i am configure ser: if (method=="INVITE") { if (uri=~"sip:[EMAIL PROTECTED]") { rewritehostport("192.168.0.183:5080"); }; };
an asterisk: sip.conf ; config Xlite [1234] ;context=sip context=from-ser type=friend auth=md5 username=1234 secret=chooseapassword ;fromdomain=sorcier.com.pe ; para prueba de ser -asterisk callerid="First Extension" <1234> host=dynamic canreinvite=no ;disallow=all ;allow=gsm ;allow=ulaw ;allow=alaw ;and conexion the ser to asterisk ; [ser-sip] type=friend ; permitimos llamadas entrantes y salientes. Usar peer si solo es MWI context=ser-asterisk ; este es el contexto que usan las llamadas entrantes ;host=sorcier.com.pe ; Este es tu hostname o IP del servidor SER host=192.168.0.183 fromdomain=sorcier.com.pe ; este es tu SER_DOMAIN (nombre de dominio del SER) ;insecure=very ; Permite que las llamadas que viene del SER pasen a Asterisk insecure=yes ;[EMAIL PROTECTED] ; esto es para listar las cuentas de voicemail ;i am copy the voip-info and the file the extensions.conf ; Configuracion al servidor ser, para llamada de ida [from-ser] exten => _X.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) [ser-asterisk] ; Ignora el dÃgito 0 ;ignorepat => 0 ; conexion a un telefono sip ;exten => _0X.,1,Dial(SIP/${EXTEN:1},90,Ttr) ;exten => _0X.,1,Dial(SIP/${EXTEN},20,Ttr) ;exten => _0X.,1,Dial(SIP/1234,20,Ttr) ;exten => _0X.,1,Dial(SIP/[EMAIL PROTECTED],20,Ttr) exten => _0X.,1,Dial(SIP/${EXTEN}) i am probe diferents combinations, but no work debug with asterisk and view itis: Sip read: INVITE sip:[EMAIL PROTECTED]:5080 SIP/2.0 Record-Route: <sip:192.168.0.183;ftag=78607191;lr=on> Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.0 Via: SIP/2.0/UDP 192.168.0.185:5060;rport=5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857 From: rbolivar <sip:[EMAIL PROTECTED]>;tag=78607191 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 3143 INVITE Max-Forwards: 16 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 299 v=0 o=rbolivar 23151399 23151750 IN IP4 192.168.0.185 s=X-Lite c=IN IP4 192.168.0.185 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 13 headers, 13 lines Using latest request as basis request Sending to 192.168.0.183 : 5060 (non-NAT) Found peer 'ser-sip' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.185:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 1234 in ser-asterisk Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.0 Via: SIP/2.0/UDP 192.168.0.185:5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857 From: rbolivar <sip:[EMAIL PROTECTED]>;tag=78607191 To: <sip:[EMAIL PROTECTED]>;tag=as1ca211c4 Call-ID: [EMAIL PROTECTED] CSeq: 3143 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]:5080> Content-Length: 0 to 192.168.0.183:5060 Sip read: INVITE sip:[EMAIL PROTECTED]:5080 SIP/2.0 Record-Route: <sip:192.168.0.183;ftag=78607191;lr=on> Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1 Via: SIP/2.0/UDP 192.168.0.185:5060;rport=5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857 From: rbolivar <sip:[EMAIL PROTECTED]>;tag=78607191 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 3143 INVITE Max-Forwards: 16 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 299 v=0 o=rbolivar 23151399 23151750 IN IP4 192.168.0.185 s=X-Lite c=IN IP4 192.168.0.185 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 13 headers, 13 lines Ignoring this request Sip read: ACK sip:[EMAIL PROTECTED]:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.0 From: rbolivar <sip:[EMAIL PROTECTED]>;tag=78607191 Call-ID: [EMAIL PROTECTED] To: <sip:[EMAIL PROTECTED]>;tag=as1ca211c4 CSeq: 3143 ACK User-Agent: Sip EXpress router(0.10.99-dev14-tcp (i386/linux)) Content-Length: 0 8 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Sip read: INVITE sip:[EMAIL PROTECTED]:5080 SIP/2.0 Record-Route: <sip:192.168.0.183;ftag=78607191;lr=on> Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1 Via: SIP/2.0/UDP 192.168.0.185:5060;rport=5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857 From: rbolivar <sip:[EMAIL PROTECTED]>;tag=78607191 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 3143 INVITE Max-Forwards: 16 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 299 v=0 o=rbolivar 23151399 23151750 IN IP4 192.168.0.185 s=X-Lite c=IN IP4 192.168.0.185 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 13 headers, 13 lines Using latest request as basis request Sending to 192.168.0.183 : 5060 (non-NAT) Found peer 'ser-sip' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.185:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 1234 in ser-asterisk Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1 Via: SIP/2.0/UDP 192.168.0.185:5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857 From: rbolivar <sip:[EMAIL PROTECTED]>;tag=78607191 To: <sip:[EMAIL PROTECTED]>;tag=as5a7c3a50 Call-ID: [EMAIL PROTECTED] CSeq: 3143 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]:5080> Content-Length: 0 to 192.168.0.183:5060 Sip read: ACK sip:[EMAIL PROTECTED]:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1 From: rbolivar <sip:[EMAIL PROTECTED]>;tag=78607191 Call-ID: [EMAIL PROTECTED] To: <sip:[EMAIL PROTECTED]>;tag=as5a7c3a50 CSeq: 3143 ACK User-Agent: Sip EXpress router(0.10.99-dev14-tcp (i386/linux)) Content-Length: 0 8 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Sip read: 0 headers, 0 lines the client sip (SER) call to client sip (asterisk) and return error 404 WHAT IS THE PROBLEM??? OR HOW TO FIX THE ERROR?? _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users