similar to: BUG or ???

So I have a route that looks like this: scope "4" do scope "public" do scope ":apikey" do resources :shops end end end And a bunch of controller specs, an example of which looks like this: describe ShopsController do describe "when responding to a GET" do context "#new" do it "should create a new instance

HI ALL got small question i use call-limit=1 on peers but call limit is not working if user is not registered on PBX and making calls so the main question is -- how to Disallow CALLS without registering on PBX -- Best regards Antony tel. +380669197533 tel2. +380636564340 Paypal http://paypal.me/Satskiy

> On 11/02/2017, at 3:40 am, Frank Vanoni <mailinglist at linuxista.com> wrote: > > On Thu, 2017-02-09 at 14:58 +0200, ????? ?????? wrote: > > >> so the main question is -- how to Disallow CALLS without registering >> on PBX > > sip.conf configuration > In the [general] section, define: > > > [general] > ... > allowguest=no >

Here are two patches that let xm create, save and restore extract and preserve elfnotes read by the domain builder. This is handy for a few things. In particular, I''d like it so that xm can decide whether or not guest domains support fast resume (if save fails, or for checkpointing). _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com

Hi guys when I execute the piece of code bellow on RoR console it works fine: url = URI.parse("https://us1.api.mailchimp.com/2.0/helper/ping") request = Net::HTTP::Post.new(url.path) http = Net::HTTP.new(url.host, url.port) http.use_ssl = true http.verify_mode = OpenSSL::SSL::VERIFY_NONE request.body = "{\"apikey\": \"myapikey\"}" response = http.start

Hi. I was studying about how to use databases in Asterisk, accessing it from the dial plan. In my project, my dial plan will have to store simple data (ex: IP number, port number, device name, etc) in a persistent way, so that it will be possible to retrieve such information in future moments, still via dial plan. For this case, I would like to know? 1. What is the best choice for storing and

Hello colleagues, I have tried to use the package bigmemory, biganalytics and biglm. I want to specify a multivariate regression with a weight. I have imported a large dataset with the library(bigmemory). I load the library (biglm) and specified a regression with a weight. But I get everytime an error message like "object not found" or "`weights' must be a

hi, do you have someone example of http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/ in node.js asterisk-ari ? thanks Marek

Hello, fail2ban does not ban offending IP. NOTICE[29784] chan_sip.c: Registration from '"user3"<sip:1005 at asterisk-ip:5060>' failed for 'offending-IP:53417' - Wrong password NOTICE[29784] chan_sip.c: Registration from '"user3"<sip:1005 at asterisk-ip:5060>' failed for ?offending-IP:53911' - Wrong password systemctl status

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

Dear list, I've been using R for a while, but am new to web services. I'm a relatively novice programmer; advance apologies for incorrect terminology. I'm trying to send queries and get results back from a SOAP server, using the SSOAP package. My code contains sensitive API keys and URLs, and unfortunately I'm unable to share it uncensored. I have not been able to reproduce the

The Asterisk Development Team has announced the release of Asterisk 11.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

Hi, I am able to get Xend running on Fedora 8 but when I try to create any VM it fails. The xm create osc.cfg throws following error - [root@asds174 vm_images]# xm create osc.cfg Modified arg=osc.cfg stop3 Using config file "./osc.cfg". group_id=None [''create'', ''osc.cfg''] Stop 1 None [''vm'', [''name'',

HI LIST CAN U HELP ME If there are multiple sip trunks with the same ITSP then an incoming call is arbitarily matched to the last peer with the same host IP address. This is not a serious problem because the DID is still correct but it does have many insidious effects due to the incorrect channel name Example register=myaccount1 at sip.myitsp.com/line1 register=myaccount2 at

hi list can U help me caller id in USTM if now working -- Starting switch on '4211 at 4211-1' to 4203 -- Executing [4203 at office:1] DumpChan("USTM/4211 at 4211-0x7f7ba4228fd0", "") in new stack Dumping Info For Channel: USTM/4211 at 4211-0x7f7ba4228fd0: ================================================================================ Info: Name=

*friends help me * *cant get incoming calls in asterisk* *(when i connect **80081 in softphone ---every thing is ok**)* *<--- SIP read from UDP:200.152.125.221:5060 <http://200.152.125.221:5060> --->* *INVITE sip:80081 at 10.47.10.10:5060 <http://sip:80081 at 10.47.10.10:5060> SIP/2.0* *Record-Route: <sip:200.152.125.221;lr;ftag=as6872d065>* *Via: SIP/2.0/UDP

-- Luca Pradovera luca.pradovera at gmail.com Hello, sorry, I managed to lose the reply amidst the traffic. What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer. Leg B then decides that user C knows more about the subject, and initiates a blind transfer to C?s phone

The Asterisk Development Team would like to announce the release of Asterisk 13.33.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.33.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.33.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.33.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

facing problem with originating webrtc calls 1-when iam doing call from webrtc iget ice working <--- SIP read from WS:91.196.158.205:1466 ---> INVITE sip:0669197533 at 77.91.132.9 SIP/2.0 Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315 Max-Forwards: 69 To: <sip:0669197533 at 77.91.132.9> From: "Anton" <sip:1065 at 77.91.132.9>;tag=5i21qaop43 Call-ID: