Likes: 
Ik gebruik een TrixBox (Asterisk/FreePBX) machine voor het afhandelen van mijn telefoonverkeer. Nu probeer ik echter een inkomend telefoonnummer van VoipBuster aan de praat te krijgen, maar dit wil niet lukken.
Ik gebruik de volgende instellingen in sip.conf
de volgende instellingen in sip_additional.confCode:; Note: If your SIP devices are behind a NAT and your Asterisk ; server isn't, try adding "nat=1" to each peer definition to ; solve translation problems. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw ; If you need to answer unauthenticated calls, you should change this ; next line to 'from-trunk', rather than 'from-sip-external'. ; You'll know this is happening if when you call in you get a message ; saying "The number you have dialed is not in service. Please check the ; number and try again." context = from-pstn ; Send unknown SIP callers to this context callerid = Unknown tos=0x68 ; #, in this configuration file, is NOT A COMMENT. This is exactly ; how it should be. #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf
Als ik nu bel op mijn VoipBuster nummer krijg ik de volgende debug uitvoerCode:register=***username***:***password***@sip1.voipbuster.com/***username*** ***local sip extensions*** [voipbuster] username=***username*** type=friend srvlookup=yes secret=***password*** qualify=yes nat=yes insecure=very host=sip1.voipbuster.com fromuser=***username*** fromdomain=sip1.voipbuster.com dtmfmode=inband disallow=all context=from-pstn allow=alaw allow=ulaw allow=gsm
Waarbij het er naar mijn idee op lijkt dat asterisk de user niet kan vinden in mijn configuratie, en dus dit terug geeft aan voipbuster, en de verbinding dus wordt verbroken.Code:<-- SIP read from 194.120.0.203:5060: INVITE sip:***username***@sip1.voipbuster.com:5060 SIP/2.0 Via: SIP/2.0/UDP 194.120.0.203:5060;branch=z9hG4bK67612ccbaa874e689a8e70ebcf70918a From: "Anonymous"<sip:Anonymous@sip1.voipbuster.com:5060>;tag=cb0078c2448d399f848d To: <sip:***username***@sip1.voipbuster.com:5060> Contact: sip:Anonymous@194.120.0.203:5060 Call-ID: 59c175cda127480babd2a68d05638a33 CSeq: 0 INVITE User-Agent: (Very nice Sip Registrar Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS Content-Type: application/sdp Content-Length: 169 v=0 o=***username*** 1150192228 1150192228 IN IP4 80.239.235.179 s=SIP Call c=IN IP4 80.239.235.179 t=0 0 m=audio 57832 RTP/AVP 8 a=rtpmap:8 pcma/8000 a=ptime:20 --- (11 headers 8 lines)--- Using INVITE request as basis request - 59c175cda127480babd2a68d05638a33 Sending to 194.120.0.203 : 5060 (non-NAT) Found no matching peer or user for '194.120.0.203:5060' Found RTP audio format 8 Peer audio RTP is at port 80.239.235.179:57832 Found description format pcma Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for ***username*** in from-pstn (domain sip1.voipbuster.com) Reliably Transmitting (no NAT) to 194.120.0.203:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 194.120.0.203:5060;branch=z9hG4bK67612ccbaa874e689a8e70ebcf70918a;received=194.120.0.203 From: "Anonymous"<sip:Anonymous@sip1.voipbuster.com:5060>;tag=cb0078c2448d399f848d To: <sip:***username***@sip1.voipbuster.com:5060>;tag=as2cd087a9 Call-ID: 59c175cda127480babd2a68d05638a33 CSeq: 0 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:***username***@192.168.0.15> Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 194.120.0.203:5060: ACK sip:***username***@sip1.voipbuster.com:5060 SIP/2.0 Via: SIP/2.0/UDP 194.120.0.203:5060;branch=z9hG4bK67612ccbaa874e689a8e70ebcf70918a From: "Anonymous"<sip:Anonymous@sip1.voipbuster.com:5060>;tag=cb0078c2448d399f848d To: <sip:***username***@sip1.voipbuster.com:5060>;tag=as2cd087a9 Contact: sip:Anonymous@194.120.0.203:5060 Call-ID: 59c175cda127480babd2a68d05638a33 CSeq: 0 ACK User-Agent: (Very nice Sip Registrar Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS --- (9 headers 0 lines)--- Destroying call '59c175cda127480babd2a68d05638a33' Destroying call '1afd65e3358d3bd71770dd02425434b2@127.0.0.1'
Iemand een idee wat er hier fout gaat. Het heeft eerder wel al eens gewerkt met een oudere Asterisk@Home server. Maar die configuratie heb ik helaas niet meer tot mijn beschikking.
Groeten
Quote
Oorspronkelijk geplaatst door Zork538
