Ramal externo cai após 6 segundos!

[Wagner De Queiroz]

Boa tarde Pessoal,

Já não chega o problema da Vono, agora tem mais um.

Eu estou com um elastix 2.4 aqui e quando tento logar qualquer ramal pelo endereço ip publico do meu asterisk ele registra normalmente e quando tento fazer qualquer ligação seja para um outro ramal, serviço ou ligacao externa após um tempo que gira em média 6 segundos a ligação cai. Olhando nos logs o asterisk tenta dar a dica do SIP_Retransmission, mas não sei ao certo o que esta faltando.

ja setei externalip=enderecoippublico (que é fixo) no sip, mas mesmo assim, continuo tendo o mesmo problema. Com o Asterisk antigo isso não ocorria (no mesmo ip e mesma máquina)

A mensagem no /var/asterisk/full é:

 WARNING[2688] chan_sip.c: Retransmission timeout reached on transmission xxxx. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6017ms with no response
 WARNING[2688] chan_sip.c: Hanging up call xxxx. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

Analisando o trafego SIP eu tenho isso:

Asterisk 11.6.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <mark...@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.6.0 currently running on pcpabx (pid = 2641)

<--- SIP read from UDP:ip.externo.do.ramal.com.br:24234 --->
INVITE sip:*4...@meu.servidor.asterisk.com.br SIP/2.0
Via: SIP/2.0/UDP ip.externo.do.ramal.com.br:24234;branch=z9hG4bK-d8754z-014a11411d0bee68-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:10...@ip.externo.do.ramal.com.br:24234>
To: "*43"<sip:*4...@meu.servidor.asterisk.com.br>
From: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 484

v=0
o=- 8 2 IN IP4 ip.externo.do.ramal.com.br
s=CounterPath X-Lite 3.0
c=IN IP4 ip.externo.do.ramal.com.br
t=0 0
m=audio 63872 RTP/AVP 107 119 100 106 6 0 97 105 98 8 102 3 5 101
a=alt:1 1 : LDRAgH1I gf6GaWtF ip.interno.ramal 63872
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:102 L16/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (12 headers 18 lines) ---
Sending to ip.externo.do.ramal.com.br:24234 (NAT)
Sending to ip.externo.do.ramal.com.br:24234 (NAT)
Using INVITE request as basis request - N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
Found peer '1000' for '1000' from ip.externo.do.ramal.com.br:24234

<--- Reliably Transmitting (NAT) to ip.externo.do.ramal.com.br:24234 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP ip.externo.do.ramal.com.br:24234;branch=z9hG4bK-d8754z-014a11411d0bee68-1---d8754z-;received=ip.externo.do.ramal.com.br;rport=24234
From: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
To: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as25bd3957
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 1 INVITE
Server: FPBX-2.8.1(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="596939fe"
Content-Length: 0
 

<------------>
Scheduling destruction of SIP dialog 'N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.' in 10688 ms (Method: INVITE)
 pcpabx*CLI>  0K
<--- SIP read from UDP:ip.externo.do.ramal.com.br:24234 --->
ACK sip:*4...@meu.servidor.asterisk.com.br SIP/2.0
Via: SIP/2.0/UDP ip.externo.do.ramal.com.br:24234;branch=z9hG4bK-d8754z-014a11411d0bee68-1---d8754z-;rport
To: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as25bd3957
From: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
 pcpabx*CLI>  0K
<--- SIP read from UDP:ip.externo.do.ramal.com.br:24234 --->
INVITE sip:*4...@meu.servidor.asterisk.com.br SIP/2.0
Via: SIP/2.0/UDP ip.externo.do.ramal.com.br:24234;branch=z9hG4bK-d8754z-943e86726a00594d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:10...@ip.externo.do.ramal.com.br:24234>
To: "*43"<sip:*4...@meu.servidor.asterisk.com.br>
From: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="1000",realm="asterisk",nonce="596939fe",uri="sip:*4...@meu.servidor.asterisk.com.br",response="d0535faf028f82b844d60902fa0dd6c5",algorithm=MD5
Content-Length: 484

v=0
o=- 8 2 IN IP4 ip.externo.do.ramal.com.br
s=CounterPath X-Lite 3.0
c=IN IP4 ip.externo.do.ramal.com.br
t=0 0
m=audio 63872 RTP/AVP 107 119 100 106 6 0 97 105 98 8 102 3 5 101
a=alt:1 1 : LDRAgH1I gf6GaWtF ip.interno.ramal 63872
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:102 L16/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (13 headers 18 lines) ---
Sending to ip.externo.do.ramal.com.br:24234 (NAT)
Using INVITE request as basis request - N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
Found peer '1000' for '1000' from ip.externo.do.ramal.com.br:24234
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 105
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 102
Found RTP audio format 3
Found RTP audio format 5
Found RTP audio format 101
Found unknown media description format BV32 for ID 107
Found unknown media description format BV32-FEC for ID 119
Found audio description format SPEEX for ID 100
Found unknown media description format SPEEX-FEC for ID 106
Found audio description format SPEEX for ID 97
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format iLBC for ID 98
Found audio description format L16 for ID 102
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h261|h263|h263p|h264), peer - audio=(gsm|ulaw|alaw|adpcm|speex|speex16|ilbc|slin16)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port ip.externo.do.ramal.com.br:63872
Peer doesn't provide video
Looking for *43 in from-internal (domain meu.servidor.asterisk.com.br)
list_route: hop: <sip:10...@ip.externo.do.ramal.com.br:24234>

<--- Transmitting (NAT) to ip.externo.do.ramal.com.br:24234 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ip.externo.do.ramal.com.br:24234;branch=z9hG4bK-d8754z-943e86726a00594d-1---d8754z-;received=ip.externo.do.ramal.com.br;rport=24234
From: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
To: "*43"<sip:*4...@meu.servidor.asterisk.com.br>
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 2 INVITE
Server: FPBX-2.8.1(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*4...@ip.interno.asterisk:5060>
Content-Length: 0
 

<------------>
Audio is at 15834
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to ip.externo.do.ramal.com.br:24234 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip.externo.do.ramal.com.br:24234;branch=z9hG4bK-d8754z-943e86726a00594d-1---d8754z-;received=ip.externo.do.ramal.com.br;rport=24234
From: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
To: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as269d7b92
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 2 INVITE
Server: FPBX-2.8.1(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*4...@ip.interno.asterisk:5060>
Content-Type: application/sdp
Content-Length: 278
 
v=0
o=root 357200242 357200242 IN IP4 ip.interno.asterisk
s=Asterisk PBX 11.6.0
c=IN IP4 ip.interno.asterisk
t=0 0
m=audio 15834 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
 pcpabx*CLI>  0K
<--- SIP read from UDP:187.10.124.185:53594 --->


<------------->
 pcpabx*CLI>  0KRetransmitting #1 (NAT) to ip.externo.do.ramal.com.br:24234:
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip.externo.do.ramal.com.br:24234;branch=z9hG4bK-d8754z-943e86726a00594d-1---d8754z-;received=ip.externo.do.ramal.com.br;rport=24234
From: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
To: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as269d7b92
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 2 INVITE
Server: FPBX-2.8.1(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*4...@ip.interno.asterisk:5060>
Content-Type: application/sdp
Content-Length: 278
 
v=0
o=root 357200242 357200242 IN IP4 ip.interno.asterisk
s=Asterisk PBX 11.6.0
c=IN IP4 ip.interno.asterisk
t=0 0
m=audio 15834 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
 pcpabx*CLI>  0KRetransmitting #2 (NAT) to ip.externo.do.ramal.com.br:24234:
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip.externo.do.ramal.com.br:24234;branch=z9hG4bK-d8754z-943e86726a00594d-1---d8754z-;received=ip.externo.do.ramal.com.br;rport=24234
From: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
To: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as269d7b92
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 2 INVITE
Server: FPBX-2.8.1(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*4...@ip.interno.asterisk:5060>
Content-Type: application/sdp
Content-Length: 278
 
v=0
o=root 357200242 357200242 IN IP4 ip.interno.asterisk
s=Asterisk PBX 11.6.0
c=IN IP4 ip.interno.asterisk
t=0 0
m=audio 15834 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

Retransmitting #3 (NAT) to ip.externo.do.ramal.com.br:24234:
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip.externo.do.ramal.com.br:24234;branch=z9hG4bK-d8754z-943e86726a00594d-1---d8754z-;received=ip.externo.do.ramal.com.br;rport=24234
From: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
To: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as269d7b92
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 2 INVITE
Server: FPBX-2.8.1(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*4...@ip.interno.asterisk:5060>
Content-Type: application/sdp
Content-Length: 278
 
v=0
o=root 357200242 357200242 IN IP4 ip.interno.asterisk
s=Asterisk PBX 11.6.0
c=IN IP4 ip.interno.asterisk
t=0 0
m=audio 15834 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
 Retransmitting #4 (NAT) to ip.externo.do.ramal.com.br:24234:
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip.externo.do.ramal.com.br:24234;branch=z9hG4bK-d8754z-943e86726a00594d-1---d8754z-;received=ip.externo.do.ramal.com.br;rport=24234
From: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
To: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as269d7b92
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 2 INVITE
Server: FPBX-2.8.1(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*4...@ip.interno.asterisk:5060>
Content-Type: application/sdp
Content-Length: 278
 
v=0
o=root 357200242 357200242 IN IP4 ip.interno.asterisk
s=Asterisk PBX 11.6.0
c=IN IP4 ip.interno.asterisk
t=0 0
m=audio 15834 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

Retransmitting #5 (NAT) to ip.externo.do.ramal.com.br:24234:
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip.externo.do.ramal.com.br:24234;branch=z9hG4bK-d8754z-943e86726a00594d-1---d8754z-;received=ip.externo.do.ramal.com.br;rport=24234
From: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
To: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as269d7b92
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 2 INVITE
Server: FPBX-2.8.1(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*4...@ip.interno.asterisk:5060>
Content-Type: application/sdp
Content-Length: 278
 
v=0
o=root 357200242 357200242 IN IP4 ip.interno.asterisk
s=Asterisk PBX 11.6.0
c=IN IP4 ip.interno.asterisk
t=0 0
m=audio 15834 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

Really destroying SIP dialog 'YjBjNGY5ZTNlNTczYjA4ZTQ2OWViZjJlZjY2MjhhYzE.' Method: REGISTER
Really destroying SIP dialog 'MzhmODc1NDE2NGQ3Yjk2MmQ1M2Q1YzkyZWQzNGRmNWY.' Method: SUBSCRIBE
 Retransmitting #6 (NAT) to ip.externo.do.ramal.com.br:24234:
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip.externo.do.ramal.com.br:24234;branch=z9hG4bK-d8754z-943e86726a00594d-1---d8754z-;received=ip.externo.do.ramal.com.br;rport=24234
From: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
To: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as269d7b92
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 2 INVITE
Server: FPBX-2.8.1(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*4...@ip.interno.asterisk:5060>
Content-Type: application/sdp
Content-Length: 278
 
v=0
o=root 357200242 357200242 IN IP4 ip.interno.asterisk
s=Asterisk PBX 11.6.0
c=IN IP4 ip.interno.asterisk
t=0 0
m=audio 15834 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
 Scheduling destruction of SIP dialog 'N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.' in 10688 ms (Method: INVITE)
set_destination: Parsing <sip:10...@ip.externo.do.ramal.com.br:24234> for address/port to send to
set_destination: set destination to ip.externo.do.ramal.com.br:24234
Reliably Transmitting (NAT) to ip.externo.do.ramal.com.br:24234:
BYE sip:10...@ip.externo.do.ramal.com.br:24234 SIP/2.0
Via: SIP/2.0/UDP ip.interno.asterisk:5060;branch=z9hG4bK6863c2bd;rport
Max-Forwards: 70
From: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as269d7b92
To: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 102 BYE
User-Agent: FPBX-2.8.1(11.6.0)
Proxy-Authorization: Digest username="1000", realm="asterisk", algorithm=MD5, uri="sip:meu.servidor.asterisk.com.br", nonce="596939fe", response="6d6da7542ca7176d16f0f21b9da1ca87"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
 

---
 Retransmitting #1 (NAT) to ip.externo.do.ramal.com.br:24234:
BYE sip:10...@ip.externo.do.ramal.com.br:24234 SIP/2.0
Via: SIP/2.0/UDP ip.interno.asterisk:5060;branch=z9hG4bK6863c2bd;rport
Max-Forwards: 70
From: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as269d7b92
To: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 102 BYE
User-Agent: FPBX-2.8.1(11.6.0)
Proxy-Authorization: Digest username="1000", realm="asterisk", algorithm=MD5, uri="sip:meu.servidor.asterisk.com.br", nonce="596939fe", response="6d6da7542ca7176d16f0f21b9da1ca87"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
 

---
 <--- SIP read from UDP:ip.externo.do.ramal.com.br:24234 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip.interno.asterisk:5060;branch=z9hG4bK6863c2bd;rport=5060;received=187.121.47.91
Contact: <sip:10...@ip.externo.do.ramal.com.br:24234>
To: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
From: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as269d7b92
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 102 BYE
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.' Method: INVITE
 
<--- SIP read from UDP:ip.externo.do.ramal.com.br:24234 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip.interno.asterisk:5060;branch=z9hG4bK6863c2bd;rport=5060;received=187.121.47.91
Contact: <sip:10...@ip.externo.do.ramal.com.br:24234>
To: "1000"<sip:10...@meu.servidor.asterisk.com.br>;tag=6c6be106
From: "*43"<sip:*4...@meu.servidor.asterisk.com.br>;tag=as269d7b92
Call-ID: N2VlOTY4YWM3MDE5NjMxM2IyNGM1YzVlMDU2NzdiMTE.
CSeq: 102 BYE
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
 
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups

[Wagner De Queiroz]

Boa noite a todos da Lista.

A dois dias atrás estava eu com um problema cabeludo da ligação que cai após 6 segundos, ninguém respondeu até o momento, não se sabe pq ninguém deu atenção pq era muito idiota a solução ou é impossível de resolver. Após muita peleja, eu desconfiava do nat, daquela linha do externip no sip.conf.

O curioso é que resolvi insistir nesta vertente e descobri a solução. Estou passando o link de onde encontrei a solução, pois é útil para quem não tem muita intimidade com o Elastix (como eu, que fiz curso de Elastix)

Básicamente a linha externip=ipexterno deve ser colocado num outro arquivo, já que o reload da pagina web reescreve o sip.conf.

O arquivo é o /etc/asterisk/sip_nat.conf

só que, tem uma pegadinha: vc precisa dizer mais algumas coisinhas:

Veja que linhas são:

nat=yes
externip=200.123.123.123

localnet=192.168.0.0/255.255.255.0

externrefresh=120

O engraçado é que por padrão este arquivo não tem informação alguma

no meu caso, enquanto escrevo este email, o localnet estava errado e não condizia com a minha placa de rede. então talvez essa liha localnet possa não ter utilidade.
Como é um servidor de produção, não pretendo ficar brincando com ele no momento para testar qual foi a linha que fez a diferença, pois eu já havia colocado neste sip_nat aquela linha do externip e reiniciado o pabx e não havia funcionado.

O bom é que agora esta funcionando e fica a dica para os novatos não cairem nesta armadilha que caí

O link salvador:

http://and-inside.com/elastix-external-call-disconnect-after-6-seconds/

[NTECH SOLUÇÕES EM TELECOM]

Olá Wagner,

Parabéns pela iniciativa em postar o que resolveu o problema. Particularmente não havia passado por este problema, mas a sua dica já ficou anotada na minha lista.

Atenciosamente,

Neimar Bueno

---

Date: Wed, 27 Nov 2013 19:04:53 -0200
Subject: [ElastixBrasil] [RESOLVIDO] Ramal externo cai após 6 segundos!
From: wagnerd...@gmail.com
To: elasti...@googlegroups.com

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