extension sip no se le escucha

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extension sip no se le escucha

hgmnetwork

Hola, tengo un problema raro, hoy a un cliente le han cambiado de internet por el operador orange (españa)
este operador utiliza el puerto 5060 para su telefonia, el problema es que el cliente se conecta por voip a nosotros bien, y puede recibir y emitir llamadas bien, se le escucha bien, pero a el no se le escucha.

Y no se que pueda ser, he abierto puertos del 10 mil a 20 mil nada.

pego el log por si ayuda a que detecten algo que no me diera cuenta veo que va al puerto 5061 y tenemos el 5060 no se si se puede poner que escuche en ambos o donde cambiarlo por si sea eso

<--- SIP read from UDP:92.185.176.157:5061 --->
INVITE sip:1001@voz.xxxx.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 92.185.176.157:5061;branch=z9hG4bK-d8754z-b9291fa24e5e483b-1---d8754z-
Max-Forwards: 70
Contact: <sip:5002@92.185.176.157:5061;transport=UDP>
To: <sip:1001@voz.xxxxxx.com;transport=UDP>
From: <sip:5002@voz.xxxxx.com;transport=UDP>;tag=9310777f
Call-ID: NTFkOTVjZWRiM2M5OTgwMGNhZjk3YzM4NDg1MmJkYjk.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.43 r24984
Allow-Events: presence, kpml
Content-Length: 331

<--- Reliably Transmitting (NAT) to 92.185.176.157:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 92.185.176.157:5061;branch=z9hG4bK-d8754z-b9291fa24e5e483b-1---d8754z-;received=92.185.176.157;rport=5061
From: <sip:5002@voz.xxxxx.com;transport=UDP>;tag=9310777f
To: <sip:1001@voz.xxxxxx.com;transport=UDP>;tag=as344c6dd2
Call-ID: NTFkOTVjZWRiM2M5OTgwMGNhZjk3YzM4NDg1MmJkYjk.
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7d9596cd"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'NTFkOTVjZWRiM2M5OTgwMGNhZjk3YzM4NDg1MmJkYjk.' in 11840 ms (Method: INVITE)

<--- SIP read from UDP:92.185.176.157:5061 --->
ACK sip:1001@voz.xxxxxx.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 92.185.176.157:5061;branch=z9hG4bK-d8754z-b9291fa24e5e483b-1---d8754z-
Max-Forwards: 70
To: <sip:1001@voz.xxxxx.com;transport=UDP>;tag=as344c6dd2
From: <sip:5002@voz.xxxxxxx.com;transport=UDP>;tag=9310777f
Call-ID: NTFkOTVjZWRiM2M5OTgwMGNhZjk3YzM4NDg1MmJkYjk.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:92.185.176.157:5061 --->
INVITE sip:1001@voz.xxxxxxx.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 92.185.176.157:5061;branch=z9hG4bK-d8754z-aae0cf0f09b54ef8-1---d8754z-
Max-Forwards: 70
Contact: <sip:5002@92.185.176.157:5061;transport=UDP>
To: <sip:1001@voz.xxxxxx.com;transport=UDP>
From: <sip:5002@voz.xxxxxx.com;transport=UDP>;tag=9310777f
Call-ID: NTFkOTVjZWRiM2M5OTgwMGNhZjk3YzM4NDg1MmJkYjk.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.43 r24984
Authorization: Digest username="5002",realm="asterisk",nonce="7d9596cd",uri="sip:1001@voz.xxxxx.com;transport=UDP",response="8e4321b5de8c71f8611c727018044bc0",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 331

v=0
o=Zoiper_user 0 0 IN IP4 92.185.176.157
s=Zoiper_session
c=IN IP4 92.185.176.157
t=0 0
m=audio 8016 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 15 lines) ---
Sending to 92.185.176.157:5061 (NAT)
Using INVITE request as basis request - NTFkOTVjZWRiM2M5OTgwMGNhZjk3YzM4NDg1MmJkYjk.
Found peer '5002' for '5002' from 92.185.176.157:5061
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 92.185.176.157:8016
Looking for 1001 in from-internal (domain voz.xxxxx.com)
list_route: hop: <sip:5002@92.185.176.157:5061;transport=UDP>

<--- Transmitting (NAT) to 92.185.176.157:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 92.185.176.157:5061;branch=z9hG4bK-d8754z-aae0cf0f09b54ef8-1---d8754z-;received=92.185.176.157;rport=5061
From: <sip:5002@voz.xxxxx.com;transport=UDP>;tag=9310777f
To: <sip:1001@voz.xxxxx.com;transport=UDP>
Call-ID: NTFkOTVjZWRiM2M5OTgwMGNhZjk3YzM4NDg1MmJkYjk.
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1001@213.231.90.192:5060>
Content-Length: 0

<------------>
-- Executing [1001@from-internal:1] GotoIf("SIP/5002-000079ee", "0?ext-local,1001,1") in new stack
-- Executing [1001@from-internal:2] Macro("SIP/5002-000079ee", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/5002-000079ee", "TOUCH_MONITOR=1509720409.32062") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/5002-000079ee", "AMPUSER=5002") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/5002-000079ee", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/5002-000079ee", "1?Set(REALCALLERIDNUM=5002)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/5002-000079ee", "AMPUSER=5002") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/5002-000079ee", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/5002-000079ee", "AMPUSERCIDNAME=Agente 5002") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/5002-000079ee", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/5002-000079ee", "AMPUSERCID=5002") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/5002-000079ee", "DIAL_OPTIONS=trTwW") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/5002-000079ee", "CALLERID(all)="Agente 5002" <5002>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/5002-000079ee", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("SIP/5002-000079ee", "0?Set(GROUP(concurrency_limit)=5002)") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("SIP/5002-000079ee", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/5002-000079ee", "0?continue") in new stack
-- Executing [s@macro-user-callerid:16] Set("SIP/5002-000079ee", "TTL=64") in new stack
-- Executing [s@macro-user-callerid:17] GotoIf("SIP/5002-000079ee", "1?continue") in new stack
-- Goto (macro-user-callerid,s,28)
-- Executing [s@macro-user-callerid:28] Set("SIP/5002-000079ee", "CALLERID(number)=5002") in new stack
-- Executing [s@macro-user-callerid:29] Set("SIP/5002-000079ee", "CALLERID(name)=Agente 5002") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/5002-000079ee", "CDR(cnum)=5002") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/5002-000079ee", "CDR(cnam)=Agente 5002") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/5002-000079ee", "CHANNEL(language)=es") in new stack
-- Executing [1001@from-internal:3] Set("SIP/5002-000079ee", "DIAL_OPTIONS=trTwWI") in new stack
-- Executing [1001@from-internal:4] Set("SIP/5002-000079ee", "CONNECTEDLINE(num)=1001") in new stack
-- Executing [1001@from-internal:5] Gosub("SIP/5002-000079ee", "sub-presencestate-display,s,1(1001)") in new stack
[2017-11-03 15:46:49] WARNING[4690][C-000249ec]: func_presencestate.c:132 presence_read: PRESENCE_STATE unknown
-- Executing [s@sub-presencestate-display:1] Goto("SIP/5002-000079ee", "state-,1") in new stack
-- Goto (sub-presencestate-display,state-,1)
-- Executing [state-@sub-presencestate-display:1] Set("SIP/5002-000079ee", "PRESENCESTATE_DISPLAY=") in new stack
-- Executing [state-@sub-presencestate-display:2] Return("SIP/5002-000079ee", "") in new stack
-- Executing [1001@from-internal:6] Set("SIP/5002-000079ee", "CONNECTEDLINE(name,i)=Moises") in new stack
-- Executing [1001@from-internal:7] Set("SIP/5002-000079ee", "FM_DIALSTATUS=NOT_INUSE") in new stack
-- Executing [1001@from-internal:8] Set("SIP/5002-000079ee", "EXTTOCALL=1001") in new stack
-- Executing [1001@from-internal:9] Set("SIP/5002-000079ee", "PICKUPMARK=1001") in new stack
-- Executing [1001@from-internal:10] Macro("SIP/5002-000079ee", "blkvm-setifempty,") in new stack
-- Executing [s@macro-blkvm-setifempty:1] GotoIf("SIP/5002-000079ee", "1?init") in new stack
-- Goto (macro-blkvm-setifempty,s,4)
-- Executing [s@macro-blkvm-setifempty:4] Set("SIP/5002-000079ee", "BLKVM_CHANNEL=SIP/5002-000079ee") in new stack
-- Executing [s@macro-blkvm-setifempty:5] Set("SIP/5002-000079ee", "SHARED(BLKVM,SIP/5002-000079ee)=TRUE") in new stack
-- Executing [s@macro-blkvm-setifempty:6] Set("SIP/5002-000079ee", "GOSUB_RETVAL=TRUE") in new stack
-- Executing [s@macro-blkvm-setifempty:7] MacroExit("SIP/5002-000079ee", "") in new stack
-- Executing [1001@from-internal:11] GotoIf("SIP/5002-000079ee", "1?skipov") in new stack
-- Goto (from-internal,1001,14)
-- Executing [1001@from-internal:14] Set("SIP/5002-000079ee", "RRNODEST=") in new stack
-- Executing [1001@from-internal:15] Set("SIP/5002-000079ee", "NODEST=1001") in new stack
-- Executing [1001@from-internal:16] GosubIf("SIP/5002-000079ee", "0?sub-fmsetcid,s,1()") in new stack
-- Executing [1001@from-internal:17] Set("SIP/5002-000079ee", "RecordMethod=Group") in new stack
-- Executing [1001@from-internal:18] Gosub("SIP/5002-000079ee", "sub-record-check,s,1(exten,1001,)") in new stack
-- Executing [s@sub-record-check:1] Set("SIP/5002-000079ee", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:2] GotoIf("SIP/5002-000079ee", "1?check") in new stack
-- Goto (sub-record-check,s,7)
-- Executing [s@sub-record-check:7] Set("SIP/5002-000079ee", "MON_FMT=WAV") in new stack
-- Executing [s@sub-record-check:8] GotoIf("SIP/5002-000079ee", "1?next") in new stack
-- Goto (sub-record-check,s,11)
-- Executing [s@sub-record-check:11] ExecIf("SIP/5002-000079ee", "0?Return()") in new stack
-- Executing [s@sub-record-check:12] ExecIf("SIP/5002-000079ee", "0?Set(REC_POLICY_MODE=)") in new stack
-- Executing [s@sub-record-check:13] GotoIf("SIP/5002-000079ee", "0?exten,1") in new stack
-- Executing [s@sub-record-check:14] Set("SIP/5002-000079ee", "REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:15] Set("SIP/5002-000079ee", "NOW=1509720410") in new stack
-- Executing [s@sub-record-check:16] Set("SIP/5002-000079ee", "DAY=03") in new stack
-- Executing [s@sub-record-check:17] Set("SIP/5002-000079ee", "MONTH=11") in new stack
-- Executing [s@sub-record-check:18] Set("SIP/5002-000079ee", "YEAR=2017") in new stack
-- Executing [s@sub-record-check:19] Set("SIP/5002-000079ee", "TIMESTR=20171103-154650") in new stack
-- Executing [s@sub-record-check:20] Set("SIP/5002-000079ee", "FROMEXTEN=5002") in new stack
-- Executing [s@sub-record-check:21] Set("SIP/5002-000079ee", "CALLFILENAME=exten-1001-5002-20171103-154650-1509720409.32062") in new stack
-- Executing [s@sub-record-check:22] Goto("SIP/5002-000079ee", "exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [exten@sub-record-check:1] GotoIf("SIP/5002-000079ee", "0?callee") in new stack
-- Executing [exten@sub-record-check:2] Set("SIP/5002-000079ee", "REC_POLICY_MODE=always") in new stack
-- Executing [exten@sub-record-check:3] GotoIf("SIP/5002-000079ee", "0?caller") in new stack
-- Executing [exten@sub-record-check:4] GotoIf("SIP/5002-000079ee", "0?callee") in new stack
-- Executing [exten@sub-record-check:5] ExecIf("SIP/5002-000079ee", "2?Set(CALLER_PRI=10):Set(CALLER_PRI=0)") in new stack
-- Executing [exten@sub-record-check:6] ExecIf("SIP/5002-000079ee", "2?Set(CALLEE_PRI=10):Set(CALLEE_PRI=0)") in new stack
-- Executing [exten@sub-record-check:7] GotoIf("SIP/5002-000079ee", "1?caller:callee") in new stack
-- Goto (sub-record-check,exten,10)
-- Executing [exten@sub-record-check:10] Set("SIP/5002-000079ee", "REC_POLICY_MODE=always") in new stack
-- Executing [exten@sub-record-check:11] GosubIf("SIP/5002-000079ee", "1?record,1(exten,1001,5002)") in new stack
-- Executing [record@sub-record-check:1] Set("SIP/5002-000079ee", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [record@sub-record-check:2] MixMonitor("SIP/5002-000079ee", "2017/11/03/exten-1001-5002-20171103-154650-1509720409.32062.WAV,,") in new stack
-- Executing [record@sub-record-check:3] Set("SIP/5002-000079ee", "REC_STATUS=RECORDING") in new stack
-- Executing [record@sub-record-check:4] Set("SIP/5002-000079ee", "CDR(recordingfile)=exten-1001-5002-20171103-154650-1509720409.32062.WAV") in new stack
-- Executing [record@sub-record-check:5] Return("SIP/5002-000079ee", "") in new stack
-- Executing [exten@sub-record-check:12] Return("SIP/5002-000079ee", "") in new stack
-- Executing [1001@from-internal:19] Set("SIP/5002-000079ee", "RingGroupMethod=ringallv2") in new stack
-- Executing [1001@from-internal:20] Set("SIP/5002-000079ee", "_FMGRP=1001") in new stack
-- Executing [1001@from-internal:21] GotoIf("SIP/5002-000079ee", "0?doconfirm") in new stack
-- Executing [1001@from-internal:22] Macro("SIP/5002-000079ee", "dial,20,m(default),1001") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/5002-000079ee", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/5002-000079ee", "dialparties.agi") in new stack
== Begin MixMonitor Recording SIP/5002-000079ee
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'Agente 5002' number is '5002'
dialparties.agi: Methodology of ring is 'ringallv2'
-- dialparties.agi: Added extension 1001 to extension map
> dialparties.agi: got fmgrp_prering: 2, fmgrp_grptime: 20
> dialparties.agi: fmgrp_totalprering: 22
> dialparties.agi: found extension in pre-ring and array
> dialparties.agi: ringallv2 ring times: REALPRERING: 22, PRERING: 2
-- dialparties.agi: Extension 1001 cf is disabled
-- dialparties.agi: Extension 1001 do not disturb is disabled
> dialparties.agi: extnum 1001 has: cw: 1; hascfb: 0 [] hascfu: 0 []
-- dialparties.agi: dbset CALLTRACE/1001 to 5002
-- dialparties.agi: Filtered ARG3: 1001
> dialparties.agi: NODEST: 1001 adding M(auto-blkvm) to dialopts: m(default)M(auto-blkvm)
> dialparties.agi: NODEST: 1001 blkvm enabled macro already in dialopts: m(default)M(auto-blkvm)
-- <SIP/5002-000079ee>AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/5002-000079ee", "SIP/1001,22,m(default)M(auto-blkvm)") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/1001
-- Started music on hold, class 'default', on SIP/5002-000079ee
[2017-11-03 15:46:50] WARNING[4690][C-000249ec]: translate.c:343 framein: no samples for alawtolin
Audio is at 12442
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 92.185.176.157:5061 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 92.185.176.157:5061;branch=z9hG4bK-d8754z-aae0cf0f09b54ef8-1---d8754z-;received=92.185.176.157;rport=5061
From: <sip:5002@voz.xxxxx.com;transport=UDP>;tag=9310777f
To: <sip:1001@voz.xxxxxxxx.com;transport=UDP>;tag=as5c010a60
Call-ID: NTFkOTVjZWRiM2M5OTgwMGNhZjk3YzM4NDg1MmJkYjk.
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1001@213.231.90.192:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 240

v=0
o=root 1342093195 1342093195 IN IP4 213.231.90.192
s=Asterisk PBX 11.17.1
c=IN IP4 213.231.90.192
t=0 0
m=audio 12442 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
-- SIP/1001-000079ef is ringing
> 0xb0e58c20 -- Probation passed - setting RTP source address to 92.185.176.157:8016
-- SIP/1001-000079ef answered SIP/5002-000079ee
-- Executing [s@macro-auto-blkvm:1] Set("SIP/1001-000079ef", "__MACRO_RESULT=") in new stack
-- Executing [s@macro-auto-blkvm:2] Set("SIP/1001-000079ef", "CFIGNORE=") in new stack
-- Executing [s@macro-auto-blkvm:3] Set("SIP/1001-000079ef", "MASTER_CHANNEL(CFIGNORE)=") in new stack
-- Executing [s@macro-auto-blkvm:4] Set("SIP/1001-000079ef", "FORWARD_CONTEXT=from-internal") in new stack
-- Executing [s@macro-auto-blkvm:5] Set("SIP/1001-000079ef", "MASTER_CHANNEL(FORWARD_CONTEXT)=from-internal") in new stack
-- Executing [s@macro-auto-blkvm:6] Macro("SIP/1001-000079ef", "blkvm-clr,") in new stack
-- Executing [s@macro-blkvm-clr:1] Set("SIP/1001-000079ef", "SHARED(BLKVM,SIP/5002-000079ee)=") in new stack
-- Executing [s@macro-blkvm-clr:2] Set("SIP/1001-000079ef", "GOSUB_RETVAL=") in new stack
-- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/1001-000079ef", "") in new stack
-- Executing [s@macro-auto-blkvm:7] ExecIf("SIP/1001-000079ef", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=1001)") in new stack
-- Executing [s@macro-auto-blkvm:8] ExecIf("SIP/1001-000079ef", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Moises)") in new stack
-- Stopped music on hold on SIP/5002-000079ee
Audio is at 12442
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 92.185.176.157:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.185.176.157:5061;branch=z9hG4bK-d8754z-aae0cf0f09b54ef8-1---d8754z-;received=92.185.176.157;rport=5061
From: <sip:5002@voz.xxxxxx.com;transport=UDP>;tag=9310777f
To: <sip:1001@voz.xxxxxxx.com;transport=UDP>;tag=as5c010a60
Call-ID: NTFkOTVjZWRiM2M5OTgwMGNhZjk3YzM4NDg1MmJkYjk.
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1001@213.231.90.192:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 240

<------------>
> 0xb4c00018 -- Probation passed - setting RTP source address to 213.231.90.192:62076
> 0xb4c00018 -- Probation passed - setting RTP source address to 213.231.90.192:62076

<--- SIP read from UDP:92.185.176.157:5061 --->
ACK sip:1001@213.231.90.192:5060 SIP/2.0
Via: SIP/2.0/UDP 92.185.176.157:5061;branch=z9hG4bK-d8754z-b9948572e5ecb6ad-1---d8754z-
Max-Forwards: 70
Contact: <sip:5002@92.185.176.157:5061;transport=UDP>
To: <sip:1001@voz.hgmnetwork.com;transport=UDP>;tag=as5c010a60
From: <sip:5002@voz.hgmnetwork.com;transport=UDP>;tag=9310777f
Call-ID: NTFkOTVjZWRiM2M5OTgwMGNhZjk3YzM4NDg1MmJkYjk.
CSeq: 2 ACK
User-Agent: Zoiper for Windows 2.43 r24984
Authorization: Digest username="5002",realm="asterisk",nonce="7d9596cd",uri="sip:1001@voz.hgmnetwork.com;transport=UDP",response="8e4321b5de8c71f8611c727018044bc0",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:92.185.176.157:5061 --->

[2017-11-03 15:47:21] WARNING[3147]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 4b954cafc9b63d56a353bc37572739b5 for seqno 1 (Non-critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[2017-11-03 15:47:21] WARNING[3147]: chan_sip.c:4086 retrans_pkt: Timeout on 4b954cafc9b63d56a353bc37572739b5 on non-critical invite transaction.

<--- SIP read from UDP:92.185.176.157:5061 --->

<------------->
Reliably Transmitting (NAT) to 92.185.176.157:5061:
OPTIONS sip:5002@92.185.176.157:5061;rinstance=11b9d0019b2053c5;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 213.231.90.192:5060;branch=z9hG4bK6efece2d;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@213.231.90.192>;tag=as0c64e461
To: <sip:5002@92.185.176.157:5061;rinstance=11b9d0019b2053c5;transport=UDP>
Contact: <sip:Unknown@213.231.90.192:5060>
Call-ID: 6c1f9cd324e011e85ccf4b0765169339@213.231.90.192:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.17.1)
Date: Fri, 03 Nov 2017 14:47:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<--- SIP read from UDP:92.185.176.157:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.231.90.192:5060;branch=z9hG4bK6efece2d;rport=5060
Contact: <sip:192.168.1.138:5060>
To: <sip:5002@92.185.176.157:5061>
To: <sip:5002@92.185.176.157:5061;rinstance=11b9d0019b2053c5;transport=UDP>;tag=cb49cf44
From: "Unknown"<sip:Unknown@213.231.90.192>;tag=as0c64e461
Call-ID: 6c1f9cd324e011e85ccf4b0765169339@213.231.90.192:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.43 r24984
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Really destroying SIP dialog '6c1f9cd324e011e85ccf4b0765169339@213.231.90.192:5060' Method: OPTIONS
[2017-11-03 15:47:40] NOTICE[3147]: chan_sip.c:29196 check_rtp_timeout: Disconnecting call 'SIP/5002-000079ee' for lack of RTP activity in 31 seconds
-- Executing [h@macro-dial:1] Macro("SIP/5002-000079ee", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/5002-000079ee", "0?endmixmoncheck") in new stack
-- Executing [s@macro-hangupcall:2] Set("SIP/5002-000079ee", "MIXMON_CALLFILENAME=/var/spool/asterisk/monitor/exten-1001-5002-20171103-154650-1509720409.32062.WAV") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/5002-000079ee", "1?defaultmixmondir") in new stack
-- Goto (macro-hangupcall,s,5)
-- Executing [s@macro-hangupcall:5] System("SIP/5002-000079ee", "test -e /var/spool/asterisk/monitor/exten-1001-5002-20171103-154650-1509720409.32062.WAV") in new stack
-- Executing [s@macro-hangupcall:6] NoOp("SIP/5002-000079ee", "SYSTEMSTATUS = APPERROR") in new stack
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/5002-000079ee", "0?endmixmoncheck") in new stack
-- Executing [s@macro-hangupcall:8] Set("SIP/5002-000079ee", "CDR(recordingfile)=") in new stack
-- Executing [s@macro-hangupcall:9] NoOp("SIP/5002-000079ee", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/5002-000079ee", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,28)
-- Executing [s@macro-hangupcall:28] NoOp("SIP/5002-000079ee", "End of MEETME check") in new stack
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/5002-000079ee", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] NoOp("SIP/5002-000079ee", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:35] GotoIf("SIP/5002-000079ee", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,41)
-- Executing [s@macro-hangupcall:41] NoOp("SIP/5002-000079ee", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:42] GotoIf("SIP/5002-000079ee", "0?noautomon3") in new stack
-- Executing [s@macro-hangupcall:43] System("SIP/5002-000079ee", "test -e /var/spool/asterisk/monitor/2017/11/03/exten-1001-5002-20171103-154650-1509720409.32062.WAV*") in new stack
-- Executing [s@macro-hangupcall:44] NoOp("SIP/5002-000079ee", "SYSTEMSTATUS = SUCCESS") in new stack
-- Executing [s@macro-hangupcall:45] GotoIf("SIP/5002-000079ee", "0?errornoautomon2") in new stack
-- Executing [s@macro-hangupcall:46] Set("SIP/5002-000079ee", "CDR(recordingfile)=/var/spool/asterisk/monitor/2017/11/03/exten-1001-5002-20171103-154650-1509720409.32062.WAV") in new stack
-- Executing [s@macro-hangupcall:47] NoOp("SIP/5002-000079ee", "End of MIXMONITOR_FILENAME check") in new stack
-- Executing [s@macro-hangupcall:48] NoOp("SIP/5002-000079ee", "MIXMONITOR_FILENAME=/var/spool/asterisk/monitor/2017/11/03/exten-1001-5002-20171103-154650-1509720409.32062.WAV") in new stack
-- Executing [s@macro-hangupcall:49] GotoIf("SIP/5002-000079ee", "1?noautomon4") in new stack
-- Goto (macro-hangupcall,s,51)
-- Executing [s@macro-hangupcall:51] NoOp("SIP/5002-000079ee", "ONETOUCH_RECFILE=") in new stack
-- Executing [s@macro-hangupcall:52] GotoIf("SIP/5002-000079ee", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,55)
-- Executing [s@macro-hangupcall:55] GotoIf("SIP/5002-000079ee", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,58)
-- Executing [s@macro-hangupcall:58] GotoIf("SIP/5002-000079ee", "1?theend") in new stack
-- Goto (macro-hangupcall,s,60)
-- Executing [s@macro-hangupcall:60] AGI("SIP/5002-000079ee", "hangup.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
-- <SIP/5002-000079ee>AGI Script hangup.agi completed, returning 0
-- Executing [s@macro-hangupcall:61] Hangup("SIP/5002-000079ee", "") in new stack
== Spawn extension (macro-hangupcall, s, 61) exited non-zero on 'SIP/5002-000079ee' in macro 'hangupcall'
== Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/5002-000079ee'
== Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/5002-000079ee' in macro 'dial'
== Spawn extension (from-internal, 1001, 22) exited non-zero on 'SIP/5002-000079ee'
Scheduling destruction of SIP dialog 'NTFkOTVjZWRiM2M5OTgwMGNhZjk3YzM4NDg1MmJkYjk.' in 11840 ms (Method: ACK)
set_destination: Parsing <sip:5002@92.185.176.157:5061;transport=UDP> for address/port to send to
set_destination: set destination to 92.185.176.157:5061
Reliably Transmitting (NAT) to 92.185.176.157:5061:
BYE sip:5002@92.185.176.157:5061;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 213.231.90.192:5060;branch=z9hG4bK37587b85;rport

basicamente he visto esto

que nose porque da un 401 si sera porque le tengo que añadir el puerto 5061 en algun sitio al servidor

posteriormente la llamada se corta sola y he detectado esto
[2017-11-03 15:47:21] WARNING[3147]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 4b954cafc9b63d56a353bc37572739b5 for seqno 1 (Non-critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[2017-11-03 15:47:21] WARNING[3147]: chan_sip.c:4086 retrans_pkt: Timeout on 4b954cafc9b63d56a353bc37572739b5 on non-critical invite transaction.

igual es configurar algo que no me doy cuenta o no doy con que es, agradezco cualquier ayuda

dariohimo

y externip
exterhost
localnet

hgmnetwork

dariohimo gracias por responder.

creo que lo tengo bien, ya que solo ocurre con esa extensión y creo que el motivo es por el operador que es orange españa y utiliza el puerto 5060 y lo bloquea para el ( usa su telefono por voip)

no obstante esto es lo que tengo
nat=yes
externhost=ip del servidor externa
localnet=192.168.0.0/255.255.255.0
qualify=yes
externrefresh=20
externip=ip del servidor externa
localhost=ip interna

supa

That is almost always a NAT issue. Sometimes codec.

hgmnetwork

El códec no es ya que uso alas uñas y GSM funciona bien eso es creo por el proveedor de internet que bloquea los puertos

navaismo

Hola, en el log que muestras esta es la IP y puerto a la que va el Audio: 92.185.176.157:8016. En tu firewall esta permitida esa. Cuando hagas otra llamada ejecuta RTP set debug sobre ese peer y verifica las IPs involucradas.

El puerto 5060 o 5061 solo son para SIP si no tuvieras acceso o lo estuviera bloqueando Orange la llamada no se establecería. De modo que el problema está con el RTP no con SIP. Otra cosa que puedes verificar es el Softphone de acuerdo a tu log estas usando Zoiper. Algunas veces el STUN de zoiper hace que el audio se comporte de esta manera.

hgmnetwork

navaismo he probado con lo que me comentas y he abierto el puerto 8016 pero lo raro es que si lo hago cambia el puerto de forma automatica.

luego cambie y abri del puerto 8010 al puerto 8050 para tener un rango y evitar en udp

y cuando llame cambia y usa el 8052 luego abri del 8010 al 8060 por ampliar y cambio al 8062

-- Executing [state-@sub-presencestate-display:2] Return("SIP/1001-00007ac6", "") in new stack
-- Executing [s@macro-dial-one:40] Set("SIP/1001-00007ac6", "CONNECTEDLINE(name,i)=Agente 5002") in new stack
-- Executing [s@macro-dial-one:41] Set("SIP/1001-00007ac6", "CONNECTEDLINE(num)=5002") in new stack
-- Executing [s@macro-dial-one:42] Set("SIP/1001-00007ac6", "D_OPTIONS=trTwWI") in new stack
-- Executing [s@macro-dial-one:43] Dial("SIP/1001-00007ac6", "SIP/5002,,trTwWI") in new stack

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/5002
[2017-11-06 13:06:01] WARNING[27345][C-0002556d]: translate.c:343 framein: no samples for gsmtolin
-- Connected line update to SIP/1001-00007ac6 prevented.
> 0xb2eece00 -- Probation passed - setting RTP source address to 213.231.90.192:62619
> 0xb2eece00 -- Probation passed - setting RTP source address to 213.231.90.192:62619
-- SIP/5002-00007ac7 is ringing
[2017-11-06 13:06:01] WARNING[27345][C-0002556d]: translate.c:343 framein: no samples for gsmtolin
-- SIP/5002-00007ac7 is ringing
> 0xb6938a0 -- Probation passed - setting RTP source address to 92.185.176.157:8000
-- Connected line update to SIP/1001-00007ac6 prevented.
-- SIP/5002-00007ac7 answered SIP/1001-00007ac6
> 0xb6938a0 -- Probation passed - setting RTP source address to 92.185.176.157:8062
-- Executing [h@macro-dial-one:1] Macro("SIP/1001-00007ac6", "hangupcall,") in new stack

veo esto pero no se interpretarlo bien ya que parece que lo intenta en el puerto 8000 pero luego salta justo al que no tengo abierto y ponga lo que ponga siempre lo aumenta

   > 0xb6938a0 -- Probation passed - setting RTP source address to 92.185.176.157:8000
-- Connected line update to SIP/1001-00007ac6 prevented.
-- SIP/5002-00007ac7 answered SIP/1001-00007ac6
   > 0xb6938a0 -- Probation passed - setting RTP source address to 92.185.176.157:8062

yo tengo abierto ahora mismo del 8010 al 8060

me tiene loco el problema este es la unica extension de mas de 200 que no funciona todos los demas funcionan bien es mas esa misma extension la conecto por otro medio y funciona

he probado a abrir tambien el 8000 osea del 8000 al 8060 en udp y me pasa esto

-- Executing [s@macro-dial-one:43] Dial("SIP/1001-00007ad9", "SIP/5002,,trTwWI") in new stack

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/5002
[2017-11-06 13:13:03] WARNING[27385][C-0002557e]: translate.c:343 framein: no samples for gsmtolin
-- Connected line update to SIP/1001-00007ad9 prevented.
> 0xb6938a0 -- Probation passed - setting RTP source address to 213.231.90.192:62980
> 0xb6938a0 -- Probation passed - setting RTP source address to 213.231.90.192:62980
-- SIP/5002-00007ada is ringing
[2017-11-06 13:13:03] WARNING[27385][C-0002557e]: translate.c:343 framein: no samples for gsmtolin
-- SIP/5002-00007ada is ringing
> 0xa6526e0 -- Probation passed - setting RTP source address to 92.185.176.157:8000
-- Connected line update to SIP/1001-00007ad9 prevented.
-- SIP/5002-00007ada answered SIP/1001-00007ad9
> 0xa6526e0 -- Probation passed - setting RTP source address to 92.185.176.157:8000
> 0xa6526e0 -- Probation passed - setting RTP source address to 92.185.176.157:8000
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

es decir sale bien en el 8000 pero sigue sin escucharse, sin embargo esa extension oye bien.

probe tambien con el debig del rtp y siempre sale lo mismo

Sent RTP packet to 92.185.176.157:8000 (type 00, seq 006824, ts 648176392, len 000160)
Got RTP packet from 213.231.90.192:61273 (type 03, seq 035690, ts 648176554, len 000033)
Sent RTP packet to 92.185.176.157:8000 (type 00, seq 006825, ts 648176552, len 000160)
Got RTP packet from 213.231.90.192:61273 (type 03, seq 035691, ts 648176714, len 000033)
Sent RTP packet to 92.185.176.157:8000 (type 00, seq 006826, ts 648176712, len 000160)
Got RTP packet from 213.231.90.192:61273 (type 03, seq 035692, ts 648176874, len 000033)
Sent RTP packet to 92.185.176.157:8000 (type 00, seq 006827, ts 648176872, len 000160)
Got RTP packet from 213.231.90.192:61273 (type 03, seq 035693, ts 648177034, len 000033)
Sent RTP packet to 92.185.176.157:8000 (type 00, seq 006828, ts 648177032, len 000160)
Got RTP packet from 213.231.90.192:61273 (type 03, seq 035694, ts 648177194, len 000033)
Sent RTP packet to 92.185.176.157:8000 (type 00, seq 006829, ts 648177192, len 000160)
Got RTP packet from 213.231.90.192:61273 (type 03, seq 035695, ts 648177354, len 000033)
Sent RTP packet to 92.185.176.157:8000 (type 00, seq 006830, ts 648177352, len 000160)
Got RTP packet from 213.231.90.192:61273 (type 03, seq 035696, ts 648177514, len 000033)
Sent RTP packet to 92.185.176.157:8000 (type 00, seq 006831, ts 648177512, len 000160)
Got RTP packet from 213.231.90.192:61273 (type 03, seq 035697, ts 648177674, len 000033)
Sent RTP packet to 92.185.176.157:8000 (type 00, seq 006832, ts 648177672, len 000160)

RiveraPer

Pues me gustara saber que router tienes, por que yo ya he desistido de trabajar con Orange.
A mi me pasa lo mismo pero solo en las llamadas entrantes a una centralita.
Ademas, soy incapaz de configurar el router ya que Orange no me da el usuario y password de configuración avanzada y cuando hablo con ellos, no saben de que les estoy hablando.

dariohimo

Pero puede tener un puerto para esa extensión o cambiarlos para todos. los puertos estan con el standard pero se puede cambiar.
El firewal tiene activado alg.

dariohimo

nat=yes
externhost=ip del servidor externa o solo deja uno externhost o externip
localnet=192.168.0.0/255.255.255.0
qualify=yes
externrefresh=20
externip=ip del servidor externa Se utiliza con solo la ip.
localhost=ip interna exto no va.

directmedia=yes
Pero revisa la extension configurada que parametro tiene esta.
en consoloa
sip show settings numero de la extensió.

hgmnetwork

Ok muchas gracias a todos probaré con eso a ver.

El router es el que ponen ellos por defecto un livebox creo

Si lo soluciono lo posteare para los que les pase lo mismo

navaismo

Esta IP la tienes abierta en el firewall: 92.185.176.157 ?
Esa IP es la de media(RTP) de acuerdo al debug y sí los puertos son random. intenta abriendo la IP.

hgmnetwork

si esta añadida a la lista, incluso el mismo portatil lo llevo a otro sitio y funciona bien, solo pasa cuando se conecta en ese lugar ( con el operador orange españa )

navaismo

Hmmm interesante, si conectas esa extensión via VPN? Una captura con tcpdump desde el server y una con wireshark desde la computadora en ese sitio para comparar los flujos SIP y RTP ¿es posble?

ixama

La vieja historia, probablemente es el router y la compresión gsip hay que quitarla. Por otro lado, también puede ser el router el que cause el problema y no permita el paso de los puertos correctamente, es un inconveniente normal con sip. Probaría el gsip, y también en lo posible con otro router comprobado que funcione con telefonia ip. Saludos.

hgmnetwork

Según he probado debe ser algo del router pongo el portátil en otra conexión y funciona manana intentaré por wifi
Y les comentodo

ixama

hgmnetwork Es lo mas probable, es el firewall de los routers, si es una conexión gpon probablemente tengas que pedirle al operador que te quite ciertas reglas del firewall, o bien verificar si podes cambiar el router o ponerlo en bridge.

supa

Disable SIP ALG on the router if it's enabled.

hgmnetwork

the sip alg are disabled but the problem are operator use 5060 for your phone

supa

hgmnetwork Using 5060 should not be a problem as long as SIP ALG is disabled. You can also try enable SIP ALG. It actually works on some routers now. It lets you do some nice things on your PBX such as canreinvite/directmedia.