Try throw the following options into your sip.conf peer: port=5060 insecure=invite,port
Phibee Network Operation Center wrote: > 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:0426000...@192.168.50.130:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.50.125:5060 > From: <sip:477000...@192.168.50.125>;tag=6950F0-25C7 > To: <sip:0426000...@192.168.50.130> > Date: Wed, 28 Oct 2009 05:16:26 GMT > Call-ID: e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 > 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:477000...@192.168.50.125>;party=calling;screen=yes;privacy=off > Timestamp: 1256706986 > Contact: <sip:477000...@192.168.50.125:5060> > 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 - > e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 > [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:477000...@192.168.50.125>;tag=6950F0-25C7 > To: <sip:0426000...@192.168.50.130>;tag=as25696e60 > Call-ID: e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 > 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:0426000...@192.168.50.130>" > 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 _______________________________________________ -- 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