[AsteriskBrasil] Problema com Asterisk 1.8
Emerson Mochon Borsatti
borsatti.em em gmail.com
Sexta Junho 17 16:50:25 BRT 2011
Boa tarde pessoal,
Estou com um problema com meu asterisk 1.8 rodando no Ubuntu. Estava tudo
funcionando perfeitamente há varios meses, mas tive de dar um boot na
máquina, estou com vários problemas.
Usuários internos à rede, ligam e recebem ligação sem problema. Agora *quem
esta externo (conecta de um ip público) ouve mas não consegue falar* (a
outra ponta nãp recebe o áudio).
O que pode ser?
Já testei o nat como comedia, force_rport, yes, no , tb já
testei directmedia=no ; directrtpsetup=yes e no
Eis algumas configurações:
[3124]
CID_3124 = XXXX3124
type=friend
context=imti-broad
callerid=Alguem cel <3124>
secret=3124
host=dynamic
nat=yes
directmedia=no
directrtpsetup=yes
mailbox=
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
allow=g723.1
allow=ilbc
qualify=yes
amaflags=billing
Alguns trechos do sip debug:
<------------->
--- (15 headers 12 lines) ---
Sending to 189.99.147.248:50229 (no NAT)
Using INVITE request as basis request -
98880C4AF1C2D5A495758A0536857027ECB5C605
Found peer '3124' for '3124' from 189.99.147.248:50229
<--- Reliably Transmitting (no NAT) to 189.99.147.248:50229 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 189.99.147.248:50229
;branch=z9hG4bKNYHl4TEKjJOEdah1;received=189.99.147.248;rport=50229
From: "Napa" <sip:3124 em voip.capital.ms.gov.br
>;tag=F653FF04FBC97E9DDA8F463736A1EACD
To: <sip:091012026 em voip.capital.ms.gov.br>;tag=as4b635a6b
Call-ID: 98880C4AF1C2D5A495758A0536857027ECB5C605
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="", nonce="3b9b9f7a"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'98880C4AF1C2D5A495758A0536857027ECB5C605' in 86400 ms (Method: INVITE)
<--- SIP read from UDP:189.99.147.248:50229 --->
<--- SIP read from UDP:172.17.0.27:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.17.0.30:5060
;branch=z9hG4bK63996783;received=172.17.0.30
From: "Napa cel" <sip:33043124 em 172.17.0.30>;tag=as5f695838
To: <sip:91012026 em 172.17.0.27>;tag=as0c5ebd9a
Call-ID: 4b3c1ec75a15b5055f6135bd5532e081 em 172.17.0.30:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:91012026 em 172.17.0.27>
Content-Length: 0
<------------->
<------------->
Reliably Transmitting (no NAT) to 189.99.147.248:50229:
OPTIONS sip:3124 em 189.99.147.248:50229;rinstance=C0E3D765 SIP/2.0
Via: SIP/2.0/UDP 172.17.0.30:5060;branch=z9hG4bK63c167cf
Max-Forwards: 30
From: "asterisk" <sip:asterisk em 172.17.0.30>;tag=as36b7d17f
To: <sip:3124 em 189.99.147.248:50229;rinstance=C0E3D765>
Contact: <sip:asterisk em 172.17.0.30:5060>
Call-ID: 11fdeb005a975be12673b175417d892d em 172.17.0.30:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.0
Date: Fri, 17 Jun 2011 19:09:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
Really destroying SIP dialog 'pSaqJ-7hH0441f2 em capital.ms.gov.br' Method:
REGISTER
Retransmitting #1 (no NAT) to 189.99.147.248:50229:
OPTIONS sip:3124 em 189.99.147.248:50229;rinstance=C0E3D765 SIP/2.0
Via: SIP/2.0/UDP 172.17.0.30:5060;branch=z9hG4bK63c167cf
Max-Forwards: 30
From: "asterisk" <sip:asterisk em 172.17.0.30>;tag=as36b7d17f
To: <sip:3124 em 189.99.147.248:50229;rinstance=C0E3D765>
Contact: <sip:asterisk em 172.17.0.30:5060>
Call-ID: 11fdeb005a975be12673b175417d892d em 172.17.0.30:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.0
Date: Fri, 17 Jun 2011 19:09:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
Really destroying SIP dialog '12743f7a63a87dae3acfb44f781f59a5 em 172.17.0.27'
Method: ACK
<--- SIP read from UDP:10.4.0.152:5060 --->
<------------->
<--- SIP read from UDP:189.99.147.248:50229 --->
SIP/2.0 200 OK
From: "asterisk" <sip:asterisk em 172.17.0.30>;tag=as36b7d17f
Call-ID: 11fdeb005a975be12673b175417d892d em 172.17.0.30:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 172.17.0.30:5060;branch=z9hG4bK63c167cf
To: <sip:3124 em 189.99.147.248:50229;rinstance=C0E3D765>
Contact: <sip:3124 em 189.99.147.248:50229>
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces
Supported: path
Accept: application/sdp
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '
11fdeb005a975be12673b175417d892d em 172.17.0.30:5060' Method: OPTIONS
<--- SIP read from UDP:172.17.0.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.0.30:5060
;branch=z9hG4bK63996783;received=172.17.0.30
From: "Napa cel" <sip:33043124 em 172.17.0.30>;tag=as5f695838
To: <sip:91012026 em 172.17.0.27>;tag=as0c5ebd9a
Call-ID: 4b3c1ec75a15b5055f6135bd5532e081 em 172.17.0.30:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:91012026 em 172.17.0.27>
Content-Type: application/sdp
Content-Length: 331
v=0
o=root 810974455 810974455 IN IP4 172.17.0.27
s=Asterisk PBX 1.6.2.14
c=IN IP4 172.17.0.27
t=0 0
m=audio 10922 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 15 lines) ---
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c
(ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.17.0.27:10922
list_route: hop: <sip:91012026 em 172.17.0.27>
set_destination: Parsing <sip:91012026 em 172.17.0.27> for address/port to send
to
set_destination: set destination to 172.17.0.27:5060
Transmitting (no NAT) to 172.17.0.27:5060:
ACK sip:91012026 em 172.17.0.27 SIP/2.0
Via: SIP/2.0/UDP 172.17.0.30:5060;branch=z9hG4bK4eb008f1
Max-Forwards: 30
From: "Napa cel" <sip:33043124 em 172.17.0.30>;tag=as5f695838
To: <sip:91012026 em 172.17.0.27>;tag=as0c5ebd9a
Contact: <sip:33043124 em 172.17.0.30:5060>
Call-ID: 4b3c1ec75a15b5055f6135bd5532e081 em 172.17.0.30:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.0
Content-Length: 0
---
-- SIP/voipe1-000001d8 answered SIP/3124-000001d7
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 189.99.147.248:50229 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.99.147.248:50229
;branch=z9hG4bKErQ5G8Qstb3cnCaI;received=189.99.147.248;rport=50229
From: "Napa" <sip:3124 em voip.capital.ms.gov.br
>;tag=F653FF04FBC97E9DDA8F463736A1EACD
To: <sip:091012026 em voip.capital.ms.gov.br>;tag=as63c3ce2e
Call-ID: 98880C4AF1C2D5A495758A0536857027ECB5C605
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:091012026 em 172.17.0.30:5060>
Content-Type: application/sdp
Content-Length: 327
v=0
o=root 1704978075 1704978075 IN IP4 172.17.0.30
s=Asterisk PBX 1.8.0
c=IN IP4 172.17.0.30
t=0 0
m=audio 12602 RTP/AVP 0 8 3 102 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
-- Locally bridging SIP/3124-000001d7 and SIP/voipe1-000001d8
--
Att.,
*Emerson M. Borsatti*
Mandriva Linux System Administrator Certified
ITIL V3 Intermediate Certified
ISO 20000 Certified
*(67) 9101-2026*
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