search for: cnonce

...ting to setup Asterisk with sipcall.co.uk. They use Intertex kit to provide the SIP service. Unfortunately Asterisk cannot seem to authenticate against Intertex. Having provided SIP debug info the provider has informed me that Asterisk does not appear to support 'qop', 'nc' and 'cnonce' which are used to stop replay attacks. So, does Asterisk support 'qop', 'nc' and 'cnonce'? If not, is anyone working on support? Before anyone says code it yourself, I would if I could! Thanks, Nathan.

...5060;transport=udp> Authorization: Digest username="111",realm="asterisk",uri="sip:*777 at 192.168.1.4;user=phone",response="8b90970d8fc724893e876263ce8c2cd3",nonce="1424929962/9af5af19e633c82d2a9e17ec97afb72b",opaque="2776507e426bda2b",cnonce="945bf4a1",qop=auth,nc=00000001,algorithm=md5 Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Allow-Events: kpml,dialog Content-Length: 322 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=...

...D11dGYtOCxjbm9uY2U9IjFlOTZkYmZiZWEwZjUxNmVhZjEyYmM0NjU1M2JmZjVlIixkaWdlc3QtdXJpPSJwb3AzL25hbm8iLG5jPTAwMDAwMDAxLG5vbmNlPSJPVGpGWmhjS2FIVjZSVGMyZlRDTXJ3PT0iLHFvcD1hdXRoLHJlYWxtPSIiLHJlc3BvbnNlPTVmMTQyZWVlN2FmMWVmYTJhYWI5ZmM0ODNiOGJjOTJhLHVzZXJuYW1lPSJvdHJzIg== authzid="otrs",charset=utf-8,cnonce="1e96dbfbea0f516eaf12bc46553bff5e", digest-uri="pop3/nano",nc=00000001,nonce="OTjFZhcKaHV6RTc2fTCMrw==", qop=auth,realm="",response=5f142eee7af1efa2aab9fc483b8bc92a,username="otrs" Client is a Perl script using Net::POP3 and Authen::SASL Modules D...

..."?> <Container> <id>aQlfVHX+qPM</id> <lifetime>2009-09-19T08:14:55Z</lifetime> </Container> The response should contain the next headers: Content-Type=`application/vnd.3gpp+xml` Authentication-Info={rspauth=`xxxxxxxxxxx...xxxxxxxxxxx`,qop=`auth-int`,cnonce=`xxxxxxxxxxxxxxxxxx....xxxxxxxxxxxxx`,nc=`000 00001`} How can I do that? I always see {"Cache-Control"=>"no-cache", "Content-Type"=>"application/xml"} when I print the headers... I can use xml builder ok to generate the xml, but I have the problem wi...

...risk PBX 1.8.1.1 Authorization: Digest username="<phone number>", realm="BroadWorks", algorithm=MD5, uri="sip:0871271201 at sip.internode.on.net", nonce="BroadWorksXgjz10gvqTmcdnmtBW", response="829397204c8dd43c8fc3435baa253075", qop=auth, cnonce="03a77924", nc=00000001 Date: Thu, 10 Feb 2011 02:04:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 738 v=0 o=root 51098296 51098297 IN IP4 <my static ip> s=Asterisk...

Oh god it works ! to switch cisco to upd I used config: <transportLayerProtocol>2</transportLayerProtocol> with udp it works well, thanks for your help :) > On 24 Feb 2015, at 17:02, Joshua Colp <jcolp at digium.com> wrote: > > If you use UDP with force_rport=no it'll work. > If you use TCP then set rewrite_contact=yes so it'll reuse the established TCP

...ent: MailVision Sip Phone 1.0 Expires: 3600 Authorization: Digest username="912345678",realm="multico.es", nonce="474c3ba3aca1b4a444188e1adbe5b2da", response="20632e97a55c06337981cb4750b88ef8",uri="sip:1.2.3.4", algorithm=MD5,qop=auth,nc=00000001,cnonce="c770968528636542776ba318fef0080e" =============================================================== ========== Received ========= I VALIDATE YOU = transport UDP = remote 1.2.3.4:32769 = local 24.132.244.120:5060 ============================= SIP/2.0 200 OK Via: SIP/2.0/UDP 24.132.244.12...

...TIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: Xlite release 1100l stamp 49022 Authorization: Digest username="8887776666",realm="192.168.50.20",nonce="d999e9471b1d",uri="sip:Xlite release 1100l stamp 49022",response="ba26805d2f0b97a70565c37e81444e44",cnonce="820e1f348b49cd73d92e1bc793be5ad7",nc=00000001,qop=auth,algorithm=MD5 Content-Length: 0 REGISTER sip:Xlite release 1100l stamp 49022 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.221:22818;branch=z9hG4bK-d8754z-c605aa61ac248834-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:8887776666 at 33.33....

...;testrealm at host.com\", " - "nonce=\"dcd98b7102dd2f0e8b11d0f600bfb0c093\", " - "uri=\"/dir/index.html\", " - "qop=auth, " - "nc=00000001, " - "cnonce=\"0a4f113b\", " - "response=\"6629fae49393a05397450978507c4ef1\", " - "opaque=\"5ccc069c403ebaf9f0171e9517f40e41\"", + } +}; +static const struct http_auth_credentials_test mufasa[] = { + { .scheme = "Dig...

...uot; #include "base64.h" #include "buffer.h" @@ -29,45 +31,22 @@ /* sent: */ const char *server_first_message; - unsigned char salt[16]; - unsigned char salted_password[SHA1_RESULTLEN]; + const char *snonce; /* received: */ const char *gs2_cbind_flag; const char *cnonce; - const char *snonce; const char *client_first_message_bare; const char *client_final_message_without_proof; buffer_t *proof; + + /* stored */ + buffer_t *stored_key; + buffer_t *server_key; }; -static void Hi(const unsigned char *str, size_t str_size, - const unsigned char *salt,...

On Mon, Mar 7, 2016 at 2:53 PM, Jean-Denis Girard <jd.girard at sysnux.pf> wrote: > Hi, > > Le 07/03/2016 09:28, George Joseph a ?crit : > > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released. > > I have tried GIT-master-ee5a944M on my Fedora 23 test server, and got: > > [pjproject] Unpacking /tmp/pjproject-2.4.5.tar.bz2 > [pjproject]

http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ -- Dave Topping e: info at dntopping.uk t: 03445 888 888 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170901/ae060564/attachment.html>

I have done additional testing and I haven't been able to figure out why it's failing. Since my original testing we now set the realm on the authentication section to match what we receive from NEC. It's of the format abc at xyz.com I have verified the md5_cred several times and it matches the user:realm:password formula 3016:insiph at something0a646666.com:3016 where username is

I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001 at 192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes.

...zation: Digest username="1004",realm="asterisk",nonce="1485271992/b1bebde5cb4a763ed85b1d8e52c8e30d",uri="sip:333 at xxx.xxx.xxx.xxx",algorithm=MD5,response="8dd827e9910c2446fb0b8497f5944b81",opaque="66e52\ 68a2111e777",qop=auth,nc=00000001,cnonce="9dda9e0d"^M Contact: "1004" <sip:1004 at yyy.yyy.yyy.yyy:5063>^M Expires: 240^M User-Agent: Cisco/SPA504G-7.4.8a^M Content-Length: 401^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE^M Supported: replaces^M Content-Type: application/sdp^M ^M v=0^M...

Hi! I have a new problem with my SIP device.I have done some changes and now I receive continuosly a SIP message: "501" "Not impelmented" back from the SIP Gateway. I can see that it doesn't register to Asterisk. I have in the SIP device: Registrar 1: UnRegistered to: 2222 registrar: 188.208.12.237 5060 expires: 2000 name: gateway passwd: 123 My

...ample.com> Call-ID: 54ddf28f87c7-ak5cx3jtjc8c CSeq: 12480 PUBLISH Max-Forwards: 70 Event: presence Authorization: Digest username="1001",realm="newsip",nonce="1423831696/f03cd493c485d56261cbcc0648f97e54",uri=" sip:1001 at example.com ",qop=auth,nc=00000001,cnonce="6e4402d8",response="34dcaab743eb5f0837529fbb54105f5a",opaque="20c8892f17d503b0",algorithm=MD5 Expires: 3600 Content-Type: application/pidf+xml Content-Length: 480 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:param...

...equipment provider. Here is their answer:- The reason the registration fails is because not all of the required authentication parameters are provided by Linphone. The Linphone does not send nc (nonce-count), qop (quality of protection) and cnonce (client nonce). It is then not fully compatible with the standard specification on the authentication scheme used for SIP (RFC 2617). Some SIP servers (sipcall.co.uk for example) may not accept this as it would make them vulnerable fo...

...rization: Digest username="3073334444", realm="voip.myfavoriteisp.net", algorithm=MD5, uri="sip:voip.myfavoriteisp.net", nonce="27a4f823b14f298bbbbbbbbbbbbbbbb", response="6ccccccccccccccccccccccccccccccc", opaque="", qop="auth", cnonce="abcdefab", nc=00000001 Expires: 120 Contact: <sip:s@192.168.134.247> Event: registration Content-Length: 0 --- phone*CLI> <-- SIP read from 45.123.29.97:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.134.247:5060;received=216.98.159.235;branch=z9hG4bK32fe0b71;rport...

...DED = 1 AUTHENTICATIONMODE = 0 ISTWOSTAGEDIAL = 0 CHANNELSELECTMODE = 2 GWDEBUGLEVEL = 5 ENABLEPROXYKEEPALIVE = 0 ENABLEEARLYMEDIA = 1 ISUSERPHONE = 0 DEFAULTNUMBER = '1000' PROXYNAME = 'myproxy' REGISTRARIP = '' SIPGATEWAYNAME = '192.168.0.244' USERNAME = '' CNONCE = '0a123bcf' PASSWORD = '787899' STATICNATIP = 0.0.0.0 ENABLESILENCEDISCONNECT = 1 RXDTMFOPTION = 3 ISFALLBACKUSED = 0 ALWAYSSENDTOPROXY = 0 ISPROXYHOTSWAP = 0 PROXYKEEPALIVETIME = 60 PROXYREDUNDANCYMODE = 0 PROXYHOTSWAPRTX = 3 SENDINVITETOPROXY = 0 REGISTRATIONRETRYTIME = 30 USEGAT...