search for: registeres

Hi, If I''m using STI with Devise, I have a Admin model inheriting the base Devise User model. I would like to remove ''registerable'' from the Admin model but it inherits registerable from the user model. How would i disable registration for admins? -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To

Hi List; I have a problem when trying to let an SIP ATA endpoint (got it from broadtel company), I am getting the following message: - Registered SIP 'bilal_sip" at 0.0.0.0 port 5060 expires 60 I do not know why it takes it 0.0.0.0 while it has an IP address (192.168.8.3). In the sip.conf, the following configuration to the bilal_sip done: [bilal_sip] type=friend context=internal

On 8 August 2010 23:39, Abder-Rahman Ali <lists-fsXkhYbjdPsEEoCn2XhGlw@public.gmane.org> wrote: > I tried to make the application from scratch again, and > notices that I get the following when I run: $ rake db:migrate > > (in /Users/abder/Desktop/Rails/auth) > ==  DeviseCreateUsers: migrating > ============================================== > -- create_table(:users)

Dear Philipp; Kindly find the part of the configuration as below: [general] allow=all disallow is comment by ( ; ). [bilal_sip] type=friend context=internal host=dynamic canreinvite=no dtmfmode=rfc2833 So where is the problem? The endpoint does not register and nothing appear on trace level 3. And the amazing thing that if the endpoint send wrong username (for example: bilal_sip100) then it

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040108/748d21b3/attachment.htm -------------- next part -------------- Hello I downloaded latest CVS version and got following error messages while registeration. ( CVS Version Asterisk CVS-01/08/04-14:20:34) As a result IP Phone don't register with the Asterisk. Is it broken ? How can I

Hi List; I noticed that if I disabled secret in the context by placing ( ; ) before it, then at the asterisk the log will be: -- Registered SIP 'bilal_sip' at 0.0.0.0 port 5060 expired The IP address of the endpoint was not captured!!! Why? If secret enabled, then some endpoints can not register (maybe due to compatibility in reading the negotiation packets), so what is the solution?

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

Hi List; How can I configure asterisk to receive a call from SIP end point without being registered at asterisk and its IP address is dynamic, and authentication to be based on the username and password or any other string? I know that if I place the host with static IP then no need to register, but what if the voip gateway was having dynamic IP and I do not need to register on asterisk, but I

Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal

Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. -------------- next part -------------- An HTML attachment was

I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is any registerable incoming volume from these lines. I've been running them at rxgain = 25 (zapata.conf) to make the audio audible, however this creates poor call quality issues

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

On Thu, Sep 7, 2017 at 11:11 AM, Robinson, Paul <paul.robinson at sony.com> wrote: > Different intrinsics sounds like a good solution to me. J > > > > So what happens with the case where a variable is registerized but later > we decide to spill it? Presumably we'd have a dbg.addr to point to the > spill slot. In past compilers I've used, spill slots were

...unks> <============> (<Secondary PBX>) Clients are connected to the Secondary PBX. this pbx handles registration of all clents. The billing irregularities happen on the Secondary PBX. When a call is maked from the Secondary and it is routed across the trunks, call disposition always registeres 'AWNSERED', unless the Primary PBX sends back a busy signal. the duration and billsecs are always equla. this means that the user gets billed for ring time, and calls disconnected from the Secondary PBX Can someone help me out here ? Thanks -------------- next part -------------- An HTML...

Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password

help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it

Hello! I am using icecast 2.3-kh2 (for the 302 redirect feature), and all my relays are authenticating with user/pass. Is it possible to know how many listeners I have for any given mountpoint counting all the relays? []s Pablo -- Pablo Lorenzzoni <pablo@propus.com.br> GnuPG ID: 0x268A084D at pgp.mit.edu http://www.propus.com.br/ - Propus Inform?tica

Hello all, As I've been exporing libvirt api, I've seen event registerers, and types of events are defined, undefined, and you know the rest. Is it all for event types? Any ideas about how to look for a device access event, for example? Kind regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Concerning the shift instructions: In C, the effect of shifting by any amount larger than the operand width is undefined. In the design of the IR, either these operations need to be explicitly stated as undefined or a well-defined value needs to be established for them. Concrete example: What does a left shift of an i32 by 33 bits produce? The risk of making these operations defined is that

On Mon, 2008-04-28 at 21:39 -0500, Lane Schwartz wrote: > On Mon, Apr 28, 2008 at 8:31 PM, Gordon Henriksen > <gordonhenriksen at mac.com> wrote: > > On 2008-04-28, at 21:19, Lane Schwartz wrote: > > > > > On Mon, Apr 28, 2008 at 2:13 PM, Gordon Henriksen <gordonhenriksen at mac.com > > >> stack and discover the return address of each call frame.