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  • Best Ruby Git library?

    - by Jeff Welling
    Which is the best Git library in Ruby to use? Git, Grit, Rugged, Other? Background: I'm the current maintainer of TicGit-ng which is a distributed offline ticket system built on git, and I've read and heard over and over again that Grit is the one I should use because it supersedes the Git gem, but there seems to be either a lack of documentation or a lack of features because myself and others have failed in trying to switch from the deprecated-but-functional Git to the newer Grit gem.

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  • Paramount Pictures Movies Dynamic Theme for Windows 7

    - by Asian Angel
    Would you like to have all of the latest movie wallpapers from Paramount Pictures delivered straight to your desktop? Then this is the theme you are looking for. This dynamic RSS-fed theme brings you the latest wallpapers from movies such as Thor, True Grit, Kung Fu Panda 2, Super 8, and Transformers 3. Download the Paramount Pictures Movies Dynamic Theme [Windows 7 Personalization Gallery] You can learn more about Microsoft’s dynamic themes here: RSS-fed dynamic themes FAQ How To Easily Access Your Home Network From Anywhere With DDNSHow To Recover After Your Email Password Is CompromisedHow to Clean Your Filthy Keyboard in the Dishwasher (Without Ruining it)

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  • EF4 Import/Lookup thousands of records - my performance stinks!

    - by Dennis Ward
    I'm trying to setup something for a movie store website (using ASP.NET, EF4, SQL Server 2008), and in my scenario, I want to allow a "Member" store to import their catalog of movies stored in a text file containing ActorName, MovieTitle, and CatalogNumber as follows: Actor, Movie, CatalogNumber John Wayne, True Grit, 4577-12 (repeated for each record) This data will be used to lookup an actor and movie, and create a "MemberMovie" record, and my import speed is terrible if I import more than 100 or so records using these tables: Actor Table: Fields = {ID, Name, etc.} Movie Table: Fields = {ID, Title, ActorID, etc.} MemberMovie Table: Fields = {ID, CatalogNumber, MovieID, etc.} My methodology to import data into the MemberMovie table from a text file is as follows (after the file has been uploaded successfully): Create a context. For each line in the file, lookup the artist in the Actor table. For each Movie in the Artist table, lookup the matching title. If a matching Movie is found, add a new MemberMovie record to the context and call ctx.SaveChanges(). The performance of my implementation is terrible. My expectation is that this can be done with thousands of records in a few seconds (after the file has been uploaded), and I've got something that times out the browser. My question is this: What is the best approach for performing bulk lookups/inserts like this? Should I call SaveChanges only once rather than for each newly created MemberMovie? Would it be better to implement this using something like a stored procedure? A snippet of my loop is roughly this (edited for brevity): while ((fline = file.ReadLine()) != null) { string [] token = fline.Split(separator); string Actor = token[0]; string Movie = token[1]; string CatNumber = token[2]; Actor found_actor = ctx.Actors.Where(a => a.Name.Equals(actor)).FirstOrDefault(); if (found_actor == null) continue; Movie found_movie = found_actor.Movies.Where( s => s.Title.Equals(title, StringComparison.CurrentCultureIgnoreCase)).FirstOrDefault(); if (found_movie == null) continue; ctx.MemberMovies.AddObject(new MemberMovie() { MemberProfileID = profile_id, CatalogNumber = CatNumber, Movie = found_movie }); try { ctx.SaveChanges(); } catch { } } Any help is appreciated! Thanks, Dennis

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  • Asterisk SIP digest authentication username mismatch

    - by Matt
    I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue of older 3com phones that come without a SIP image. The 3com phones are communicating SIP with the Asterisk, but are unable to register because they present a digest username value that doesn't match what Asterisk thinks it should. As an example, here are the relevant lines from a successful registration from a soft phone: Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1cac3853" Phone responds: Authorization: Digest username="2321", realm="asterisk", nonce="1cac3853", uri="sip:192.168.254.12", algorithm=md5, response="d32df9ec719817282460e7c2625b6120" For the 3com phone, those same lines look like this (and fails): Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Phone responds: Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" Basically, Asterisk wants to see a username in the Digest username field of 2321, but the 3com phone is sending sip:[email protected]. Anyone know how to tell asterisk to accept this format of username in the digest authentication? Here is the sip.conf info for that extension: [2321] deny=0.0.0.0/0.0.0.0 disallow=all type=friend secret=1234 qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes mailbox=2321@device host=dynamic dtmfmode=rfc2833 dial=SIP/2321 context=from-internal canreinvite=no callerid=device <2321 allow=ulaw, alaw call-limit=50 ... and for those interested in the grit, here is the debug output of the registration attempt: REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (no NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) confbridge*CLI REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 403 Authentication user name does not match account name Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) Thanks for your input!

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  • Asterisk SIP digest authentication username mismatch

    - by Matt
    I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue of older 3com phones that come without a SIP image. The 3com phones are communicating SIP with the Asterisk, but are unable to register because they present a digest username value that doesn't match what Asterisk thinks it should. As an example, here are the relevant lines from a successful registration from a soft phone: Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1cac3853" Phone responds: Authorization: Digest username="2321", realm="asterisk", nonce="1cac3853", uri="sip:192.168.254.12", algorithm=md5, response="d32df9ec719817282460e7c2625b6120" For the 3com phone, those same lines look like this (and fails): Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Phone responds: Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" Basically, Asterisk wants to see a username in the Digest username field of 2321, but the 3com phone is sending sip:[email protected]. Anyone know how to tell asterisk to accept this format of username in the digest authentication? Here is the sip.conf info for that extension: [2321] deny=0.0.0.0/0.0.0.0 disallow=all type=friend secret=1234 qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes mailbox=2321@device host=dynamic dtmfmode=rfc2833 dial=SIP/2321 context=from-internal canreinvite=no callerid=device <2321 allow=ulaw, alaw call-limit=50 ... and for those interested in the grit, here is the debug output of the registration attempt: REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (no NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) confbridge*CLI REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 403 Authentication user name does not match account name Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) Thanks for your input!

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