Re: [Asterisk-Users] Problem with SIP 407
G'day Marc, On Wed, 25 Feb 2004, Marc Fargas wrote: Im in trouble with SIP. Ive got a SIP FXS gateway from www.micronet.info (SP5002/S) and traed to register to asterisk, It seems to autentícate but sniffing the net it shows a 407 proxy authen required error message and I cannot make any outgoing calls from that gateway. I captured a flow between * and an ATA-186 the other day, because I had the same problem (well, the symptom was the same). The 407 message from * is part of the registration flow. It tells the client that it needs to resend its REGISTER, this time including a Proxy-Authentication (sp?) header in the request. That header contains the authentication data (authuser, password). I'd suggest getting into the network with ethereal or the like and start sniffing the packet flow. In my case, a hardware incompatibility was preventing my client from receiving the 407 from *, so it never responded to it... (Getting a packet trace will also be essential in getting further support, either from your FXS vendor or the SIP mavens on this list.) Ive tried putting Domain = Asterisk on the FXS and other things, also played with codecs but everything seems to come from the 407 message, how can I avoid that message? Well, you could try removing the password (secret=XXX) from the entry in sip.conf, allowing the client to register without authentication. Might be something to try, but I don't think I'd run live that way... ;-) Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP 407
Vic Cross wrote: G'day Marc, On Wed, 25 Feb 2004, Marc Fargas wrote: Im in trouble with SIP. Ive got a SIP FXS gateway from www.micronet.info (SP5002/S) and traed to register to asterisk, It seems to autentcate but sniffing the net it shows a 407 proxy authen required error message and I cannot make any outgoing calls from that gateway. I captured a flow between * and an ATA-186 the other day, because I had the same problem (well, the symptom was the same). The 407 message from * is part of the registration flow. It tells the client that it needs to resend its REGISTER, this time including a Proxy-Authentication (sp?) header in the request. That header contains the authentication data (authuser, password). Let's clear this up: A SIP ua sends a REGISTER to a location server to tell the server where it can be reached. At registration, the server challenges the UA with a www-authentication. When authenticated, the server stores the IP address and contact header for some time (expiry=) to be able to place calls to the UA. This is a SIP peer in asterisk. The standard sip channels has a bug here and issues a Proxy-authentication. The chan_sip2 channel issues a www-auth. When a SIP UA want to call through asterisk, asterisk want's to know for certain who it is before admitting any services (except default context). To let the SIP ua through, we issue a Proxy-auth. If it succeeds, the asterisk sip user is allowed to reach whatever is reachable in the user's SIP context. A type=friend SIP client is both a user and a peer. Neither form of authentication sends the password in clear. This is nowadays forbidden in SIP. We use digest authentication, a challenge-response mechanism. I'm a bit afraid that Asterisk's authentication in the SIP channels is a bit out of date and that may be your problem. Please forward SIP debug output so we can go through the various stages that leads to the 407. Ive tried putting Domain = Asterisk on the FXS and other things, also played with codecs but everything seems to come from the 407 message, how can I avoid that message? Well, you could try removing the password (secret=XXX) from the entry in sip.conf, allowing the client to register without authentication. Might be something to try, but I don't think I'd run live that way... ;-) If so, add an ACL so you limit the IP addresses that may use this account. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with SIP 407
On this cuts note that the gateway has username 'Republica', you could see some reference to Republica2 which corresponds to a second line on the gateway that I have disabled. Thanks for your help! That's SIP debug when dialling '9' (9 would do Goto(s,1)) === *CLI *CLI 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK0f92815a From: asterisk sip:[EMAIL PROTECTED];tag=as0bc66d50 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 192.168.6.2:5060 Sip read: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.2.2:5060;received=192.168.2.2;branch=z9hG4bK0f92815a From: asterisksip:[EMAIL PROTECTED] ;tag=as0bc66d50 To: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-30d-bef46-6225 Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Content-Length:0 7 headers, 0 lines Feb 25 21:03:04 WARNING[98311]: chan_sip.c:4875 handle_response: Host '192.168.6.2' does not implement 'NOTIFY' Sip read: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.6.2:5060 Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: FXS_GW (2asipfxs.106) From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-30e-befaf-7ba1 To: sip:[EMAIL PROTECTED]:5060;user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Content-Type: application/sdp Content-Length:234 v=0 o=FXS_GW 12367 0 IN IP4 192.168.6.2 s=Audio Session i=Audio Session c=IN IP4 192.168.6.2 t=0 0 m=audio 16384 RTP/AVP 18 4 0 8 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 10 headers, 11 lines Using latest request as basis request Sending to 192.168.6.2 : 5060 (non-NAT) Found audio format UNKN Found audio format ULAW Found audio format UNKN Found audio format ALAW Found description format G729 Found description format G723 Found description format PCMU Found description format PCMA Capabilities: us - 6, them - 269/854015, combined - 6 Non-codec capabilities: us - 1, them - 0, combined - 1 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.6.2:5060 From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-30e-befaf-7ba1 To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as76b77fc8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=2f9d85fe Content-Length: 0 to 192.168.6.2:5060 Sip read: ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.6.2:5060 Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: FXS_GW (2asipfxs.106) From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-30e-befaf-7ba1 To: sip:[EMAIL PROTECTED]:5060;user=phone ;tag=as76b77fc8 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Content-Length:0 9 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.6.2:5060 Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=Republica, realm=asterisk, nonce=2f9d85fe, uri=sip:[EMAIL PROTECTED]:5060;user=phone, re sponse=4b434a0e18166c573b006cf9cbd2f3bc, algorithm=MD5 User-Agent: FXS_GW (2asipfxs.106) From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-30e-befaf-7ba1 To: sip:[EMAIL PROTECTED]:5060;user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Content-Type: application/sdp Content-Length:234 v=0 o=FXS_GW 12367 0 IN IP4 192.168.6.2 s=Audio Session i=Audio Session c=IN IP4 192.168.6.2 t=0 0 m=audio 16384 RTP/AVP 18 4 0 8 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 11 headers, 11 lines Using latest request as basis request Sending to 192.168.6.2 : 5060 (non-NAT) Found audio format UNKN Found audio format ULAW Found audio format UNKN Found audio format ALAW Found description format G729 Found description format G723 Found description format PCMU Found description format PCMA Capabilities: us - 6, them - 269/854015, combined - 6 Non-codec capabilities: us - 1, them - 0, combined - 1 Looking for 9 in nacer list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.6.2:5060 From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-30e-befaf-7ba1 To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as76915db6 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.6.2:5060 -- Executing Goto(SIP/Republica-a2aa, default|s|1) in new stack -- Goto (default,s,1) -- Executing Answer(SIP/Republica-a2aa, ) in new stack We're at 192.168.2.2 port 19466 Video is at 192.168.2.2 port 18490 Answering with preferred capability 4 Answering with preferred capability 2 Answering with
Re: [Asterisk-Users] Problem with SIP 407
Seems like republica registers ok, but not republica2. Republica2 failes to authenticate. You have a normal registration sequense here: -Client sends a REGISTER without authentication -Server sends trying... -Server sends 407 Proxy auth (should be WWW auth) with challenge -Clients ACK -Client sends a new REGISTER with authentication -Server tries auth -If auth fails (propably wrong secret/password) a 401 unauthorized is issued -If auth succeeds a 200 OK is issued Apart from that, the Asterisk server after authentication wants to tell the client that it has voicemail, and the client responds that it has no clue of what the server is trying to say. Take away the mailbox= parameter in sip.conf to avoid this. You client seems to send a lot of REGISTERs without waiting for response. Other than that, check the passwords for republica2. Republica1 should be able to receive calls from asterisk. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with SIP 407
I have commented de Republica2 in sip.conf 'cause if I uncomment it neither Republica and Republica2 register (maybe because they're on the same gateway?) Well, inspite it register well when I try tocall any extension It plays 'busy' tone immediately after Asterisk takes the calls I thought it was a codecs problem but I have another gateway similar with H.323 and hav codecs configured same way both on asterisk and the gateways, the H.323 one goes right but the SIP one can't do anything, it just plays around with 'busy' tones. In my previous post you can see the output of sip debug on Asterisk when trying to call an extension, on the gateway side that's what I get: Line : 1, Start Inviting strDes To:sip:[EMAIL PROTECTED]:5060;user=phone, strOri From:sip:[EMAIL PROTECTED] 1-RvSipCallLegMgrCreateCallLeg() ok! Success to rvSdpMsgEncodeToBuf * -- Message Sent (Message type: 0) (call-leg 58e04c) INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.6.2:5060 Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: FXS_GW (2asipfxs.106) From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-403d1740-5870d2-7301 To: sip:[EMAIL PROTECTED]:5060;user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Content-Type: application/sdp Content-Length:234 v=0 o=FXS_GW 12367 0 IN IP4 192.168.6.2 s=Audio Session i=Audio Session c=IN IP4 192.168.6.2 t=0 0 m=audio 16384 RTP/AVP 18 4 0 8 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 1-RVSIP_CALL_LEG_STATE_INVITING -- Message Received (Message Type: 1) (call-leg 58e04c) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.6.2:5060 User-Agent: Asterisk PBX Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER Contact: sip:[EMAIL PROTECTED] From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-403d1740-5870d2-7301 To: sip:[EMAIL PROTECTED]:5060;user=phone ;tag=as07a0b938 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Content-Length:0 1-RVSIP_CALL_LEG_STATE_TERMINATED 1-Gen_BusyTone ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
