similar to: SPA3000 dial plan?

This this definition: group { "puppettestgroup2": gid => 1235, ensure => present } user { "puppettestuser2": gid => "puppettestgroup2", uid => 1235, ensure => present } I see this: notice: Starting

I have an issue where * tries to route a call over IAX to another server even if the server is down. I have included the relevant entries from my iax.conf, extensions.conf, and some debug output. If someone could tell me what I have configured incorrectly, I would appreciate it. Thanks, Stephen -----------iax.conf on voip2---------- [voip1] type=friend username=voip1 host=x.x.x.x (ip

Hi, I just upgraded from 1.2 to 1.4. In 1.2, when I did a "show uptime" I used to see a second line telling me the time since the last reload. Has this been removed in 1.4? The following is the output of my two test boxes: Connected to Asterisk 1.4.18.1 currently running on voip2 (pid = 10605) Verbosity is at least 3 voip2*CLI> show uptime System uptime: 15 hours, 55 seconds

I'm looking to use * for a HQ/branch office topology with fairly few calls over the WAN. The questions I have all pertain to the following architectural pic: http://www.45891.com/misc/arch.jpg I'm looking at a distributed architecture so users are somewhat functional when the link to HQ is down, with a centralized voicemail server to allow for transfer of voicemail messages from user to

On Wed, Sep 9, 2020 at 9:53 AM Phil Perry <pperry at elrepo.org> wrote: > On 09/09/2020 07:15, Dominik Holler wrote: > > > > > > On Wed, Sep 9, 2020 at 4:05 AM Brian Stinson <brian at bstinson.com > > <mailto:brian at bstinson.com>> wrote: > > > > __ > > On Fri, Sep 4, 2020, at 14:26, Mason Loring Bliss wrote: > >>

On 09/09/2020 07:15, Dominik Holler wrote: > > > On Wed, Sep 9, 2020 at 4:05 AM Brian Stinson <brian at bstinson.com > <mailto:brian at bstinson.com>> wrote: > > __ > On Fri, Sep 4, 2020, at 14:26, Mason Loring Bliss wrote: >> On Mon, Aug 31, 2020 at 05:28:58PM -0400, Jonathan Billings wrote: >> >> > This is pretty

On Fri, Sep 4, 2020, at 14:26, Mason Loring Bliss wrote: > On Mon, Aug 31, 2020 at 05:28:58PM -0400, Jonathan Billings wrote: > > > This is pretty misleading. Isn't Stream just a collection of repos that > > offer pre-release packages for the next RHEL point release? It's not a > > midstream between Fedora and RHEL. > > Along those lines, if someone's

On Wed, Sep 9, 2020 at 4:05 AM Brian Stinson <brian at bstinson.com> wrote: > On Fri, Sep 4, 2020, at 14:26, Mason Loring Bliss wrote: > > On Mon, Aug 31, 2020 at 05:28:58PM -0400, Jonathan Billings wrote: > > > This is pretty misleading. Isn't Stream just a collection of repos that > > offer pre-release packages for the next RHEL point release? It's not a

Is there a way I can have "voice mail check" calls coming from my internal users automatically get to the right extension, without having the user enter their extension? I'm thinking that I could have the local SPA boxes translate, or have each user live in a context where the extension in question exists uniquely per user, but both of these seem kludgey. Thanks in advance for

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

Hi, all. I'm running the Asterisk 1.0.7 package that comes with Debian Sarge. I'm new to Asterisk, but so far almost everything is going well. However, I'm having a Hell of a time with Music On Hold. I've got an extension set up to test Music On Hold, as follows: exten => 3000,1,WaitMusicOnHold(300) I've tried the following in musiconhold.conf: default =>

On Mon, Aug 31, 2020 at 05:28:58PM -0400, Jonathan Billings wrote: > This is pretty misleading. Isn't Stream just a collection of repos that > offer pre-release packages for the next RHEL point release? It's not a > midstream between Fedora and RHEL. Along those lines, if someone's adjusting text in there, I'd tend to think "rolling release" indicates that one

Hi I have an account with mynetphone (australia), which gives me two voip (sip) accounts, which i used to have connected to a spa9000. this is behind a firewall, so on the spa9000 I would listen on another port apart from 5060. so on the firewall 5060 would go to voip1 and 5061 to voip2. I moved to asterisk (+tdm410) and the machine was also the firewall and I had no problem - well atleast it

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

Hello, all. I've recently switched over to Dovecot from a long stretch of using Courier IMAP, and there's one thing I don't have working that I'd quite like. With Courier I had a magic trash folder in which things I deleted would live for a specified amount of time, after which they'd be deleted. I see how to get the expiry behaviour I want, but not the magic saving-to-mailbox

CentOS 6.9 has support for the Haswell processor which is a subset of the Skylake. All Skylake server ancillary components work perfectly with CentOS 6.9. The AST2500 used with the Skylake is very good and will process UHD. However this is not backwards compatible with legacy variants. To make it work an ancillary graphics card is required which is less than $100. This will not break the

Dear all, Can someone from Netherlands who has SPA3000 send me the Regional & Line1 settings. I'm not sure what is wrong but I can call from asterisk to phone attached to spa300 FXO, but not the other way. I tested three phones: siemens gigaset, tiptel 160 and hpoj k80(fax). Only tiptel 160 can call to the asterisk, other two just dial but nothing happens. I guess this has something with

Hi, I'm trying to get zapras do work, I had downloaded the pppd-source and the 2 patches. I succefull compiled and install the patched version of pppd, but got this error in message-log Jul 15 11:43:32 voip1 pppd[9296]: In file /etc/ppp/filters: unrecognized option 'active-filter' Jul 15 11:43:57 voip1 pppd[9299]: Plugin zaptel.so loaded. Jul 15 11:43:57 voip1 pppd[9299]: Zaptel

Hello, I'm french, so excuse my poor English. I'm face to a terrible thing, with has stole a lot of my time. On the .184 machine, I've the following iax.conf : [general] rtcachefriends=yes bandwidth=high tos=reliability jitterbuffer=no autokill=yes #include "iax.voip1.conf" #include "iax.renoir.conf" The iax.voip1.conf file contains : [VOIP1] type=friend