similar to: Formatting in sip.conf...can you have 2 @ signs for register?

Hi all, I''m just starting out... the solution to this must be really trivial but can''t figure out what''s wrong. I''m using Webrick + Rails 1.0 The error is: undefined method `useremail'' error What am I missing? Thanks, Lorenzo class User < ActiveRecord::Base set_table_name "\"tblUser\"" set_primary_key

Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of

Greetings I have been running * for about a month now. Configuration. (5) Cisco 79xx IP phones (1) XP100P Pentium III (300mhz) 192meg memory Redat 8.0 (updated) It seems to run for about 3-6 hours, then the process stops. I have noticed, that * does not stop, if I do NOT have it register to other sip servers. (FWD and PCH). Here is are the last few lines in the /var/log/asterisk/messages

Hi to everybody, I need some help in configuration of the spa 3102. I created an account for line 1 (user 208, sip port 5061) correctly registered in asterisk, then i create an account in sip.conf like this: [general] register = line01:pwdsipura:line01@192.168.1.222:5060/095377078 [line01] username = line01 fromuser = line01 secret = pwdsipura host = 192.168.1.222 fromdomain = 192.168.1.222

First off, here is what I want to do: SIP Clients -> SER -> Asterisk -> VoIP provider Where SER will handle communications between SIP clients (since I would prefer that my SIP clients not use all of my bandwidth) Asterisk will handle calls to a VoIP provider I have read that people have similar setups working, but I have not seen any documentation of these setups. So far, SIP Clients

Hi, I have been trying to configure my Asterisk to use a Sip provider for out and incoming calls. I only have one user and password for connect to my sip provider. My sip.conf is: [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls callerid=No

I am having a problem when a SIP registration fails. I get the following messages in the log: Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 2874 (sip_reg_timeout): Registration for '<user>@fwd.pulver.com@65.39.205.114' timed out, trying again Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 5119 (handle_request): Registration from

Hello, I current using dovecot 2.0.11 and with all other 2x versions occurring the same problem. When i create the file .dovecot.sieve and compiled one .dovecot.svbin on user homedir, the imap show it like a folder, example: doveadm mailbox list -u user at domain.com .. .. *dovecot.sieve* *dovecot.svbin* .. .. I current using maildir and mail location is: *mail_location =

Hi guys, I have a server in Debian 8 with Samba 4.2.2 compiled about sources. This server is my project for small busines and run all network funcitons. The functions like AD, DNS, DHCP, NTP, File Server, Print Server, Openfire, OCS, GLPI, Squid and Squidguard are work integrated on Domain perfectly. My problem is that some clients need run a windows aplication so I put a windows server as member

Hi, I use samba 4 like domain controler, file server and print server. I am using 4.4.13 version and all its ok. I decide test the new version 4.6 and I have a problem with print server. I can upload 64bits drivers but I cannot upload 32bits drivres. Always give an error. I try upload driver with Windows 7 64 and 32 bits. Some one can help me? King regards. André Freire Gerente de Tecnologia

Sorry for the simple question. I'm trying to match "sipprovider.nocredit" but the following doesn't execute NoOp (it runs "context" but not "context-custom"). What am I doing wrong? [context] include => context-custom exten => _.,1,Set(GROUP()=1) exten => _.,n,Goto(destcontext,${EXTEN},1) [context-custom] exten => sipprovider.nocredit,1,NoOp(No

Hi, I am diverting an incoming call to a mobile phone and a landline using the following:- exten => 0203000000,3,Dial(SIP/442080000000 at sipprovider&SIP/44700000000 at sipprovider,120,r) For billing purposes, i need to be able to work out whether the diverted call was answered by the mobile or whether it was answered by the landline. The CDR only shows the full Dial() information, and

Hi, I´m have a Samba 4 Domain Member that I use like a Proxy Server. I use Squid with NTLM Athentication and work perfecly. My problem is Squidguard with NTLM Authentication. If I use Samba 4.2.X in my Samba 4 Domain Controler I watch in Squid LOG only the user name but If I use Samba 4.1.x or 4.3.0 in my Domain Controler I watch in Squid LOG domain\\user name and Squidguard Authentication not

Hi All, I have a server setup with an incomming PSTN line and a bunch of Grandstream BT100 phones. Is there a way for asterisk to foward an incomming callerID from the PSTN to the SIP phone that is setup as an extension? We have a Voice menu setup for incomming calls and I would like to recieve the caller ID of the calls we are recieving after the incomming caller reaches their final

Hi, I use Samba4 domain controler since 4.0 version over Debian. Today I use Debian 9.8 and Samba 4.9.6 without problems in many productions environments. Last week I decide make a test with Debian 9.8 and Samba 4.10. All ocours very fine trough compile process but when I try create domain with 'samba-tool domain provision" nothing happen. The shell freeze and the process don't go

(hardware in my computer: linux, asterisk, x100p, grandstream budge tone-100 ) Hi, When i run #asterisk ?v It show me a messages but when i try to incomming the call it show me that. Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'me@192.168.0.6' timed out, trying again Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration

Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the

On 19/10/15 16:46, mathias dufresne wrote: > AD from Samba or Microsoft is mainly a database for storing users (and > associated stuffs). It comes also with stuffs (protocols) to connect and > retrieve information. > > How the client uses these information is, as always, a choice from that > specific client. > > Your AD client is your Squid/Squidguard(ian) server. Its job

Hi All, I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as well as any configuration in asterisk thats needed, on this phone? Thank you, James -------------- next part -------------- An HTML attachment was scrubbed...

Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider "inode" fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422