similar to: Using CURL

Hi, I'm starting testing 1.6.2 beta. CentOs 5.2 I found my first crash, first I have [Mar 20 20:30:41] WARNING[11201]: res_config_mysql.c:611 update_mysql: Attempted to update column 'useragent' in table 'sip', but column does not exist! [Mar 20 20:30:41] ERROR[11201]: res_config_mysql.c:581 update_mysql: MySQL RealTime: Updating on column 'lastms', but

hi we wanted to embed a rails application into our cms generated php files. small example: static site: www.someweb.com/someweb/somepage.php the code in the somepage.php file will use curl (remote http call) to load the rails application into its layout. now the problem is that all the urls (links, forms) generated by the rails application will leave somepage.php and point to something like

Hello, Hopefully I'm posting to the correct list, but if not, please shun me ;). I'm running Asterisk 1.4, with Festival 1.4.1. I've got a test extension setup, Festival configured and for some reason, when I dial that extension I get this: [Jan 10 17:16:05] WARNING[9082]: app_festival.c:511 festival_exec: Festival returned ER See the full debug below: [Jan 10 17:16:05]

Hi. I have the following line in the default context of all my internal extensions: exten => 9876,1,Transfer(125) When I dial extension 9876 from any sip phone, * dutifully transferrs it to extension 125, which is just what I want. Unfortunately when I dial 9786 from my Zap connected analogue phone, the transfer doesn't go through and the dialplan drops through to a hangup. debug

Hello Rowland, the home folder creation works now(!) It was a misunderstanding on my side. The key phrase in the wiki is: "Close the users properties window with „OK“ to save the modification. **The users home directory is created on the fly during the save processes.**" This is a different behaviour as with the "profiles" folders which are created during the first login

Hi all, Now that there is an effort underway to improve the look/theme of the Wiki. I would like to discuss the content of the wiki. IMO there are too many links on the frontpage. With a new theme, the News/Events items on top can go into a sidebar, so that will be taken care of the news items. I will not discuss them here. But currently, there is little structure in all the links/titles on the

somple question here but pretty confusing on my side. i have a model towns that is mapped to my towns table. can i add an extra variable to the class and access it in my view? basicall what i am trying to do is return a list of towns, and next to each town, display a total number of what i have in each town. ive tried both attr_reader and @total but no avail. here is my view <% for

?? Hello.? We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2).? We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex.? If a person starts talking they will cut off others on the call.? Is this normal behavior?? Are there any options I can change to change this? ?? Thanks! James -------------- next

Hiya everyone. I have been working on a fun little app to watch what's going on in your asterisk box via its manager interface. There's a screenshot up and some info at http://sitkavoip.com/astsee/ -- Sorry it requires allegro, but I was more keen about getting the ideas down than worrying about the framework. Comments/questions welcome, but probably off-list is best unless they

Add a new Curl_failed exception, and raise it when Curl.run fails, instead of exiting directly with error. This allows users of Curl to handle failed downloads gracefully. Add wrappers to the "main" functions of virt-builder, virt-v2v, and virt-v2v-copy-to-local to catch Curl_failed, and show a better message for them. --- builder/builder.ml | 9 ++++++++- common/mltools/curl.ml

Hi All. In curl manual: --ciphers -> "(SSL) Specifies which ciphers to use in the connection. The list of ciphers must specify valid ciphers. Read up on SSL cipher list details on this URL:http://www.openssl.org/docs/apps/ciphers.html" In openssl: http://www.openssl.org/docs/apps/ciphers.html#CIPHER_SUITE_NAMES there are names such as "SSL_RSA_WITH_RC4_128_SHA RC4-SHA" but

Hello All, I'm having an issue where asterisk doesn't hear any audio after transferring to voicemail. Here is the dial plan and console output. DIAL PLAN [voicepulse-in] exten => _14259491337,1,NoOp(Incoming call from VoicePulse) exten => _14259491337,2,Ringing exten => _14259491337,3,Wait(1) exten => _14259491337,4,Dial(SIP/1337,20) exten => _14259491337,5,VoiceMail(1337)

hi Tuomas, https://www.centos.org/somepage - our 404 pages dont look the best they could, and dont have any urls leading off to good starting points for people following stale urls. I was wondering if you might be able to conjure up something for us to use as a 404 destination ? no rush, whenever you are able to. and as always, thanks in advance! regards, -- Karanbir Singh +44-207-0999389 |

Hi List; I read the following sentence: "The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable" In the following link: http://www.voip-info.org/wiki/index.php page=Asterisk+CLI+prompt The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? Also where I can find information about it? Regards Bilal Ghayad

So I finally started building a script that does it. Anyone interested in helping with it so that it produces readable output? First step is getting raw wiki pages. I'll do this directly by getting them from the filesystem. You can get them from here: http://dovecot.org/tmp/wikipages.tar.gz Next there's http://dovecot.org/tmp/wiki2txt.pl which is my horrible Perl script that kind of

Hiya, hope I don't bore anybody with this. There are certainly a lot of monitor-y things out there and they just didn't fit my need, so maybe this will fit someone's besides mine. http://horanappraisals.com/asterisk/pbxmonitor/ contains two files. one is a php script called pbxmonitor, and one is a flat file of extensions to extension name mappings of internal users. It

Hi, Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)

I've been trying to fix MoinMoin's text_plain formatter to produce readable output. It's almost correct, except I don't understand why I can't get the LF-adding rules correct. Either there are too many LFs or too few. And whenever I change something, something else breaks.. It's getting pretty annoying. So if anyone else wants to try getting it fixed that'd be great.

These are now passed using a curl configuration file, which is a little bit safer than using command lines. virt-builder doesn't need to pass usernames and passwords to curl, but if it ever does in future this will be a lot safer. --- builder/Makefile.am | 1 + builder/builder.ml | 6 +-- builder/downloader.ml | 94

We've encountered a strange issue with the trunk version of asterisk. Our dialplan makes CURL calls and occasionally CURL stops working. The dialplan looks something like this: [macro-curl] ; ${ARG1} CURL URL ; ${ARG2} CURL POST exten => s,1,NoOp(CURL) ... exten => s,n(post),Set(RF_CURL_POST=userID=${RF_DIALER_USERID}&password=${RF_PASSWORD}&${ARG2}) exten =>