search for: registeration

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

...168.8.3). In the sip.conf, the following configuration to the bilal_sip done: [bilal_sip] type=friend context=internal host=dynamic canreinvite=no dtmfmode=rfc2833 In the general context, I did allow=all for the codec and there is not any disallow. The same SIP ATA endpoint can do a successful registeration on an softswitch by just put the IP address of this softswitch instead of the IP address of my Asterisk. Also, why in IAX2 we keep receive registeration messages every periodically while we do not receive that periodic messages in SIP? Who can help? What kind of traces I can do to know the reason...

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)

...that's probably not what you want. > Last point: I noticed that some endpoints that are not > able to register (when secret is required), then I was > not able to see any log at the asterisk side while SIP > client still not registered. At least, it should > display the fail for registeration, why does not > display it? Is it related to my v tracing level? Where > in the same tracing level, I am able to see the > registeration fail if the endpoint sent an wrong > username. For example if the context was [bilal_sip] > and the endpoint username was "bilal_1000" th...

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 go back to older CVS version ? Please recommend relatively stable version. *CLI> WARNING[1142106560]: File chan_sip.c, Line 428 (__sip_xmit): sip_xmit of 0x80de5a...

...le I can see this in IAX2? Is it related to my v tracing level? Last point: I noticed that some endpoints that are not able to register (when secret is required), then I was not able to see any log at the asterisk side while SIP client still not registered. At least, it should display the fail for registeration, why does not display it? Is it related to my v tracing level? Where in the same tracing level, I am able to see the registeration fail if the endpoint sent an wrong username. For example if the context was [bilal_sip] and the endpoint username was "bilal_1000" then I see a the message (l...

...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 request to register on the softswitch or it can send directly without registeration? (Note: the trunk is SIP). Please check the below configuration and advise me if it is correct: [aloonet] type=peer qualify=yes host=193.111.196.240 ; IP Address of the softswitch canreinvite=yes context=outbound disallow=all allow=g723 nat=no Is it OK? Will it register on my softswitch or w...

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 ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/pro...

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

..., how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed in its sip.conf? Anyone can advise me for the main needed thing to be done to acheive this? By the way: I succeed to let Polycom SIP phone to place a call via Asterisk without registeration, and without setting the IP address also. But from Asterisk Box to another Asterisk Box, I can not until now. Any help? Regards Bilal

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

Hello, i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for even uncompleted calls (i.e. calls they make whaich have already be setup but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my

...n the following: #/usr/sbin/asterisk -cvvv CLI>reload But always I get a message at the firefly that an error occured while trying to connect to the network. What else I have to do? By the way: what is the command that I can type it to do tracing on the user [iax2user1] or to do traces on any registeration attempts from the clients? Last thing, if I am outside the console (in unix mode), is there any command from unix I can type it to know if asterisk is running or not? Regards Bilal ____________________________________________________________________________________ Moody friends....

...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 can be done with firefly softphone? C) One time I succeed to register my asterisk on another softswitch (sip registeration), but when I routed calls via this IP Trunk, then the calls are not deliver to the softswitch at all, and the error at asterisk says that eveyone is bussy. I do not know why? Registeration succeed but calls are not appear at all on the softswitch screen. By the way: my Asterisk still does not suppo...

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.