "1?skip1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-dial-one,s,
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:8] GotoIf("SIP/nec-silf-00000004", "0?nodial") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:9] GotoIf("SIP/nec-silf-00000004", "0?continue") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:10] Set("SIP/nec-silf-00000004", "EXTHASCW=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:11] GotoIf("SIP/nec-silf-00000004", "1?next1:cwinusebusy") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-dial-one,s,12)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:12] GotoIf("SIP/nec-silf-00000004", "0?docfu:skip3") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-dial-one,s,16)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:16] GotoIf("SIP/nec-silf-00000004", "1?next2:continue") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-dial-one,s,17)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:17] GotoIf("SIP/nec-silf-00000004", "1?continue") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-dial-one,s,25)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:25] GotoIf("SIP/nec-silf-00000004", "0?nodial") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:26] GosubIf("SIP/nec-silf-00000004", "1?dstring,1:dlocal,1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:1] Set("SIP/nec-silf-00000004", "DSTRING=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:2] Set("SIP/nec-silf-00000004", "DEVICES=3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/nec-silf-00000004", "0?Return()") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/nec-silf-00000004", "0?Set(DEVICES=801)") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:5] Set("SIP/nec-silf-00000004", "LOOPCNT=1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:6] Set("SIP/nec-silf-00000004", "ITER=1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:7] Set("SIP/nec-silf-00000004", "THISDIAL=SIP/3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/nec-silf-00000004", "1?zap2dahdi,1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/nec-silf-00000004", "0?Return()") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/nec-silf-00000004", "NEWDIAL=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/nec-silf-00000004", "LOOPCNT2=1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/nec-silf-00000004", "ITER2=1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/nec-silf-00000004", "THISPART2=SIP/3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/nec-silf-00000004", "0?Set(THISPART2=DAHDI/3801)") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/nec-silf-00000004", "NEWDIAL=SIP/3801&") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/nec-silf-00000004", "ITER2=2") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/nec-silf-00000004", "0?begin2") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/nec-silf-00000004", "THISDIAL=SIP/3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/nec-silf-00000004", "") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:9] Set("SIP/nec-silf-00000004", "DSTRING=SIP/3801&") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:10] Set("SIP/nec-silf-00000004", "ITER=2") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/nec-silf-00000004", "0?begin") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:12] Set("SIP/nec-silf-00000004", "DSTRING=SIP/3801") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [dstring@macro-dial-one:13] Return("SIP/nec-silf-00000004", "") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:27] GotoIf("SIP/nec-silf-00000004", "0?nodial") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:28] GotoIf("SIP/nec-silf-00000004", "1?skiptrace") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Goto (macro-dial-one,s,30)
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:30] Set("SIP/nec-silf-00000004", "D_OPTIONS=tr") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:31] ExecIf("SIP/nec-silf-00000004", "0?SIPAddHeader(Alert-Info: )") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:32] ExecIf("SIP/nec-silf-00000004", "0?SIPAddHeader()") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:33] ExecIf("SIP/nec-silf-00000004", "0?Set(CHANNEL(musicclass)=)") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:34] GosubIf("SIP/nec-silf-00000004", "0?qwait,1") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:35] Set("SIP/nec-silf-00000004", "__CWIGNORE=") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:36] Set("SIP/nec-silf-00000004", "__KEEPCID=TRUE") in new stack
[May 8 13:17:27] VERBOSE[10685] pbx.c: -- Executing [s@macro-dial-one:37] Dial("SIP/nec-silf-00000004", "SIP/3801,"",tr") in new stack
[May 8 13:17:27] VERBOSE[10685] netsock2.c: == Using SIP RTP TOS bits 184
[May 8 13:17:27] VERBOSE[10685] netsock2.c: == Using SIP RTP CoS mark 5
[May 8 13:17:27] VERBOSE[10685] app_dial.c: -- Called SIP/3801
[May 8 13:17:27] VERBOSE[10685] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Length: 0


<------------>
[May 8 13:17:28] VERBOSE[10685] app_dial.c: -- SIP/3801-00000005 is ringing
[May 8 13:17:28] VERBOSE[10685] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Length: 0


<------------>
[May 8 13:17:28] VERBOSE[10685] app_dial.c: -- SIP/3801-00000005 is ringing
[May 8 13:17:28] VERBOSE[10572] chan_sip.c:
<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>
Contact: <sip:100@192.168.1.11:5060>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:28] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:28] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:28] VERBOSE[10572] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Length: 0


<------------>
[May 8 13:17:29] VERBOSE[10572] chan_sip.c:
<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>
Contact: <sip:100@192.168.1.11:5060>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:29] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:29] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:29] VERBOSE[10572] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Length: 0


[May 8 13:17:30] VERBOSE[10685] app_dial.c: -- SIP/3801-00000005 answered SIP/nec-silf-00000004
[May 8 13:17:30] VERBOSE[10685] chan_sip.c: Audio is at 5060
[May 8 13:17:30] VERBOSE[10685] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[May 8 13:17:30] VERBOSE[10685] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<------------>
[May 8 13:17:31] VERBOSE[10572] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:31] VERBOSE[10572] chan_sip.c:
<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>
Contact: <sip:100@192.168.1.11:5060>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:31] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:31] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:31] VERBOSE[10572] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Length: 0


<------------>
[May 8 13:17:31] VERBOSE[10572] chan_sip.c: Audio is at 5060
[May 8 13:17:31] VERBOSE[10572] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[May 8 13:17:31] VERBOSE[10572] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763839 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<------------>
[May 8 13:17:32] VERBOSE[10572] chan_sip.c: Retransmitting #2 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:34] VERBOSE[10572] chan_sip.c: Retransmitting #3 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [h@macro-dial-one:1] Macro("SIP/nec-silf-00000004", "hangupcall,") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/nec-silf-00000004", "1?endmixmoncheck") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,9)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:9] NoOp("SIP/nec-silf-00000004", "End of MIXMON check") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:10] GotoIf("SIP/nec-silf-00000004", "1?nomeetmemon") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,15)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:15] NoOp("SIP/nec-silf-00000004", "MEETME_RECORDINGFILE=") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:16] GotoIf("SIP/nec-silf-00000004", "1?noautomon") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,1
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:18] NoOp("SIP/nec-silf-00000004", "TOUCH_MONITOR_OUTPUT=") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:19] GotoIf("SIP/nec-silf-00000004", "1?noautomon2") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,25)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:25] NoOp("SIP/nec-silf-00000004", "MONITOR_FILENAME=") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:26] GotoIf("SIP/nec-silf-00000004", "1?skiprg") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,29)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:29] GotoIf("SIP/nec-silf-00000004", "1?skipblkvm") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,32)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:32] GotoIf("SIP/nec-silf-00000004", "1?theend") in new stack
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Goto (macro-hangupcall,s,34)
[May 8 13:17:34] VERBOSE[10685] pbx.c: -- Executing [s@macro-hangupcall:34] Hangup("SIP/nec-silf-00000004", "") in new stack
[May 8 13:17:34] VERBOSE[10685] app_macro.c: == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/nec-silf-00000004' in macro 'hangupcall'
[May 8 13:17:34] VERBOSE[10685] features.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/nec-silf-00000004'
[May 8 13:17:34] VERBOSE[10685] app_macro.c: == Spawn extension (macro-dial-one, s, 37) exited non-zero on 'SIP/nec-silf-00000004' in macro 'dial-one'
[May 8 13:17:34] VERBOSE[10685] app_macro.c: == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/nec-silf-00000004' in macro 'exten-vm'
[May 8 13:17:34] VERBOSE[10685] pbx.c: == Spawn extension (from-trunk, 3801, 1) exited non-zero on 'SIP/nec-silf-00000004'
[May 8 13:17:34] VERBOSE[10685] chan_sip.c: Scheduling destruction of SIP dialog '0201C1A90C81400000000010@192.168.1.11' in 32000 ms (Method: INVITE)
[May 8 13:17:35] VERBOSE[10572] chan_sip.c:
<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>
Contact: <sip:100@192.168.1.11:5060>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:35] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:35] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:35] NOTICE[10572] chan_sip.c: Unable to create/find SIP channel for this INVITE
[May 8 13:17:35] VERBOSE[10572] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[May 8 13:17:35] VERBOSE[10572] chan_sip.c: Scheduling destruction of SIP dialog '0201C1A90C81400000000010@192.168.1.11' in 32000 ms (Method: INVITE)
[May 8 13:17:38] VERBOSE[10572] chan_sip.c: Retransmitting #4 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:42] VERBOSE[10572] chan_sip.c: Retransmitting #5 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:43] VERBOSE[10572] chan_sip.c:
<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>
Contact: <sip:100@192.168.1.11:5060>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:43] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:43] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request
[May 8 13:17:46] VERBOSE[10572] chan_sip.c: Retransmitting #6 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:50] VERBOSE[10572] chan_sip.c: Retransmitting #7 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:54] VERBOSE[10572] chan_sip.c: Retransmitting #8 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:17:58] VERBOSE[10572] chan_sip.c: Retransmitting #9 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---

[May 8 13:17:59] VERBOSE[10572] chan_sip.c:
<--- SIP read from UDP:192.168.1.11:5060 --->
INVITE sip:3801@192.168.1.16 SIP/2.0
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>
Contact: <sip:100@192.168.1.11:5060>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 06.01
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD
Content-Length: 220

v=0
o=- 0 0 IN IP4 192.168.1.11
s=T059
c=IN IP4 192.168.1.20
t=0 0
m=audio 10026 RTP/AVP 8 2 18 9
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=ptime:30
<------------->
[May 8 13:17:59] VERBOSE[10572] chan_sip.c: --- (14 headers 12 lines) ---
[May 8 13:17:59] VERBOSE[10572] chan_sip.c: Ignoring this INVITE request

[May 8 13:18:02] VERBOSE[10572] chan_sip.c: Retransmitting #10 (no NAT) to 192.168.1.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK5C765A0A070A2FBD;received=192.168.1.11
From: "100"<sip:100@192.168.1.11>;tag=338C324631353641000B6B8E
To: <sip:3801@192.168.1.16:5060>;tag=as33edd479
Call-ID: 0201C1A90C81400000000010@192.168.1.11
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3801@192.168.1.16:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 1201763838 1201763838 IN IP4 192.168.1.16
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.16
t=0 0
m=audio 11408 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
[May 8 13:18:02] WARNING[10572] chan_sip.c: Retransmission timeout reached on transmission 0201C1A90C81400000000010@192.168.1.11 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

Salut,
Este oare posibil a ruta apelurile ce vin pe o linie a echipamentului fxo ( audiocodes mp118 ) direct catre telefonul Aastra 55i ?
Toate acestea fara un pbx intermediar.
 
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