[AsteriskBrasil] Skype connect + asterisk - recebimento ligações
Anivaldo
w.nelson em ig.com.br
Terça Setembro 16 16:53:44 BRT 2014
Boa tarde lista.
Estou testando em casa o skype connect, pois tenho interesse de receber
ligações no asterisk de outros usuarios skype que estejam utilizando o
software skype no computador pessoal.
No entanto, o asterisk não está recebendo as chamadas do skype. Não
apresenta nenhum erro. já coloquei o sip debug on e não ocorre nenhum
evento.
Alguém já utilizou skype connect e recebe ligações normalmente ?
Dei uma pesquisada na lista e encontrei relatos de pessoas que não
estavam conseguindo efetuar ligações, mas estavam recebendo normalmente.
A conta skype esta registrando
anivaldo-G31M-S2L*CLI> sip show registry
Host dnsmgr Username Refresh State
Reg.Time
sip.skype.com:5060 N 990510002341 105
Registered Tue, 16 Sep 2014 15:15:18
1 SIP registrations.
Meu sip.conf está assim
[general]
register => 99051000234174:<senha>@sip.skype.com:5060/99051000234174
[skype]
username=99051000234174
fromuser=99051000234174
secret=<senha>
type=friend
fromdomain=sip.skype.com
host=sip.skype.com
context=incoming
dtmfmode=rfc2833
nat=no
qualify=yes
allow=g729
allow=ulaw
allow=alaw
port=5060
Com o sip debug on tenho esses eventos:
[Sep 16 15:37:10] Reliably Transmitting (NAT) to 192.168.1.2:54521:
OPTIONS
sip:20 em 189.70.117.152:54521;rinstance=8ce13a5ae824f3a4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK10316817;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 192.168.1.2>;tag=as3af22fe5
To: <sip:20 em 189.70.117.152:54521;rinstance=8ce13a5ae824f3a4;transport=UDP>
Contact: <sip:asterisk em 192.168.1.2:5060>
Call-ID: 5ac6189277e9a250777bedd54e30a0fa em 192.168.1.2:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.1
Date: Tue, 16 Sep 2014 18:37:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[Sep 16 15:37:10]
<--- SIP read from UDP:192.168.1.2:54521 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK10316817;rport=5060
Contact: <sip:192.168.1.2:54521>
To:
<sip:20 em 189.70.117.152:54521;rinstance=8ce13a5ae824f3a4;transport=UDP>;tag=185a1e66
From: "asterisk"<sip:asterisk em 192.168.1.2>;tag=as3af22fe5
Call-ID: 5ac6189277e9a250777bedd54e30a0fa em 192.168.1.2:5060
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: Z 3.2.21357 r21103
Allow-Events: presence, kpml
Content-Length: 0
<------------->
[Sep 16 15:37:10] --- (14 headers 0 lines) ---
[Sep 16 15:37:10] Really destroying SIP dialog
'5ac6189277e9a250777bedd54e30a0fa em 192.168.1.2:5060' Method: OPTIONS
[Sep 16 15:37:12]
<--- SIP read from UDP:192.168.1.2:54521 --->
<------------->
[Sep 16 15:37:13] Reliably Transmitting (NAT) to 63.209.144.201:5060:
OPTIONS sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6df58385;rport
Max-Forwards: 70
From: "asterisk" <sip:99051000234174 em 192.168.1.2>;tag=as6f943042
To: <sip:sip.skype.com>
Contact: <sip:99051000234174 em 192.168.1.2:5060>
Call-ID: 7b8094001526e29a2fd044706e9a5a41 em 192.168.1.2:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.1
Date: Tue, 16 Sep 2014 18:37:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[Sep 16 15:37:13]
<--- SIP read from UDP:63.209.144.201:5060 --->
SIP/2.0 200 OK
From: "asterisk" <sip:99051000234174 em 192.168.1.2>;tag=as6f943042
To: <sip:sip.skype.com>;tag=c990d13f-4cb577d9-0-7fc324f9-0
Call-ID: 7b8094001526e29a2fd044706e9a5a41 em 192.168.1.2:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 63.209.144.201:5060;branch=z9hG4bK6df58385;rport=5060
Content-Length: 0
<------------->
[Sep 16 15:37:13] --- (7 headers 0 lines) ---
[Sep 16 15:37:13] Really destroying SIP dialog
'7b8094001526e29a2fd044706e9a5a41 em 192.168.1.2:5060' Method: OPTIONS
[Sep 16 15:37:42]
<--- SIP read from UDP:192.168.1.2:54521 --->
<------------->
[Sep 16 15:37:59] > doing dnsmgr_lookup for 'sip.skype.com'
[Sep 16 15:37:59] > ast_get_srv: SRV lookup for
'_sip._udp.sip.skype.com' mapped to host 1.sip.skype.com, port 5060
[Sep 16 15:37:59] REGISTER 10 headers, 0 lines
[Sep 16 15:37:59] Reliably Transmitting (NAT) to 63.209.144.201:5060:
REGISTER sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK66c35710;rport
Max-Forwards: 70
From: <sip:99051000234174 em sip.skype.com>;tag=as180cbab7
To: <sip:99051000234174 em sip.skype.com>
Call-ID: 586f81a877c050f755343c171c2f20e5 em 127.0.0.1
CSeq: 107 REGISTER
User-Agent: Asterisk PBX 1.8.23.1
Expires: 120
Contact: <sip:99051000234174 em 192.168.1.2:5060>
Content-Length: 0
---
[Sep 16 15:38:00]
<--- SIP read from UDP:63.209.144.201:5060 --->
SIP/2.0 200 OK
From: <sip:99051000234174 em sip.skype.com>;tag=as180cbab7
To: <sip:99051000234174 em sip.skype.com>;tag=c990d13f-4f7c8f26-0-7fc3daf7-0
Call-ID: 586f81a877c050f755343c171c2f20e5 em 127.0.0.1
CSeq: 107 REGISTER
Via: SIP/2.0/UDP 63.209.144.201:5060;branch=z9hG4bK66c35710;rport=5060
Expires: 120
Contact: <sip:99051000234174 em 189.70.117.152:5060>;expires=120
Content-Length: 0
<------------->
[Sep 16 15:38:00] --- (9 headers 0 lines) ---
[Sep 16 15:38:00] Scheduling destruction of SIP dialog
'586f81a877c050f755343c171c2f20e5 em 127.0.0.1' in 32000 ms (Method: REGISTER)
[Sep 16 15:38:10] Reliably Transmitting (NAT) to 192.168.1.2:54521:
OPTIONS
sip:20 em 189.70.117.152:54521;rinstance=8ce13a5ae824f3a4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK633689ec;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 192.168.1.2>;tag=as3c3bc47b
To: <sip:20 em 189.70.117.152:54521;rinstance=8ce13a5ae824f3a4;transport=UDP>
Contact: <sip:asterisk em 192.168.1.2:5060>
Call-ID: 3b88a799237779ee5aab4ed66be8fbe5 em 192.168.1.2:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.1
Date: Tue, 16 Sep 2014 18:38:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[Sep 16 15:38:10]
<--- SIP read from UDP:192.168.1.2:54521 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK633689ec;rport=5060
Contact: <sip:192.168.1.2:54521>
To:
<sip:20 em 189.70.117.152:54521;rinstance=8ce13a5ae824f3a4;transport=UDP>;tag=cfd0bf55
From: "asterisk"<sip:asterisk em 192.168.1.2>;tag=as3c3bc47b
Call-ID: 3b88a799237779ee5aab4ed66be8fbe5 em 192.168.1.2:5060
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: Z 3.2.21357 r21103
Allow-Events: presence, kpml
Content-Length: 0
<------------->
[Sep 16 15:38:10] --- (14 headers 0 lines) ---
[Sep 16 15:38:10] Really destroying SIP dialog
'3b88a799237779ee5aab4ed66be8fbe5 em 192.168.1.2:5060' Method: OPTIONS
[Sep 16 15:38:12]
<--- SIP read from UDP:192.168.1.2:54521 --->
<------------->
[Sep 16 15:38:13] Reliably Transmitting (NAT) to 63.209.144.201:5060:
OPTIONS sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5b2cafe4;rport
Max-Forwards: 70
From: "asterisk" <sip:99051000234174 em 192.168.1.2>;tag=as20892cf3
To: <sip:sip.skype.com>
Contact: <sip:99051000234174 em 192.168.1.2:5060>
Call-ID: 118f26a75258ca8609329aca382045b0 em 192.168.1.2:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.1
Date: Tue, 16 Sep 2014 18:38:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[Sep 16 15:38:13]
<--- SIP read from UDP:63.209.144.201:5060 --->
SIP/2.0 200 OK
From: "asterisk" <sip:99051000234174 em 192.168.1.2>;tag=as20892cf3
To: <sip:sip.skype.com>;tag=c990d13f-504e14c7-0-7fc41097-0
Call-ID: 118f26a75258ca8609329aca382045b0 em 192.168.1.2:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 63.209.144.201:5060;branch=z9hG4bK5b2cafe4;rport=5060
Content-Length: 0
<------------->
[Sep 16 15:38:13] --- (7 headers 0 lines) ---
[Sep 16 15:38:13] Really destroying SIP dialog
'118f26a75258ca8609329aca382045b0 em 192.168.1.2:5060' Method: OPTIONS
[Sep 16 15:38:32] Really destroying SIP dialog
'586f81a877c050f755343c171c2f20e5 em 127.0.0.1' Method: REGISTER
[Sep 16 15:38:42]
<--- SIP read from UDP:192.168.1.2:54521 --->
<------------->
Obrigado
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