Gracias por su pronta respuesta, sin embargo sigo obteniendo el mismo error, ejecutando el enrutamiento de ambas formas. Segun veo las 3 maneras de enviar la llamada ya las probé. El problema me parece es que no esta cambiando el transporte a UDP. (Cuando cambio en eyebeam el transporte a UDP todo funciona perfecto)
1) t_relay("udp:192.168.1.107:5060");
2)$rd="192.168.1.107"; $rp=5060; t_relay();
3)rewritehostport("192.168.1.107:5060");
*** Adjunto la captura con ngrep en el puerto 5070 (el 5060 no muestra nada) ***
interface: any filter: (ip or ip6) and ( port 5070 ) # T 2009/02/04 09:24:51.008047 192.168.1.112:4398 -> 192.168.1.80:5070 [AP] INVITE sip:5000@192.168.1.80:5070;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.112:44714;branch=z9hG4bK-d87543-9d3c566fdc403439-1--d87543-;rport
Max-Forwards: 70 Contact: sip:1000@192.168.1.112:4398;transport=TCP To: "5000"sip:5000@192.168.1.80:5070 From: "Robert"sip:1000@192.168.1.80:5070;tag=ab66e77c Call-ID: 2430a230b532db7fOWYxNjU4YjAwOGI2OGU4YzZjODJlM2I3ZmNlM2ViOTQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 600
v=0 o=- 5 2 IN IP4 192.168.1.112 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.112 t=0 0 m=audio 57656 RTP/AVP 107 100 106 6 0 105 18 3 5 101 a=alt:1 4 : aPtl3C/D 3W5T7M+K 192.168.1.112 57656 a=alt:2 3 : eVWq5uTY IpZ8M+i0 192.168.109.225 57656 a=alt:3 2 : KPku7YVD OZ4Rop+9 192.168.28.1 57656 a=alt:4 1 : 5snEu2kb 0xPPko0L 192.168.112.1 57656 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:F33753243AF744D497790C5079BD54E5
# T 2009/02/04 09:24:51.008479 192.168.1.80:5070 -> 192.168.1.112:4398 [AP] SIP/2.0 100 Giving a try Via: SIP/2.0/TCP 192.168.1.112:44714;branch=z9hG4bK-d87543-9d3c566fdc403439-1--d87543-;rport=4398
To: "5000"sip:5000@192.168.1.80:5070 From: "Robert"sip:1000@192.168.1.80:5070;tag=ab66e77c Call-ID: 2430a230b532db7fOWYxNjU4YjAwOGI2OGU4YzZjODJlM2I3ZmNlM2ViOTQ. CSeq: 1 INVITE Server: Kamailio (1.4.3-notls (i386/linux)) Content-Length: 0
# T 2009/02/04 09:24:51.008762 192.168.1.80:5070 -> 192.168.1.112:4398 [AP] SIP/2.0 477 Send failed (477/TM) Via: SIP/2.0/TCP 192.168.1.112:44714;branch=z9hG4bK-d87543-9d3c566fdc403439-1--d87543-;rport=4398
To: "5000"sip:5000@192.168.1.80:5070;tag=23de66d6b9a34b2316f98bd4265c2e08-425c
From: "Robert"sip:1000@192.168.1.80:5070;tag=ab66e77c Call-ID: 2430a230b532db7fOWYxNjU4YjAwOGI2OGU4YzZjODJlM2I3ZmNlM2ViOTQ. CSeq: 1 INVITE Server: Kamailio (1.4.3-notls (i386/linux)) Content-Length: 0
# T 2009/02/04 09:24:51.008954 192.168.1.112:4398 -> 192.168.1.80:5070 [A]
# T 2009/02/04 09:24:51.113334 192.168.1.112:4398 -> 192.168.1.80:5070 [AP] ACK sip:5000@192.168.1.80:5070;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.112:44714;branch=z9hG4bK-d87543-9d3c566fdc403439-1--d87543-;rport
To: "5000"sip:5000@192.168.1.80:5070;tag=23de66d6b9a34b2316f98bd4265c2e08-425c
From: "Robert"sip:1000@192.168.1.80:5070;tag=ab66e77c Call-ID: 2430a230b532db7fOWYxNjU4YjAwOGI2OGU4YzZjODJlM2I3ZmNlM2ViOTQ. CSeq: 1 ACK Content-Length: 0
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2009/2/3 Iñaki Baz Castillo ibc@aliax.net
El Miércoles, 4 de Febrero de 2009, Robert Contreras escribió:
if (is_method("INVITE")) { setflag(1); # do accounting if(uri=~"sip:5000@192.168.1.80:5070") #Para llamar a la extension 5000 en asterisk { t_relay("udp:192.168.1.107:5060"); #route(10); } }
¿Puedes usar lo siguiente?:
$rd = "192.168.1.107"; $rp = 5060; t_relay();
También sería óptimo una captura con ngrep.
-- Iñaki Baz Castillo
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