similar to: Installing Asterisk on MAC native

Glenn, I can't use MySQL as each node currently has MySQL however there is a lot of data that is stored locally on each box. I may have to take this route if I can't find something else but that would mean syncing all sorts of data that does not need to be synced. On Tue, Jul 30, 2019 at 10:03 PM Glenn Geller (VDOPh) <ggeller at vdo-ph.com> wrote: > Hi Dovid, > >

Hi, I am running several Asterisk boxes with realtime around the world. Does anyone have a recommendation for a "light" db that would work with Asterisk that would also allow replication between all sites (so if I add an entry to one box it will work with the rest)? TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! I would like to reopen a discussion that I saw a couple of years ago, with the subject "Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine" The use case is simpe: There are providers that want to see a separate source IP address for each of their customers SIP trunks. Now, if we have an asterisk box with several customers, we have a problem.

Hi all, I have a user who would like to stream their favorite radio station from iHeart radio for their music on hold. It this TECHNICALLY possible? If so, any pointers would be appreciated. Is this LEGAL in the US? Thanks in advance, Mike. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am looking for a replacement for my Thomson ST2030SIP. My specifications are as follows : - 2 lines. - 6 BLF keys. - PoE. Can you give me a return on the models you use ? Thanks. Sil

?I have the same exact issue. I cannot push any sounds or even Playtones to the caller, unless the channel is answered, which is not possible for billing reasons. I am also using the Local channel & Dial(PJSIP/...). I think this is a bug in Asterisk 13. The Dial function has not answered yet, so the Local channel should be able to play anything to the caller, without answering, in parallel

On Fri, Sep 16, 2016 at 5:55 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Tried with both softphone (Ekiga) and snom IP phone, contact header > contains the public IP. but from header contains the private IP. after > OPTIONS method sent by the server. client sends an Register with expires 0. > Ok, did setting from_domain work? > > Best

Oh! In that case ignore it. Asterisk won't rebind the adapter if you've only changed parameters. The message is misleading -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Vitor Mazuco Sent: Monday, August 29, 2016 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law. We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are the only ones who don't need to receive the calls in our servers to sign them. We do this over a MySQL call, easily connectable to Asterisk via

On Thu, Sep 15, 2016 at 8:38 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Thanks for the reply. > > Yes my PABX is on the cloud when I try to register to the server, the > server sends registration OK with public address but OPTION method > includes the private address of the server in from header not the public > address. I have include

I changed in asterisk.conf mindtmfduration = 50 The inter-digit duration is for this function SendDTMF when we dial out dtmf The question is, how do I change it without changing the source code? On Sat, Jun 7, 2014 at 1:00 PM, <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To

Hello, you can have a look on Homer http://sipcapture.org/ regards On 27/09/2016 10:39, Gholamreza Sabery wrote: > Hello, > > For service monitoring you can use tools like sipsak in combination > with Zabix or Zenoss. Also using Zenoss or Zabix you can monitor the > health of your servers. This way you have both top-down and bottom-up > monitoring. For monitoring call

I just see warning? 2016-08-29 11:30 GMT-03:00, Telium Technical Support <support at telium.ca>: > This shows that asterisk's IAX is already bound to all adapters - so that's > good. Symptomatically does your IAX stop working? Or do you just see a > warning? > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com >

On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad <faheem2084 at gmail.com> wrote: > > > On Wednesday, 14 September 2016, Madushan Geethanga < > mgliyanage.rc at gmail.com> wrote: > >> Hi, >> >> What is the equal option for externip in asterisk 13 with pjsip. I have >> tried >> >> external_media_address=XX.XX.XX.XX >>

2016-09-02 20:40 GMT+02:00 George Joseph <gjoseph at digium.com>: > > > On Fri, Sep 2, 2016 at 9:34 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> I had a recent case where Asterisk stopped due to a segfault. >> This reminded me that being sure that whenever such issue occurs, it's >> useful to have a core file or various

Hi, see the log below root at AsteriskSlave:~# ip addr 1: lo: <LOOPBACK,UP,LOWER_UP> mtu 65536 qdisc noqueue state UNKNOWN group default link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00 inet 127.0.0.1/8 scope host lo valid_lft forever preferred_lft forever inet6 ::1/128 scope host valid_lft forever preferred_lft forever 2: p3p1: <BROADCAST,MULTICAST> mtu

So if someone has their own hardware and infrastructure but wants a software (not FreePBX but perhaps similar) what options do we have? Would like to virtualize it and not stuck with any one virtualization technology. Discuss... :) Travis -------------- next part -------------- An HTML attachment was scrubbed... URL:

Marcelo Terres wrote: > I noticed another different behaviour. > > In older versions, when I call rasterisk, I receive some informations > about it. Fox example: > > [root at pbx2 ~]# rasterisk > Asterisk 11.22.0, Copyright (C) 1999 - 2013 Digium, Inc. and others. > Created by Mark Spencer<markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type

Hi Andrew and Willy, Thanks for sharing the info. As for enabling radius server debugging 'radiusd -X', made some test calls don't see the radiusclient sending data to radius server. However, using radtest or radiusclient testing, able to send data to radius server (after enabling debug). For further testing, on my other server using OpenSIPs, setup the radiusclient and data was

Hello, I have installed the nlme package, but every time I try to use the function "summary.lme," I get a message that the function cannot be found. I've tried to install the package several times and have re-started R several times, but to no avail. I have also loaded the lme4 package, in case that makes a difference. I'm using R on a brand new Mac. Thank you!