similar to: Asterisk Radius Module

I am trying to configure a VegaStream 50 FXO to work with asterisk. The problem that I am having is that the VegaStream does not support incoming registration from asterisk. VegaStream only allows outbound registration. My question is does asterisk allow incoming registration from an FXO? If yes how? Or better yet, has anybody been able to make the VegaStream FXO work with asterisk? According

Hi This is hard work :) I have read the Asterisk Handbook, BudgeTone User Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages and more. I am not a linux newbie but am new to Asterisk. I have failed to find any docs that explain how to get a very very simple, minimal, system up and I am trying to get the following to work: 2 BudgePhone 102D connected on a LAN to a

Sorry for the partial post a moment ago With help I got two phones communicating - PCMA/PCMU was the problem. Next stpe is to try voicemail. VM works fine, I can leave a mesage and then retrieve it - but no MWI on the phone and no stutter dialtone. I promise I've spent the requisite 4 hours + on google etc. but have really no further ideas. The setup is 2 Grandstream phones (latest

On 21/09/2020 15:00, Elias Pereira via samba wrote: > Another doubt is about this bydefaults entry. > The dc4 has this entry, but the dc3 does not. The dc3 is the fmso roles guy. > Does it work that way or is there something wrong there? Whilst there are a few attributes that do not replicate, all DN's should. > * Comparing [DOMAIN] context... > > * DN lists have different

On 21/09/2020 00:35, Elias Pereira via samba wrote: > No errors occur, but still not replicating. It is not of all entries. > > For example, there are some users that the groups they are part, differ > between the two DCs. > > In the link below there is a diff between groups of a specific user. > # ldbsearch -H /var/lib/samba/private/sam.ldb "(samAccountName=125202)"

hey all, I'm trying to double forward a DID , from PSTN to SIP DID then to my *. Double forwarding my DID , but the Digits are not getting relayed correctly ... sending double digits and sometimes missing digits? Is here anywhere I can put a delay on DTMF? Or a different solution for this? I use asterisk 1.2.1 , centos 4.2 , codec ulaw Let me know oliver@wvg-tele.com Oliver World

Hi John, Try adding username=5702 and username=5703 to each of the configs in sip.conf. I recall I had this problem with the Grandstreams. -----Original Message----- From: John Coll [mailto:john.coll@csoft.co.uk] Sent: Saturday, January 03, 2004 11:56 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)

I''m curious what would be the best way to represent people and whether the person has someone marked as their friend. Here are my thoughts so far: People table - id - name Relationships table - person_id - friend_id Now my problem is I don''t know how to set up these associations (if this is even correct). Friend_id would be an id to someone else from the people table.

On 21/09/2020 15:23, Elias Pereira via samba wrote: >> The question has to be, why do you only have it on one DC ? > > I am also asking myself this question because the dc3 was the first one I > provisioned. > > Ok. Somehow this was removed. Would it be possible to recreate this entry > or the best thing to do is to change the roles to dc4 and provision another > DC

Dear all, I have postfix delivering to dovecot lda (via spamassinassin). I have configured the quota plugin, but it does not seem to respect the details in the quota table in mysql. The query I have used looks correct to me, and the user can connect to the database. I wonder if someone might put me straight on the misconfiguration. Best wishes, j. # Relevant portions of dovecot -n:

Anyone know if #include works in ael yet? extensions.ael: #include "inc/pbx/global.conf" context test_context { }; *CLI> ael reload May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root token '#include' May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete: Requested contexts didn't get merged

On 20/09/2020 15:31, Elias Pereira via samba wrote: >> But, the error with the replication still occurs. I thought the replication >> error was linked to the ghost entry error. >> # samba-tool drs replicate DC3 DC4 dc=campus,dc=company,dc=br --full-sync >> --sync-forced >> ERROR(<class 'samba.drs_utils.drsException'>): DsReplicaSync failed - >>

Anyone with experience with RSpec care to share? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://wrath.rubyonrails.org/pipermail/rails/attachments/20060528/29c9ba64/attachment.html

Dear all, I want to setup the incoming calls, that don't use authentication in sip.conf file. My configurations as follows, [2000] type=peer host=dynamic insecure=port,invite ; (both) context=Testing But when I call '2000', I noticed the following message in Asterisk console, "NOTICE[5702]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user

Hi all, Quick couple of questions... 1) Will Ubuntu run the Wine emulator? and 2) Will Outlook 2003 function under the emulation without a whole lot of headache? THANKS for any help you might provide. Travis

Hello guys, I''m a newbie in RoR and I have a problem. My DB structure is ====================================== topics table : id(int,11) name(varchar,255) categories : id (int,11) topic_id(varchar,255) name(varchar,255) ====================================== --Models:-- class Topic < ActiveRecord::Base has_many :categories end class

Hey everyone, A few employees have noticed some problem here and there when trying to make outgoing phone calls. After it happens, they try again, and are able to call through. The dial plan for outbound calling looks like below. Which I know they are getting to the Congestion part (which explains the busy) but what I can't seem to figure out is the cause for why they are getting sent

Hi!I've installed Asterisk@home and I have a ISDN card,(Cologne Chip Design GmbH ISDN network controller [HFC-PCI](rev 0.2) This is how I installed bristuff: how to install hfc card after unload asterisk and amportal whit amportal stop type "setup" unselect zaptel in system service... and set the lan --->reboot<--- cd /usr/src wget

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

Hello all, Are you aware of the rails ActionController::Base.allow_concurrency option ? If this option is true, rails is able to process simultaneous multiples actions. It appears that rails is thread-safe (if the user application is), and that this option is mostly unknown. Webrick take care of this option in rails-1.1.2/lib/webrick_server.rb at line 80 : REQUEST_MUTEX.lock unless