[Kamailio-Users-ES] ayuda migrando de openser 1.3x a 1.4x opensips

10 Mar 2009


      señores tengo algo peculiar que me esta pasando , estoy tratando de
migrar mi proxy de la version 1.3x a opensip 1.4.4 , he estado leyendo
y se que algunas cosas cambiaron pero lo raro es que cuando llamo de
una extensión a otro a me salta directo al voicemail como si estuviera
ocupado ... y eso hablando de extensiones en la misma red , no
externas ...
este pegon si lo tengo duro , no se por donde buscarle ...
disculpen que añada este trozote de log sip ...
interface: any
filter: (ip) and ( port 5060 )
#
U +1.038668 192.168.10.19:5060 -> 192.168.10.3:5060
INVITE sip:201@192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-95de5398
From: sip:200@192.168.10.3;tag=327eaf6c2c6324f1o0
To: "kamailio-14x" sip:201@192.168.10.3
Call-ID: acc6c148-4348dcc5@192.168.10.19
CSeq: 101 INVITE
Max-Forwards: 70
Contact: sip:200@192.168.10.19:5060
Expires: 240
User-Agent: Linksys/SPA942-6.1.3(a)
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 47854 47854 IN IP4 192.168.10.19
s=-
c=IN IP4 192.168.10.19
t=0 0
m=audio 16402 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
#
U +0.000438 192.168.10.3:5060 -> 192.168.10.19:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-95de5398
From: sip:200@192.168.10.3;tag=327eaf6c2c6324f1o0
To: "kamailio-14x"
sip:201@192.168.10.3;tag=d4e9e39d125187795ad79ae40f9b4f9f.62fe
Call-ID: acc6c148-4348dcc5@192.168.10.19
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="192.168.10.3",
nonce="49b6c7130000001145db79bf52de222d013769f25c8daa66"
Server: OpenSIPS (1.4.4-notls (i386/linux))
Content-Length: 0
#
U +0.015654 192.168.10.19:5060 -> 192.168.10.3:5060
ACK sip:201@192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-95de5398
From: sip:200@192.168.10.3;tag=327eaf6c2c6324f1o0
To: "kamailio-14x"
sip:201@192.168.10.3;tag=d4e9e39d125187795ad79ae40f9b4f9f.62fe
Call-ID: acc6c148-4348dcc5@192.168.10.19
CSeq: 101 ACK
Max-Forwards: 70
Contact: sip:200@192.168.10.19:5060
User-Agent: Linksys/SPA942-6.1.3(a)
Content-Length: 0
#
U +0.003845 192.168.10.19:5060 -> 192.168.10.3:5060
INVITE sip:201@192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-82527624
From: sip:200@192.168.10.3;tag=327eaf6c2c6324f1o0
To: "kamailio-14x" sip:201@192.168.10.3
Call-ID: acc6c148-4348dcc5@192.168.10.19
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="200",realm="192.168.10.3",nonce="49b6c7130000001145db79bf52de222d013769f25c8daa66",uri="sip:201@192.168.10.3",algorithm=MD5,response="f61b7ab5b597daea13911d7630e9dc4a"
Contact: sip:200@192.168.10.19:5060
Expires: 240
User-Agent: Linksys/SPA942-6.1.3(a)
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 47854 47854 IN IP4 192.168.10.19
s=-
c=IN IP4 192.168.10.19
t=0 0
m=audio 16402 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
#
U +0.001341 192.168.10.3:5060 -> 192.168.10.19:5060
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-82527624
From: sip:200@192.168.10.3;tag=327eaf6c2c6324f1o0
To: "kamailio-14x" sip:201@192.168.10.3
Call-ID: acc6c148-4348dcc5@192.168.10.19
CSeq: 102 INVITE
Server: OpenSIPS (1.4.4-notls (i386/linux))
Content-Length: 0
#
U +0.000131 192.168.10.3:5060 -> 192.168.10.3:5070
INVITE sip:201@192.168.10.3:5070 SIP/2.0
Record-Route: sip:192.168.10.3;lr=on;ftag=327eaf6c2c6324f1o0
Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bK5afe.11d6393.0
Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-82527624
From: sip:200@192.168.10.3;tag=327eaf6c2c6324f1o0
To: "kamailio-14x" sip:201@192.168.10.3
Call-ID: acc6c148-4348dcc5@192.168.10.19
CSeq: 102 INVITE
Max-Forwards: 69
Proxy-Authorization: Digest
username="200",realm="192.168.10.3",nonce="49b6c7130000001145db79bf52de222d013769f25c8daa66",uri="sip:201@192.168.10.3",algorithm=MD5,response="f61b7ab5b597daea13911d7630e9dc4a"
Contact: sip:200@192.168.10.19:5060
Expires: 240
User-Agent: Linksys/SPA942-6.1.3(a)
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 47854 47854 IN IP4 192.168.10.19
s=-
c=IN IP4 192.168.10.19
t=0 0
m=audio 16402 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
#
U +0.000688 192.168.10.3:5070 -> 192.168.10.3:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.10.3;branch=z9hG4bK5afe.11d6393.0;received=192.168.10.3
Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-82527624
Record-Route: sip:192.168.10.3;lr=on;ftag=327eaf6c2c6324f1o0
From: sip:200@192.168.10.3;tag=327eaf6c2c6324f1o0
To: "kamailio-14x" sip:201@192.168.10.3
Call-ID: acc6c148-4348dcc5@192.168.10.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:201@192.168.10.3:5070
Content-Length: 0
#
U +0.000598 192.168.10.3:5070 -> 192.168.10.3:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.10.3;branch=z9hG4bK5afe.11d6393.0;received=192.168.10.3
Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-82527624
Record-Route: sip:192.168.10.3;lr=on;ftag=327eaf6c2c6324f1o0
From: sip:200@192.168.10.3;tag=327eaf6c2c6324f1o0
To: "kamailio-14x" sip:201@192.168.10.3;tag=as4a245f87
Call-ID: acc6c148-4348dcc5@192.168.10.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:201@192.168.10.3:5070
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 9235 9235 IN IP4 192.168.10.3
s=session
c=IN IP4 192.168.10.3
t=0 0
m=audio 17814 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
#
U +0.000278 192.168.10.3:5060 -> 192.168.10.19:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-82527624
Record-Route: sip:192.168.10.3;lr=on;ftag=327eaf6c2c6324f1o0
From: sip:200@192.168.10.3;tag=327eaf6c2c6324f1o0
To: "kamailio-14x" sip:201@192.168.10.3;tag=as4a245f87
Call-ID: acc6c148-4348dcc5@192.168.10.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:201@192.168.10.3:5070
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 9235 9235 IN IP4 192.168.10.3
s=session
c=IN IP4 192.168.10.3
t=0 0
m=audio 17814 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
#
U +0.018455 192.168.10.19:5060 -> 192.168.10.3:5060
ACK sip:201@192.168.10.3:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-c1874d9d
From: sip:200@192.168.10.3;tag=327eaf6c2c6324f1o0
To: "kamailio-14x" sip:201@192.168.10.3;tag=as4a245f87
Call-ID: acc6c148-4348dcc5@192.168.10.19
CSeq: 102 ACK
Max-Forwards: 70
Route: sip:192.168.10.3;lr=on;ftag=327eaf6c2c6324f1o0
Proxy-Authorization: Digest
username="200",realm="192.168.10.3",nonce="49b6c7130000001145db79bf52de222d013769f25c8daa66",uri="sip:201@192.168.10.3",algorithm=MD5,response="f61b7ab5b597daea13911d7630e9dc4a"
Contact: sip:200@192.168.10.19:5060
User-Agent: Linksys/SPA942-6.1.3(a)
Content-Length: 0
#
U +0.000206 192.168.10.3:5060 -> 192.168.10.3:5070
ACK sip:201@192.168.10.3:5070 SIP/2.0
Record-Route: sip:192.168.10.3;lr=on;ftag=327eaf6c2c6324f1o0
Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bK5afe.11d6393.2
Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-c1874d9d
From: sip:200@192.168.10.3;tag=327eaf6c2c6324f1o0
To: "kamailio-14x" sip:201@192.168.10.3;tag=as4a245f87
Call-ID: acc6c148-4348dcc5@192.168.10.19
CSeq: 102 ACK
Max-Forwards: 69
Proxy-Authorization: Digest
username="200",realm="192.168.10.3",nonce="49b6c7130000001145db79bf52de222d013769f25c8daa66",uri="sip:201@192.168.10.3",algorithm=MD5,response="f61b7ab5b597daea13911d7630e9dc4a"
Contact: sip:200@192.168.10.19:5060
User-Agent: Linksys/SPA942-6.1.3(a)
Content-Length: 0
#
U +4.600183 192.168.10.3:5070 -> 192.168.10.3:5060
OPTIONS sip:192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.3:5070;branch=z9hG4bK60d4a50f;rport
From: "asterisk" sip:asterisk@192.168.10.3:5070;tag=as5417e9bc
To: sip:192.168.10.3
Contact: sip:asterisk@192.168.10.3:5070
Call-ID: 2ec730764cddd1bf15e296f168d9b0bf@192.168.10.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 10 Mar 2009 20:00:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
#
U +0.000269 192.168.10.3:5060 -> 192.168.10.3:5070
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.3:5070;branch=z9hG4bK60d4a50f;rport=5070
From: "asterisk" sip:asterisk@192.168.10.3:5070;tag=as5417e9bc
To: sip:192.168.10.3;tag=d4e9e39d125187795ad79ae40f9b4f9f.0e65
Call-ID: 2ec730764cddd1bf15e296f168d9b0bf@192.168.10.3
CSeq: 102 OPTIONS
Proxy-Authenticate: Digest realm="192.168.10.3",
nonce="49b6c717000000127f72fab6bd7f924f5d24d09098273864"
Server: OpenSIPS (1.4.4-notls (i386/linux))
Content-Length: 0
mi segundo problemilla es con los permisos tengo algunas usuarios que
hacer llamadas a la pstn y otros a algunos proveedores voip , pero
esto no lo respeta el proxy cualquiera puede llamar a la pstn etc...
y la extension 201 no tiene permisos para las llamadas locales , tengo
el modulo group.so cargado y el permissions.so
dentro del opensips.cfg tengo añadido este trozo de codigo
if (uri=~"^sip:[2346597][0-9]{6}@.*") {
#        if (is_user_in("From", "local")){
       if (is_user_in("credentials", "local")){
         route(4);
y veo que esto se mantiene en esta version ....
añado algunos log
U +0.963418 192.168.10.30:5064 -> 192.168.10.3:5060
INVITE sip:2685249@192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK8cb08d7b02da46ac
From: "Opensips-14x" sip:201@192.168.10.3;tag=f89d240daba57386
To: sip:2685249@192.168.10.3
Contact: sip:201@192.168.10.30:5064;transport=udp
Supported: replaces, timer, path
Call-ID: 52934087b53c9f07@192.168.10.30
CSeq: 20723 INVITE
User-Agent: Grandstream GXP2020 1.1.6.44
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 238
v=0
o=201 8000 8000 IN IP4 192.168.10.30
s=SIP Call
c=IN IP4 192.168.10.30
t=0 0
m=audio 5050 RTP/AVP 18 0 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
#
U +0.000418 192.168.10.3:5060 -> 192.168.10.30:5064
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK8cb08d7b02da46ac
From: "Opensips-14x" sip:201@192.168.10.3;tag=f89d240daba57386
To: sip:2685249@192.168.10.3;tag=d4e9e39d125187795ad79ae40f9b4f9f.00fb
Call-ID: 52934087b53c9f07@192.168.10.30
CSeq: 20723 INVITE
Proxy-Authenticate: Digest realm="192.168.10.3",
nonce="49b6c5f00000000c81c829543f443c845ee6c9514213aa27"
Server: OpenSIPS (1.4.4-notls (i386/linux))
Content-Length: 0
#
U +0.022859 192.168.10.30:5064 -> 192.168.10.3:5060
ACK sip:2685249@192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK8cb08d7b02da46ac
From: "Opensips-14x" sip:201@192.168.10.3;tag=f89d240daba57386
To: sip:2685249@192.168.10.3;tag=d4e9e39d125187795ad79ae40f9b4f9f.00fb
Contact: sip:201@192.168.10.30:5064;transport=udp
Supported: path
Call-ID: 52934087b53c9f07@192.168.10.30
CSeq: 20723 ACK
User-Agent: Grandstream GXP2020 1.1.6.44
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
#
U +0.018038 192.168.10.30:5064 -> 192.168.10.3:5060
INVITE sip:2685249@192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bKd5d3f9baf951bf4e
From: "Opensips-14x" sip:201@192.168.10.3;tag=f89d240daba57386
To: sip:2685249@192.168.10.3
Contact: sip:201@192.168.10.30:5064;transport=udp
Supported: replaces, timer, path
Proxy-Authorization: Digest username="201", realm="192.168.10.3",
algorithm=MD5, uri="sip:2685249@192.168.10.3",
nonce="49b6c5f00000000c81c829543f443c845ee6c9514213aa27",
response="269dbb2125ca6984dbb8d427a0f6a053"
Call-ID: 52934087b53c9f07@192.168.10.30
CSeq: 20724 INVITE
User-Agent: Grandstream GXP2020 1.1.6.44
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 238
v=0
o=201 8000 8001 IN IP4 192.168.10.30
s=SIP Call
c=IN IP4 192.168.10.30
t=0 0
m=audio 5050 RTP/AVP 18 0 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
#
U +0.001127 192.168.10.3:5060 -> 192.168.10.30:5064
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bKd5d3f9baf951bf4e
From: "Opensips-14x" sip:201@192.168.10.3;tag=f89d240daba57386
To: sip:2685249@192.168.10.3
Call-ID: 52934087b53c9f07@192.168.10.30
CSeq: 20724 INVITE
Server: OpenSIPS (1.4.4-notls (i386/linux))
Content-Length: 0
alguna idea ...
-- 
rickygm

http://gnuforever.homelinux.com

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[Kamailio-Users-ES] ayuda migrando de openser 1.3x a 1.4x opensips