Phibee Network Operation Center a écrit : > Hi > > Now, my Cisco AS5300 sent call to my asterisk, but two problems: > > When i call the phone number, i have: > > [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: > Call from '' to extension '0426000000' rejected because extension not found. > [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: > Call from '' to extension '0426000000' rejected because extension not found. > > (0426000000 = my phone number) > <..> >
I have put a debug: [Kvoip*CLI> <--- SIP read from UDP://192.168.50.125:59124 ---> INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.125:5060 From: <sip:[email protected]>;tag=6950F0-25C7 To: <sip:[email protected]> Date: Wed, 28 Oct 2009 05:16:26 GMT Call-ID: [email protected] Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 3761097657-3266777566-2192416711-2957366127 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: <sip:[email protected]>;party=calling;screen=yes;privacy=off Timestamp: 1256706986 Contact: <sip:[email protected]:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 8642 2741 IN IP4 192.168.50.125 s=SIP Call c=IN IP4 192.168.50.125 t=0 0 m=audio 18726 RTP/AVP 8 101 c=IN IP4 192.168.50.125 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> [Kvoip*CLI> --- (20 headers 11 lines) --- [Kvoip*CLI> Sending to 192.168.50.125 : 5060 (no NAT) [Kvoip*CLI> Using INVITE request as basis request - [email protected] [Kvoip*CLI> No matching peer for '477000000' from '192.168.50.125:59124' [Kvoip*CLI> Found RTP audio format 8 [Kvoip*CLI> Found RTP audio format 101 [Kvoip*CLI> Peer audio RTP is at port 192.168.50.125:18726 [Kvoip*CLI> Found audio description format PCMA for ID 8 [Kvoip*CLI> Found audio description format telephone-event for ID 101 [Kvoip*CLI> Got unsupported a:fmtp in SDP offer [Kvoip*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Kvoip*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Kvoip*CLI> Peer audio RTP is at port 192.168.50.125:18726 [Kvoip*CLI> Looking for 0426000000 in default (domain 192.168.50.130) [Kvoip*CLI> <--- Reliably Transmitting (no NAT) to 192.168.50.125:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.50.125:5060;received=192.168.50.125 From: <sip:[email protected]>;tag=6950F0-25C7 To: <sip:[email protected]>;tag=as25696e60 Call-ID: [email protected] CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 Ok, i see that: 1- Cisco sent the phone number of the caller (477000000) 2- I have a "To: <sip:[email protected]>" 192.168.50.130 = My Asterisk Server 192.168.50.125 = My Cisco AS5300 3- i have a "No matching peer for '477000000' from '192.168.50.125:59124'" why he search a peer with "477000000" ?? bye Jerome _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
