Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
just a quick observation, but not sure that it is critical in this case, the first invite comes without Authorization header, then gets challenged then resends the invite (with increased cseq) with calculated response based on the challenge from the server. In your AAstra case, the first invite already contained Authorization header (which is really impossible because you don't have all the pieces to calculate the response). Normally not an issue, as UAS should challenge it, but I wonder why it does it anyway. I would compare Authorize elements between 2 cases particularly response, uri and authorization user name. if response is the same between the two, I am lost. On Nov 24, 2011, at 2:11 PM, Jonas Kellens wrote: > On 11/22/2011 06:13 PM, Alex Vishnev wrote: >> >> it is strange that Aastra acks 401, sends another invite but does not >> increase CSeq. Is that the same behavior with others? >> On Nov 22, 2011, at 11:51 AM, Jonas Kellens wrote: > This is a trace taken when an Alcatel-Lucent PBX sends an INVITE (no refusal > by Asterisk). Do you see any difference ? > > A1.A1.A1.A1 = IP-address Asterisk PBX > AL.AL.AL.AL = IP-address Alcatel-Lucent PBX > > > <--- SIP read from UDP:AL.AL.AL.AL:5060 ---> > INVITE sip:311083335533@A1.A1.A1.A1;user=phone SIP/2.0 > Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, SUBSCRIBE, OPTIONS, UPDATE > Supported: replaces, timer, 100rel > User-Agent: OmniPCX Enterprise R9.1 i1.605.21 > Session-Expires: 1800;refresher=uac > Min-SE: 900 > P-Asserted-Identity: "Dan Luc" > To: > From: "Dan Luc" > ;tag=37a49f0486bab42b240be214b2d13153 > Contact: > Call-ID: 2fae0b0266919172cac1e23dc2567cd2@192.168.8.10 > CSeq: 443337258 INVITE > Via: SIP/2.0/UDP > AL.AL.AL.AL:5060;branch=z9hG4bK5dee58e3294e4b7f9fe34c65af7b4cae > Max-Forwards: 70 > Content-Type: application/sdp > Content-Length: 292 > > v=0 > o=OXE 1322045354 1322045354 IN IP4 AL.AL.AL.AL > s=abs > c=IN IP4 AL.AL.AL.AL > t=0 0 > m=audio 34422 RTP/AVP 8 18 97 > a=sendrecv > a=rtpmap:8 PCMA/8000 > a=ptime:20 > a=maxptime:30 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=ptime:20 > a=maxptime:40 > a=rtpmap:97 telephone-event/8000 > > > <--- Reliably Transmitting (NAT) to AL.AL.AL.AL:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > AL.AL.AL.AL:5060;branch=z9hG4bK5dee58e3294e4b7f9fe34c65af7b4cae;received=AL.AL.AL.AL > From: "Dan Luc" > ;tag=37a49f0486bab42b240be214b2d13153 > To: ;tag=as1b6f387a > Call-ID: 2fae0b0266919172cac1e23dc2567cd2@192.168.8.10 > CSeq: 443337258 INVITE > Server: Asterisk PBX 1.6.2.20 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="7684ab1d" > Content-Length: 0 > > > <--- SIP read from UDP:AL.AL.AL.AL:5060 ---> > INVITE sip:311083335533@A1.A1.A1.A1;user=phone SIP/2.0 > Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, SUBSCRIBE, OPTIONS, UPDATE > Supported: replaces, timer, 100rel > User-Agent: OmniPCX Enterprise R9.1 i1.605.21 > Session-Expires: 1800;refresher=uac > Min-SE: 900 > P-Asserted-Identity: "Dan Luc" > To: > From: "Dan Luc" > ;tag=37a49f0486bab42b240be214b2d13153 > Contact: > Call-ID: 2fae0b0266919172cac1e23dc2567cd2@192.168.8.10 > CSeq: 443337259 INVITE > Max-Forwards: 70 > Authorization: Digest > username="SIPPEERusername",realm="domain.tld",nonce="7684ab1d",algorithm=MD5,uri="sip:311083335533@A1.A1.A1.A1;user=phone",response="38bb824b9081bf2eefe9f9677d3eb005" > Via: SIP/2.0/UDP > AL.AL.AL.AL:5060;branch=z9hG4bK52dae2e7816406e20a9c02aa9cb86726 > Content-Type: application/sdp > Content-Length: 292 > > v=0 > o=OXE 1322045354 1322045354 IN IP4 AL.AL.AL.AL > s=abs > c=IN IP4 AL.AL.AL.AL > t=0 0 > m=audio 34422 RTP/AVP 8 18 97 > a=sendrecv > a=rtpmap:8 PCMA/8000 > a=ptime:20 > a=maxptime:30 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=ptime:20 > a=maxptime:40 > a=rtpmap:97 telephone-event/8000 > > > <--- Transmitting (NAT) to AL.AL.AL.AL:5060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > AL.AL.AL.AL:5060;branch=z9hG4bK52dae2e7816406e20a9c02aa9cb86726;received=AL.AL.AL.AL > From: "Dan Luc" > ;tag=37a49f0486bab42b240be214b2d13153 > To: > Call-ID: 2fae0b0266919172cac1e23dc2567cd2@192.168.8.10 > CSeq: 443337259 INVITE > Server: Asterisk PBX 1.6.2.20 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Session-Expires: 1800;refresher=uac > Contact: > Content-Length: 0 > > > Thanks ! > > Jonas. > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and C
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
On 11/22/2011 06:13 PM, Alex Vishnev wrote: it is strange that Aastra acks 401, sends another invite but does not increase CSeq. Is that the same behavior with others? On Nov 22, 2011, at 11:51 AM, Jonas Kellens wrote: This is a trace taken when an Alcatel-Lucent PBX sends an INVITE (no refusal by Asterisk). Do you see any difference ? A1.A1.A1.A1 = IP-address Asterisk PBX AL.AL.AL.AL = IP-address Alcatel-Lucent PBX <--- SIP read from UDP:AL.AL.AL.AL:5060 ---> INVITE sip:311083335533@A1.A1.A1.A1;user=phone SIP/2.0 Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, SUBSCRIBE, OPTIONS, UPDATE Supported: replaces, timer, 100rel User-Agent: OmniPCX Enterprise R9.1 i1.605.21 Session-Expires: 1800;refresher=uac Min-SE: 900 P-Asserted-Identity: "Dan Luc" To: From: "Dan Luc" ;tag=37a49f0486bab42b240be214b2d13153 Contact: Call-ID: 2fae0b0266919172cac1e23dc2567cd2@192.168.8.10 CSeq: 443337258 INVITE Via: SIP/2.0/UDP AL.AL.AL.AL:5060;branch=z9hG4bK5dee58e3294e4b7f9fe34c65af7b4cae Max-Forwards: 70 Content-Type: application/sdp Content-Length: 292 v=0 o=OXE 1322045354 1322045354 IN IP4 AL.AL.AL.AL s=abs c=IN IP4 AL.AL.AL.AL t=0 0 m=audio 34422 RTP/AVP 8 18 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=maxptime:40 a=rtpmap:97 telephone-event/8000 <--- Reliably Transmitting (NAT) to AL.AL.AL.AL:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP AL.AL.AL.AL:5060;branch=z9hG4bK5dee58e3294e4b7f9fe34c65af7b4cae;received=AL.AL.AL.AL From: "Dan Luc" ;tag=37a49f0486bab42b240be214b2d13153 To: ;tag=as1b6f387a Call-ID: 2fae0b0266919172cac1e23dc2567cd2@192.168.8.10 CSeq: 443337258 INVITE Server: Asterisk PBX 1.6.2.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="7684ab1d" Content-Length: 0 <--- SIP read from UDP:AL.AL.AL.AL:5060 ---> INVITE sip:311083335533@A1.A1.A1.A1;user=phone SIP/2.0 Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, SUBSCRIBE, OPTIONS, UPDATE Supported: replaces, timer, 100rel User-Agent: OmniPCX Enterprise R9.1 i1.605.21 Session-Expires: 1800;refresher=uac Min-SE: 900 P-Asserted-Identity: "Dan Luc" To: From: "Dan Luc" ;tag=37a49f0486bab42b240be214b2d13153 Contact: Call-ID: 2fae0b0266919172cac1e23dc2567cd2@192.168.8.10 CSeq: 443337259 INVITE Max-Forwards: 70 Authorization: Digest username="SIPPEERusername",realm="domain.tld",nonce="7684ab1d",algorithm=MD5,uri="sip:311083335533@A1.A1.A1.A1;user=phone",response="38bb824b9081bf2eefe9f9677d3eb005" Via: SIP/2.0/UDP AL.AL.AL.AL:5060;branch=z9hG4bK52dae2e7816406e20a9c02aa9cb86726 Content-Type: application/sdp Content-Length: 292 v=0 o=OXE 1322045354 1322045354 IN IP4 AL.AL.AL.AL s=abs c=IN IP4 AL.AL.AL.AL t=0 0 m=audio 34422 RTP/AVP 8 18 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=maxptime:40 a=rtpmap:97 telephone-event/8000 <--- Transmitting (NAT) to AL.AL.AL.AL:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP AL.AL.AL.AL:5060;branch=z9hG4bK52dae2e7816406e20a9c02aa9cb86726;received=AL.AL.AL.AL From: "Dan Luc" ;tag=37a49f0486bab42b240be214b2d13153 To: Call-ID: 2fae0b0266919172cac1e23dc2567cd2@192.168.8.10 CSeq: 443337259 INVITE Server: Asterisk PBX 1.6.2.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: Content-Length: 0 Thanks ! Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
Jonas, I did see the traces and I agree they accurately show what you described. What traces never do is say WHY it's happening. What I suggest is that you go into the cli as root with: asterisk -r issue the following commands: core set debug 9 core set verbose 5 quit and then monitor the logs found in /var/log/asterisk (that's where they are usually anyway) you will most likely find the cause in either the debug, full or messages file, again assuming you have those files enabled. If you don't have them enabled, please contact me, either directly or via the list and I'll help you get them enabled and assist you in interpreting them. I suggest that you NOT post them to the list as they could be quite large and a risk of accidentally revealing something you might not want to reveal. On 11/22/2011 08:51 AM, Jonas Kellens wrote: > On 11/22/2011 05:42 PM, Alex Vishnev wrote: >> I doubt it. Unknown headers should be ignored by UAS. is it possible to post >> the trace? >> On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote: > > What trace do you need ? Have you read my original post ? Asterisk SIP debug > trace is posted in my original post. > > > Kind regards, > Jonas. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
it is strange that Aastra acks 401, sends another invite but does not increase CSeq. Is that the same behavior with others? On Nov 22, 2011, at 11:51 AM, Jonas Kellens wrote: > On 11/22/2011 05:42 PM, Alex Vishnev wrote: >> I doubt it. Unknown headers should be ignored by UAS. is it possible to post >> the trace? >> On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote: > > What trace do you need ? Have you read my original post ? Asterisk SIP debug > trace is posted in my original post. > > > Kind regards, > Jonas. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
On 11/22/2011 05:42 PM, Alex Vishnev wrote: I doubt it. Unknown headers should be ignored by UAS. is it possible to post the trace? On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote: What trace do you need ? Have you read my original post ? Asterisk SIP debug trace is posted in my original post. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
I doubt it. Unknown headers should be ignored by UAS. is it possible to post the trace? On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote: > On 11/22/2011 05:31 PM, Alex Vishnev wrote: >> >> Your registration should have also have the flow >> >> PEER ASTERISK >> REGISTER---> >> <--401 >> REGISTER(nonce) -> >> <200OK >> >> Then the server controls the life of the registration and 200 Expires Header >> gives you this timeout. If the invite is sent within that window, then >> Asterisk should not challenge anymore. If Invite is challenged and the peer >> responds with the correctly calculated NONCE, domain and other Auth params, >> then something is wrong with your Authentication. I suggest trapping the >> traffic with Ethereal or any other packet capture programs and examining >> that carefully from the start of the session (i.e. register) to the invite. >> I would also check where the 401 is coming from (i.e. IP address). >> >> Hope that helps >> >> Alex > > > I've already captured with Wireshark, but what to do with it if I don't know > what I'm looking for ?? > > Registration goes without problem, but every INVITE is answered with a > 401-Unauthorized. > > Like I already said : there is no problem with Avaya, Panasonic and > Alcatel-Lucent. > The only difference I see between an INVITE from Avaya and the INVITE from > Aastra PBX is the presence of the SIP-header : "P-Behind-Gsi: 192.168.6.1". > > Could this header mess up Asterisk ? > > Jonas. > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
On 11/22/2011 05:31 PM, Alex Vishnev wrote: Your registration should have also have the flow PEER ASTERISK REGISTER---> <--401 REGISTER(nonce) -> <200OK Then the server controls the life of the registration and 200 Expires Header gives you this timeout. If the invite is sent within that window, then Asterisk should not challenge anymore. If Invite is challenged and the peer responds with the correctly calculated NONCE, domain and other Auth params, then something is wrong with your Authentication. I suggest trapping the traffic with Ethereal or any other packet capture programs and examining that carefully from the start of the session (i.e. register) to the invite. I would also check where the 401 is coming from (i.e. IP address). Hope that helps Alex I've already captured with Wireshark, but what to do with it if I don't know what I'm looking for ?? Registration goes without problem, but every INVITE is answered with a 401-Unauthorized. Like I already said : there is no problem with Avaya, Panasonic and Alcatel-Lucent. The only difference I see between an INVITE from Avaya and the INVITE from Aastra PBX is the presence of the SIP-header : "P-Behind-Gsi: 192.168.6.1". Could this header mess up Asterisk ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
Your registration should have also have the flow PEER ASTERISK REGISTER---> <--401 REGISTER(nonce) -> <200OK Then the server controls the life of the registration and 200 Expires Header gives you this timeout. If the invite is sent within that window, then Asterisk should not challenge anymore. If Invite is challenged and the peer responds with the correctly calculated NONCE, domain and other Auth params, then something is wrong with your Authentication. I suggest trapping the traffic with Ethereal or any other packet capture programs and examining that carefully from the start of the session (i.e. register) to the invite. I would also check where the 401 is coming from (i.e. IP address). Hope that helps Alex On Nov 22, 2011, at 11:23 AM, Jonas Kellens wrote: > On 11/22/2011 04:37 PM, Bruce Ferrell wrote: >> >> >> >> On 11/22/2011 07:29 AM, Jonas Kellens wrote: >>> >>> On 11/22/2011 04:25 PM, Bruce Ferrell wrote: Jonas, May I suggest that you present us your sip.conf entry for this peer, properly redacted, of course. That might help more. What I do for "gateways" at known addresses is to put an entry like this into the sip.conf entry: [peer] type=peer defaultip=192.168.40.123 insecure=invite,port context=some_context >>> >>> >>> This is the peer definition in sip.conf : >>> >>> [SIPPEERusername] >>> type=friend >>> host=dynamic >>> defaultuser=SIPPEERusername >>> secret=guessthis >>> context=from-PEERTRUNK >>> nat=yes >>> dtmfmode=rfc2833 >>> canreinvite=no >>> disallow=all >>> allow=alaw >>> allow=gsm >>> >>> >>> Hope you can help me out with this extra information. >>> >>> >>> Kind regards, >>> >>> Jonas. >> From what I see in your entry, you are requiring registration from the peer. >> The next thing i would check is to see if the registration has succeeded. >> If it doesn't succeed, you will see the results you presented. I see you >> have the peer set as a dynamic host, and if the IP address of the device >> does in fact change then registration is appropriate. > > Registration of the SIP PEER is no problem. The PEER registers with a correct > REGISTER statement and Asterisk sends a 200 OK. > > So the PEER is registered and then wants to make a call (INVITE) but for some > reason this INVITE is being refused with 401-Unauthorized. > > The first 401-Unauthorized is normal, because the SIP PEER needs to send a > second INVITE with a challenge (nonce). But after this INVITE with challenge, > Asterisk still sends a 401 and that's strange !! > > Jonas. > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
On 11/22/2011 04:37 PM, Bruce Ferrell wrote: On 11/22/2011 07:29 AM, Jonas Kellens wrote: On 11/22/2011 04:25 PM, Bruce Ferrell wrote: Jonas, May I suggest that you present us your sip.conf entry for this peer, properly redacted, of course. That might help more. What I do for "gateways" at known addresses is to put an entry like this into the sip.conf entry: [peer] type=peer defaultip=192.168.40.123 insecure=invite,port context=some_context This is the peer definition in sip.conf : [SIPPEERusername] type=friend host=dynamic defaultuser=SIPPEERusername secret=guessthis context=from-PEERTRUNK nat=yes dtmfmode=rfc2833 canreinvite=no disallow=all allow=alaw allow=gsm Hope you can help me out with this extra information. Kind regards, Jonas. From what I see in your entry, you are requiring registration from the peer. The next thing i would check is to see if the registration has succeeded. If it doesn't succeed, you will see the results you presented. I see you have the peer set as a dynamic host, and if the IP address of the device does in fact change then registration is appropriate. Registration of the SIP PEER is no problem. The PEER registers with a correct REGISTER statement and Asterisk sends a 200 OK. So the PEER is registered and then wants to make a call (INVITE) but for some reason this INVITE is being refused with 401-Unauthorized. The first 401-Unauthorized is normal, because the SIP PEER needs to send a second INVITE with a challenge (nonce). But after this INVITE with challenge, Asterisk still sends a 401 and that's strange !! Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
On 11/22/2011 07:29 AM, Jonas Kellens wrote: > On 11/22/2011 04:25 PM, Bruce Ferrell wrote: >> >> Jonas, >> >> May I suggest that you present us your sip.conf entry for this peer, >> properly redacted, of course. That might help more. What I do for >> "gateways" at known addresses is to put >> an entry like this into the sip.conf entry: >> >> >> [peer] >> type=peer >> defaultip=192.168.40.123 >> insecure=invite,port >> context=some_context >> > > > This is the peer definition in sip.conf : > > [SIPPEERusername] > type=friend > host=dynamic > defaultuser=SIPPEERusername > secret=guessthis > context=from-PEERTRUNK > nat=yes > dtmfmode=rfc2833 > canreinvite=no > disallow=all > allow=alaw > allow=gsm > > > Hope you can help me out with this extra information. > > > Kind regards, > > Jonas. >From what I see in your entry, you are requiring registration from the peer. >The next thing i would check is to see if the registration has succeeded. If >it doesn't succeed, you will see the results you presented. I see you have the peer set as a dynamic host, and if the IP address of the device does in fact change then registration is appropriate. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
do you see the register messages? if your device is not registered, INVITE would be challenged. You should check to see if register messages are being properly acknowledge with 200OK. On Nov 22, 2011, at 10:29 AM, Jonas Kellens wrote: > On 11/22/2011 04:25 PM, Bruce Ferrell wrote: >> >> >> Jonas, >> >> May I suggest that you present us your sip.conf entry for this peer, >> properly redacted, of course. That might help more. What I do for >> "gateways" at known addresses is to put an entry like this into the sip.conf >> entry: >> >> >> [peer] >> type=peer >> defaultip=192.168.40.123 >> insecure=invite,port >> context=some_context >> > > > This is the peer definition in sip.conf : > > [SIPPEERusername] > type=friend > host=dynamic > defaultuser=SIPPEERusername > secret=guessthis > context=from-PEERTRUNK > nat=yes > dtmfmode=rfc2833 > canreinvite=no > disallow=all > allow=alaw > allow=gsm > > > Hope you can help me out with this extra information. > > > Kind regards, > > Jonas. > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
On 11/22/2011 04:25 PM, Bruce Ferrell wrote: Jonas, May I suggest that you present us your sip.conf entry for this peer, properly redacted, of course. That might help more. What I do for "gateways" at known addresses is to put an entry like this into the sip.conf entry: [peer] type=peer defaultip=192.168.40.123 insecure=invite,port context=some_context This is the peer definition in sip.conf : [SIPPEERusername] type=friend host=dynamic defaultuser=SIPPEERusername secret=guessthis context=from-PEERTRUNK nat=yes dtmfmode=rfc2833 canreinvite=no disallow=all allow=alaw allow=gsm Hope you can help me out with this extra information. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why
Jonas, May I suggest that you present us your sip.conf entry for this peer, properly redacted, of course. That might help more. What I do for "gateways" at known addresses is to put an entry like this into the sip.conf entry: [peer] type=peer defaultip=192.168.40.123 insecure=invite,port context=some_context On 11/22/2011 06:40 AM, Jonas Kellens wrote: > Hello list, > > this is the communication between an Aastra 5000 PBX and Asterisk, where the > Aastra makes a call to Asterisk. For some reason, Asterisk responds with > 401-Unauthorized and I don't > know why. > > Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT with > this Aastra. > > > A1.A1.A1.A1 = IP-address Asterisk PBX > AS.AS.AS.AS = IP-address Aastra PBX > > Aastra PBX makes a call to the number 3221112233... > > This is all the sip debug trace gathered with asterisk : > > > <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> > INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0 > Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport > From: ;tag=310158BD > To: > Call-ID: 0201CEFEA742 > CSeq: 1 INVITE > Contact: > Proxy-Authorization: Digest username="SIPPEERusername", realm="domain.tld", > nonce="67105ac4", uri="sip:3221112233@A1.A1.A1.A1:5060", response="60be856773 > f86450fc9ddbaf7a568505", algorithm=MD5 > Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE > Max-Forwards: 70 > Privacy: none > P-Asserted-Identity: > User-Agent: A5000 R52-H2C0205 > P-Behind-Gsi: 192.168.6.1 > Content-Type: application/sdp > Content-Length:195 > > v=0 > o=- 0 0 IN IP4 sip.domain.tld > s=- > i=(o=IN IP4 10.1.2.35) > c=IN IP4 AS.AS.AS.AS > t=0 0 > m=audio 62654 RTP/AVP 8 0 > a=rtcp:65115 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > > <-> > > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers > 11 lines) --- > [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP > RTP TOS bits 184 > [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP > RTP CoS mark 5 > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to > AS.AS.AS.AS : 61490 (no NAT) > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE > request as basis request - 0201CEFEA742 > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer > 'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490 > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] > <--- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;received=AS.AS.AS.AS;rport=61490 > From: ;tag=310158BD > To: ;tag=as68f71fe5 > Call-ID: 0201CEFEA742 > CSeq: 1 INVITE > Server: Asterisk PBX 1.6.2.20 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="46ef24d9" > Content-Length: 0 > > <> > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling > destruction of SIP dialog '0201CEFEA742' in 32000 ms (Method: INVITE) > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] > <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> > ACK sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0 > Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport > From: ;tag=310158BD > To: ;tag=as68f71fe5 > Call-ID: 0201CEFEA742 > CSeq: 1 ACK > Contact: > Max-Forwards: 70 > User-Agent: A5000 R52-H2C0205 > P-Behind-Gsi: 192.168.6.1 > Content-Length: 0 > > > <-> > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (11 headers > 0 lines) --- > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] > <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> > INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0 > Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;rport > From: ;tag=33015DBD > To: > Call-ID: 0201CCFEA242 > CSeq: 1 INVITE > Contact: > Proxy-Authorization: Digest username="SIPPEERusername", realm="domain.tld", > nonce="46ef24d9", uri="sip:3221112233@A1.A1.A1.A1:5060", > response="14ecbfc7df24b49926151284c123ea11", > algorithm=MD5 > Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE > Max-Forwards: 70 > Privacy: none > P-Asserted-Identity: > User-Agent: A5000 R52-H2C0205 > P-Behind-Gsi: 192.168.6.1 > Content-Type: application/sdp > Content-Length:195 > > v=0 > o=- 0 0 IN IP4 sip.domain.tld > s=- > i=(o=IN IP4 10.1.2.35) > c=IN IP4 AS.AS.AS.AS > t=0 0 > m=audio 62654 RTP/AVP 8 0 > a=rtcp:65115 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > > > <-> > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers > 11 lines) --- > [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP > RTP TOS bits 184 > [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15
[asterisk-users] Asterisk refuses INVITE (401) and I don't know why
Hello list, this is the communication between an Aastra 5000 PBX and Asterisk, where the Aastra makes a call to Asterisk. For some reason, Asterisk responds with 401-Unauthorized and I don't know why. Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT with this Aastra. A1.A1.A1.A1 = IP-address Asterisk PBX AS.AS.AS.AS = IP-address Aastra PBX Aastra PBX makes a call to the number 3221112233... This is all the sip debug trace gathered with asterisk : <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0 Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport From: ;tag=310158BD To: Call-ID: 0201CEFEA742 CSeq: 1 INVITE Contact: Proxy-Authorization: Digest username="SIPPEERusername", realm="domain.tld", nonce="67105ac4", uri="sip:3221112233@A1.A1.A1.A1:5060", response="60be856773 f86450fc9ddbaf7a568505", algorithm=MD5 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE Max-Forwards: 70 Privacy: none P-Asserted-Identity: User-Agent: A5000 R52-H2C0205 P-Behind-Gsi: 192.168.6.1 Content-Type: application/sdp Content-Length:195 v=0 o=- 0 0 IN IP4 sip.domain.tld s=- i=(o=IN IP4 10.1.2.35) c=IN IP4 AS.AS.AS.AS t=0 0 m=audio 62654 RTP/AVP 8 0 a=rtcp:65115 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 <-> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers 11 lines) --- [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP TOS bits 184 [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP CoS mark 5 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to AS.AS.AS.AS : 61490 (no NAT) [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE request as basis request - 0201CEFEA742 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer 'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] <--- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;received=AS.AS.AS.AS;rport=61490 From: ;tag=310158BD To: ;tag=as68f71fe5 Call-ID: 0201CEFEA742 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="46ef24d9" Content-Length: 0 <> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling destruction of SIP dialog '0201CEFEA742' in 32000 ms (Method: INVITE) [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> ACK sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0 Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport From: ;tag=310158BD To: ;tag=as68f71fe5 Call-ID: 0201CEFEA742 CSeq: 1 ACK Contact: Max-Forwards: 70 User-Agent: A5000 R52-H2C0205 P-Behind-Gsi: 192.168.6.1 Content-Length: 0 <-> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (11 headers 0 lines) --- [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0 Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;rport From: ;tag=33015DBD To: Call-ID: 0201CCFEA242 CSeq: 1 INVITE Contact: Proxy-Authorization: Digest username="SIPPEERusername", realm="domain.tld", nonce="46ef24d9", uri="sip:3221112233@A1.A1.A1.A1:5060", response="14ecbfc7df24b49926151284c123ea11", algorithm=MD5 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE Max-Forwards: 70 Privacy: none P-Asserted-Identity: User-Agent: A5000 R52-H2C0205 P-Behind-Gsi: 192.168.6.1 Content-Type: application/sdp Content-Length:195 v=0 o=- 0 0 IN IP4 sip.domain.tld s=- i=(o=IN IP4 10.1.2.35) c=IN IP4 AS.AS.AS.AS t=0 0 m=audio 62654 RTP/AVP 8 0 a=rtcp:65115 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 <-> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers 11 lines) --- [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP TOS bits 184 [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP CoS mark 5 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to AS.AS.AS.AS : 61490 (no NAT) [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE request as basis request - 0201CCFEA242 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer 'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] <--- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;received
