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Search found 271 results on 11 pages for 'asterisk'.

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  • Allowing asterisk in URL - ASP.NET MVC 2 - .NET 4.0 or encoding

    - by raRaRa
    I'm having a trouble allowing asterisk (*) in the URL of my website. I am running ASP.NET MVC 2 and .NET 4.0. Here's an example that describes the problem: http://mysite.com/profile/view/Nice* The username is Nice* and ASP.NET says there are illegal characters in the URL: Illegal characters in path. Description: An unhandled exception occurred during the execution of the current web request. Please review the stack trace for more information about the error and where it originated in the code. Exception Details: System.ArgumentException: Illegal characters in path. I have tried all the Web.config methods I've seen online such as: <pages validateRequest="false"> and <httpRuntime requestPathInvalidCharacters="" requestValidationMode="2.0" /> So my question is: Is it possible to allow asterisk in URL? If not, is there some encoding method in .NET that can encode asterisk(*) ? Thanks!

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  • When should you NOT use the asterisk (*) when declaring a variable in Objective C

    - by Jason
    I have just started learning objective c and the asterisk is giving me some trouble. As I look through sample code, sometime it is used when declaring a variable and sometimes it is not. What are the "rules" for when it should be used. I thought it had something to do with the data type of the variable. (asterisk needed for object data types, not needed for simple data types like int) However, I have seen object data types such as CGPoint declared without the asterisk as well? Is there a definitive answer or does it have to do with how and what you use the variable for?

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  • Server Fax Farm - Need suggestions, or advice

    - by Mike Curry
    We're Looking at creating a large fax farm via T.38 (Fax over Voip - hundreds of incoming and outgoing faxes) on linux servers, anyone have any suggestions on what is available? All my searches return using Asterisk 1.6.x with a commercial product from Digium called "Fax for Asterisk" (with required purchase of "channels" at $38.00 per channel). There must be an open source project out there I can't seem to find. Suggestions welcome! Here is some additional info: We're using Ubuntu 9.10, and planning to use T.38 If I have missed anything, let me know.

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  • How to send T.38 from a mac?

    - by Brian Postow
    I'm trying to set up a fax-server on a macintosh. I have Hylafax, and we're going to use an internet FOIP fax provider (Haven't decided who yet, that may be another question). The problem is how to get from Hylafax to T.38. I know of two options, but I'm not sure how to decide between them: T38modem Advantages: It's only one extra program, and i know that I can compile it for the Mac. (well, At least I can get the H323 version working on a Mac) Disadvantages: It is mostly undocumented and seems to be supported only by one guy in Russia. IAXModem/Asterisk Advantages: It's well known, and well supported. We can pay for support. It presumably does the T38 with SIP correctly, so we don't have to worry about it. Disadvantages: It's two separate programs. While I know how to get Asterisk on a mac, I'm not sure about IAXModem. (It's sourceforge, and linux, but compiling things for a mac isn't always straight forward...) It's also mostly undocumented. Do these seem like an accurate listing of the pros/cons? Anyone have any other suggestions? thanks.

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  • What does the Asterisk * mean in Objective-C?

    - by Thanks
    Is it true, that the Asterisk always means "Hey, that is a pointer!" And an Pointer always holds an memory adress? (Yes I know for the exception that a * is used for math operation) For Example: NSString* myString; or SomeClass* thatClass; or (*somePointerToAStruct).myStructComponent = 5; I feel that there is more I need to know about the Asterirsk (*) than that I use it when defining an Variable that is a pointer to a class. Because sometimes I already say in the declaration of an parameter that the Parameter variable is a pointer, and still I have to use the Asterisk in front of the Variable in order to access the value. That recently happened after I wanted to pass a pointer of an struct to a method in a way like [myObj myMethod:&myStruct], I could not access a component value from that structure even though my method declaration already said that there is a parameter (DemoStruct*)myVar which indeed should be already known as a pointer to that demostruct, still I had always to say: "Man, compiler. Listen! It IIISSS a pointer:" and write: (*myVar).myStructComponentX = 5; I really really really do not understand why I have to say that twice. And only in this case. When I use the Asterisk in context of an NSString* myString then I can just access myString however I like, without telling the compiler each time that it's a pointer. i.e. like using *myString = @"yep". It just makes no sense to me.

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  • what does the asterisk mean after a filename if you do ls -l

    - by James.Elsey
    I've done an ls -l inside a directory, and my files are displaying like this : james@nevada:~/development/tools/android-sdk-linux_86/tools$ ll total 9512 drwxr-xr-x 3 james james 4096 2010-05-07 19:48 ./ drwxr-xr-x 6 james james 4096 2010-08-21 20:43 ../ -rwxr-xr-x 1 james james 341773 2010-05-07 19:47 adb* -rwxr-xr-x 1 james james 3636 2010-05-07 19:47 android* -rwxr-xr-x 1 james james 2382 2010-05-07 19:47 apkbuilder* -rwxr-xr-x 1 james james 3265 2010-05-07 19:47 ddms* -rwxr-xr-x 1 james james 89032 2010-05-07 19:47 dmtracedump* -rwxr-xr-x 1 james james 1940 2010-05-07 19:47 draw9patch* -rwxr-xr-x 1 james james 6886136 2010-05-07 19:47 emulator* -rwxr-xr-x 1 james james 478199 2010-05-07 19:47 etc1tool* -rwxr-xr-x 1 james james 1987 2010-05-07 19:47 hierarchyviewer* -rwxr-xr-x 1 james james 23044 2010-05-07 19:47 hprof-conv* -rwxr-xr-x 1 james james 1939 2010-05-07 19:47 layoutopt* drwxr-xr-x 4 james james 4096 2010-05-07 19:48 lib/ -rwxr-xr-x 1 james james 16550 2010-05-07 19:47 mksdcard* -rw-r--r-- 1 james james 205851 2010-05-07 19:48 NOTICE.txt -rw-r--r-- 1 james james 33 2010-05-07 19:47 source.properties -rwxr-xr-x 1 james james 1447936 2010-05-07 19:47 sqlite3* -rwxr-xr-x 1 james james 3044 2010-05-07 19:47 traceview* -rwxr-xr-x 1 james james 187965 2010-05-07 19:47 zipalign* What does that asterisk mean? I'm also unable to run a particular file, as follows : james@nevada:~/development/tools/android-sdk-linux_86/tools$ ./emulator bash: ./emulator: No such file or directory EDIT : I'm trying to get Eclipse to use emulator, but it keeps complaining the files does not exist, yet it is here?

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  • Recommended FXO Gateway for UK analogue lines

    - by Bryan
    Can anybody recommend any FXO gateway devices to connect analogue telephone lines to an Asterisk VoIP system. Requirements: Minimum of 4 ports. Enterprise grade - quality is more important than price. For UK analogue telephone lines. - I don't know if this makes a difference or not? I'd also be interested to hear bad experiences, so I can get an idea of which devices to avoid.

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  • What is the significance of '*' (star, asterisk) in the file listing results?

    - by vfclists
    I have noticed that some of my files have an asterisk at end. Does the asterisk at the end have any particular significance? I think they are mostly executable and displayed in green by the ls command. You will see that ./bkmp* and ./bkmp0* have an asterisk at the end. They are executable bash scripts. Here's my output: drwxr-xr-x 7 username username 4096 Oct 2 18:28 ./ drwxr-xr-x 8 root root 4096 Oct 2 09:25 ../ -rw-r--r-- 1 username username 3724 Sep 22 03:06 .bashrc -rwxr--r-- 1 username username 319 Sep 22 03:42 .bkmp* -rwxr--r-- 1 username username 324 Sep 29 23:30 .bkmp0* drwx------ 2 username username 4096 Sep 17 13:52 .cache/ -rw-r--r-- 1 username username 675 Sep 17 13:37 .profile drwx------ 2 username username 4096 Sep 22 10:10 .ssh/ drwx------ 2 username username 4096 Sep 24 19:49 .ssh.local/ drwxr-xr-x 2 username username 4096 Sep 22 04:10 archives/ drwxr-xr-x 3 username username 4096 Sep 24 19:51 home/ -rw-r--r-- 1 username username 27511 Sep 24 19:51 username_backup.20120924_1908.tar.gz

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  • Sonicwall settings for Polycom TFTP

    - by Michael Glenn
    I'm switching our VoIP phones (Polycom 301s and 501s) to our data network. They were previously segmented to their own network. This means disabling the DHCP on the Trixbox (Asterisk) server and configuring the Sonicwall TZ 210 DHCP to indicate that Trixbox is the TFTP server. The Polycom phones are stating "could not contact boot server". All phones are configured to TFTP and were confirmed working when previously using the Trixbox server for DHCP. Trixbox DHCP is now turned off. I've configured options 66(as String), 128(as IP) and 150(as IP) in DHCP and added them to a TFTP Option Group. I've enabled "Allow BOOTP Clients to use Range" for the Dynamic IP range and assigned the Option Group TFTP as the DHCP Generic Option Group. Any idea what I'm missing? Is there a separate tool to inspect the DHCP response to compare Trixbox to the Sonicwall?

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  • Is there a SIP provider in the UK which provides the P-Asserted-Identity header?

    - by nbolton
    In the US, Flowroute (low cost SIP trunking provider) provides P-Asserted-Identity in the SIP invite request header (example screenshots). It also allows you to set the caller ID for outgoing calls, for example by using the follow in extensions.conf for Asterisk: exten => id,n,Set(CALLERID(all)=123) However, in the UK, I've tried a couple of SIP providers and none of them let me do either of those things (see P-Asserted-Identity or set the caller-ID). Is this because of some sort of restriction in the UK phone networks, or is it only available to really expensive SIP trunking providers?

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  • Troubleshoot telnet connection from Windows 7 to UNIX

    - by Sujay Ghosh
    I am trying to connect to an Asterisk server in USA. I am using telnet < IP Address 5038 from India to USA. The person in USA is able to telnet to the IP address and port from USA , but I am not able to do it from India. We are on different networks. I am using Windows 7 Ultimate, and have enabled the Telnet client. I have also used Putty without any success. Can someone suggest me what can be the problem and how can this be resolved.

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  • Test-service on Internet for testing incoming INVITE

    - by leiflundgren
    I am trying to set up Asterisk at home. I think I'm having trouble configuring my firewall, so that inbound traffic is accepted, but I am not sure. I got the idea that, perhaps, there is a service out on the Internet, where I can, though a web-browser, initiate an incoming call, an INVITE. And then see the SIP-trace that the remote-part experience. Anyone know of such a service? Note. I have a SIP-PSTN provider so I can generate inbound calls. But I cannot see the SIP-logs from my provider...

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  • Make call with alternate provider if NOANSWER

    - by adaptive
    I have two voip providers, one free an the other paid. The free provider only allows local calls to certain area codes, so I need to fall back to the the paid provider if a call fails. At the moment, I have the following context in my extensions.conf file: [globals] ; freephoneline.ca PRIMARY_PROVIDER=fpl ; voip.ms SECONDARY_PROVIDER=voipms [local] exten => _NXXNXXXXXX,1,Set(CALLERID(name)=${OUTGOING_NAME}) exten => _NXXNXXXXXX,n,Dial(SIP/${EXTEN}@${PRIMARY_PROVIDER}) exten => _NXXNXXXXXX,n,Set(CALLERID(num)=${OUTGOING_NUMBER}) exten => _NXXNXXXXXX,n,Dial(SIP/1${EXTEN}@${SECONDARY_PROVIDER}) exten => _NXXNXXXXXX,n,Hangup() I checked the logs and noticed that the free provider responds with NOANSWER if a call is not allowed (Even though it plays a message). What I want is to: Try calling the ${PRIMARY_PROVIDER} first. If NOANSWER is returned by provider (not that the callee did not answer), then call with ${SECONDARY_PROVIDER} How can I modify my dial plan to get the desired results? EDIT : The primary provider is freephoneline.ca, and I'm using asterisk v1.8.2.3-2

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  • Asterisk in eregi

    - by Mihai
    //URL START $urlregex = "^(https?|ftp)\:\/\/"; // USER AND PASS $urlregex .= "([a-z0-9+!*(),;?&=\$_.-]+(\:[a-z0-9+!*(),;?&=\$.-]+)?@)?"; // HOSTNAME OR IP $urlregex .= "[a-z0-9+\$-]+(.[a-z0-9+\$_-]+)*"; // PORT $urlregex .= "(\:[0-9]{2,5})?"; // PATH $urlregex .= "(\/([a-z0-9+\$_-].?)+)*\/?"; // GET Query $urlregex .= "(\?[a-z+&\$.-][a-z0-9;:@/&%=+\$.-]*)?"; // ANCHOR $urlregex .= "(#[a-z_.-][a-z0-9+\$_.-]*)?\$"; // check if (eregi($urlregex, $url)) {echo "good";} else {echo "bad";} but what if I have http://www.example.com/about-me/my-4*-hotel/ That eregi check isn't valid because of the asterisk. What should I do?

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  • Using an extension to block a caller

    - by Trewq
    I have a couple of SIP phones and use callcentric. I get a lot of junk calls. I'd like to implement the following feature and would like some suggestions on how to do this: Once I get a junk call, I typically hang up. I think want to dial some number (like *23 or something) and I'd like the last number that was received to be put in a database. Any future call from that number will be directed to VM or a busy tone. I'd appreciate some pointers on how I'd go about doing this.. I prefer an open source solution.

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  • not able to register sip user on red5server, using red5phone

    - by sunil221
    I start the red5, and then i start red5phone i try to register sip user , details i provide are username = 999999 password = **** ip = asteriskserverip and i got --- Registering contact -- sip:[email protected]:5072 the right contact could be --- sip :99999@asteriskserverip this is the log: SipUserAgent - listen -> Init... Red5SIP register [SIPUser] register RegisterAgent: Registering contact <sip:[email protected]:5072> (it expires in 3600 secs) RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout RegisterAgent: Failed Registration stop try. Red5SIP Client leaving app 1 Red5SIP Client closing client 35C1B495-E084-1651-0C40-559437CAC7E1 Release ports: sip port 5072 audio port 3002 Release port number:5072 Release port number:3002 [SIPUser] close1 [SIPUser] hangup [SIPUser] closeStreams RTMPUser stopStream [SIPUser] unregister RegisterAgent: Unregistering contact <sip:[email protected]:5072> SipUserAgent - hangup -> Init... SipUserAgent - closeMediaApplication -> Init... [SIPUser] provider.halt RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout please let me know if i am doing anything wrong. regards Sunil

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  • Linux DHCPD Mac-Address based Groups

    - by GruffTech
    Our Current DHCPD.conf looks like the following. subnet 10.0.32.0 netmask 255.255.255.0 { range 10.0.32.100 10.0.32.254; option subnet-mask 255.255.255.0; option broadcast-address 10.0.32.255; option domain-name-servers 208.67.222.222,208.67.220.220; option routers 10.0.32.5; host Dev-ABaird-W { hardware ethernet 00:1D:09:3E:49:13; fixed-address 10.0.32.94; } ... more static hosts .... } About as basic as it gets. The old router is 10.0.32.1, our company wanted to implement a squid proxy to better monitor web traffic while at work, and if necessary block large time-wasters, IE Facebook.com. However, we've quickly realized that this change has played a mean prank on our Polycom SIP Phones. Occasionally our phones will not ring, the end recipient hears ringing (this is artificially created by our PBX) however the handset never rings. The ONLY thing that has changed in our network is the option routers line. So, Since all Polycom MAC addresses begin with 00:04:F2 would it be possible in DHCP to say any 00:04:F2:::* MAC addresses get option routers 10.0.32.1, and anything else must talk with our Gateway?

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  • What is a fairly accepted endpoints to concurrent calls ratio for a PBX?

    - by Logan Bibby
    Hey guys! Not too positive about the relevance of this question to Server Fault, but I'll let you guys poke around at it anyway. :) I'm trying to figure out what an accepted ratio of endpoints to concurrent calls on the "average" office PBX. I'm defining concurrent calls as any call being placed, including internal calls. I'm not going to be considering call centers into this ratio, I'll be looking at those differently, anyway. To give you guys a little history behind the question: I need this ratio to be able to figure out decent specs for a virtual PBX service I'm going to be rolling out within the upcoming months. The ratio will determine the number of trunks needed per PBX system. -- Logan

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  • not able to register sip user on red5server, using red5phone

    - by sunil221
    I start the red5, and then i start red5phone i try to register sip user , details i provide are username = 999999 password = **** ip = asteriskserverip and i got --- Registering contact -- sip:[email protected]:5072 the right contact could be --- sip :99999@asteriskserverip this is the log: SipUserAgent - listen -> Init... Red5SIP register [SIPUser] register RegisterAgent: Registering contact <sip:[email protected]:5072> (it expires in 3600 secs) RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout RegisterAgent: Failed Registration stop try. Red5SIP Client leaving app 1 Red5SIP Client closing client 35C1B495-E084-1651-0C40-559437CAC7E1 Release ports: sip port 5072 audio port 3002 Release port number:5072 Release port number:3002 [SIPUser] close1 [SIPUser] hangup [SIPUser] closeStreams RTMPUser stopStream [SIPUser] unregister RegisterAgent: Unregistering contact <sip:[email protected]:5072> SipUserAgent - hangup -> Init... SipUserAgent - closeMediaApplication -> Init... [SIPUser] provider.halt RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout please let me know if i am doing anything wrong. regards Sunil

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  • Cheap Solution for Routing a Toll Free Number to a Standard POTS Number

    - by VxJasonxV
    I do some technical work for an Internet Radio Show/Podcast, and need to fix something that has been broken for a while. The hosts have a Skype-In number to take listener calls, and for convenience sake, I bought and paid for a toll free number for a period of time. I used to use Asterlink for routing calls, but they folded and sent my number to OneBox, but they're ridiculously expensive by comparison. I'm looking for a cheap solution for this one simple task. Forward toll free calls to a skype-in number. The definition of cheap is as cheap or cheaper than Asterlink was. I paid something like $2 a month, and then the termination/call rate, which was a fraction of a sent for termination, and only whole cents after some serious time on the call. A $20 preload lasted me months at a time. I don't want to be upsold too, I want a simple web based management screen (CDR/stats are fun!), and obviously, it needs to be reliable. What vendors out there are you a fan of that solves this need?

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  • Provisioning SIP Phones over the internet

    - by Jorge Fernandez
    I have a few SIP Phones that are located of site and connect to my PBX over the internet to make calls. For some reason one of these phones has become unprovisioned. In my office phones get provisioned by the server via TFTP. The ones that I have off site I pre-provisioned manually before I sent them off-site (I'm in Florida the phone is in New Jersey). Whats the best way to provision these over the internet? TFTP is very insecure. Sending the plain text profiles with the SIP Account and Password over the internet is out of the question. The phones have been off-site for about 6 months without any issues. Im using Trixbox and Cisco 7940 Phones.

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  • Cheap Solution for Routing a Toll Free Number to a Standard POTS Number

    - by VxJasonxV
    I do some technical work for an Internet Radio Show/Podcast, and need to fix something that has been broken for a while. The hosts have a Skype-In number to take listener calls, and for convenience sake, I bought and paid for a toll free number for a period of time. I used to use Asterlink for routing calls, but they folded and sent my number to OneBox, but they're ridiculously expensive by comparison. I'm looking for a cheap solution for this one simple task. Forward toll free calls to a skype-in number. The definition of cheap is as cheap or cheaper than Asterlink was. I paid something like $2 a month, and then the termination/call rate, which was a fraction of a sent for termination, and only whole cents after some serious time on the call. A $20 preload lasted me months at a time. I don't want to be upsold too, I want a simple web based management screen (CDR/stats are fun!), and obviously, it needs to be reliable. What vendors out there are you a fan of that solves this need?

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