similar to: TDM400P takes too long to ring

What response do you get to *CLI> pri show spans ? On 30 June 2015 at 09:34, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13

On several occassions lately, my home Asterisk box has stopped registering with my VoIP provider. I haven't been able to reproduce the problem, and the log doesn't contain anything useful. How can I increase the log verbosity for SIP registration-related events? I've looked through logger.conf and tried searching with Google, but I haven't been able to find a clear answer. This

How about: exten => s,1,Dial(SIP/alice&LOCAL/555 at delayed-announce,40) [delayed-announce] exten => 555,1,Wait(20) same => n,Playback(myannouncement,noanswer) same => n,NoOP(Whatever else you want to do goes here) The 'noanswer' option on the Playback means that SIP/alice should continue to ring for the remaining 20 of the 40 seconds, as the Playback will not answer

Maybe try progress() instead of answer () ?????? 23 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???: > Thank you, I just tried your suggestion. Strangely, the announcement is > played only if I try to dial a SIP peer which is not available (not > registered to be more precise). If the SIP peer is available, I only get > the ring tone, and never hear

I got a call from an overseas call center telling me about the problems with the Windows machine I was using. They wanted to remote in and fix things for me ... (Ignore the fact I use a MacBook Pro or an ASUS laptop with Debian). What I found curious was the caller's name was Asterisk, and the caller's number was Asterisk at 10 or or Astrk at 10 similar. (I don't recall the exact

Background: I dabbled with asterisk years ago, and more recently have more-or-less functioning IncrediblePBX systems for experimenting, but I want to understand more so I'm now working with distro packages (Ubuntu) and hand edited configurations files. I have three SIP "trunks", each providing me with a UK telephone number. They are for my home (1), my wife's home-based

Hello users, I have a Digium Te235 and asterisk 13 which have worked well with 1 carrier but we have failed to add a 2nd carrier. The second telco brings their E1 line over finer, terminated in a RAD modem and they give me ethernet to the E1 card. It's the first time i am having install such a solution, which ideally would be not a big problem. However, The physical layer has failed to

Thank you for the idea. The problem with RetryDial, is that it will cancel the first call, play the announce and then dial the SIP peer once again, so the telephone will display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 ? 17:57, John Kiniston a ?crit : > You could try using RetryDial() instead of Dial, It supports playing > an announcement. >

As we plan for our 10th AstriCon, which will be in Atlanta, GA the week commencing October 7th, we want to make sure that our conference sessions are the best they've ever been! That's why we need YOU to submit a speaking proposal - to share you experiences and ideas around Asterisk! The best sessions at AstriCon are always those from the Community that show what you've been up

I am looking for any information of the possible use of Asterisk in any of the systems used by/for the London Olympics. Please get in contact if you know of any such use(s). Thank you, David Digium logo David Duffett Digium, Inc.??? Director, Worldwide Asterisk Community 6 Landscape Close, Weston on the Green ??? Bicester, Oxfordshire OX25 3SX ??? UK direct/fax: ? +1 256 428 6119 ??

It's amazing what you can learn in a few days... Having just found out that Queen Elizabeth has a great sense of humor, it has now emerged that Kevin Fleming - a man who (both with and without his moustache) has been an amazing contributor and influencer in the Asterisk project is set to move on to a new challenge outside the project - but still within the realms of Open Source. Kevin has

I am wondering what the best solution is for initiating a call from Outlook Contacts. I imagine something that would start the call from the outlook card (or similar) and then dial the user's extension and the contact's phone number and place them in a bridge. Anyone use something like this? Travis Ryan Director of Information Technologies Oscar Winski Company 2407 North Ninth Street

What is the best virtual server tech (and most stable, etc) to use for a asterisk virtual hosting environment? I have a client that wants to do virtual hosting of Asterisk (only SIP or IAX, no PRI, etc) and I'm wondering if Xen or something else would be best? We'd like to stay away from the costs of VMWare if possible. Thanks! Travis -------------- next part -------------- An HTML

I have tried with xen and kvm both are working fine. On Wed, Apr 6, 2016 at 3:44 PM, Loic Chabert <loic.chabert at voxity.fr> wrote: > Hello, > > Work well with kvm and centos 7. > Some ajustements has to be made with systemd. > > I'm using it in production since 1.5 year now, no issue to report. > > Regards. > Le 6 avr. 2016 21:13, "Yves"

yessir From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Marcelo Terres Sent: Wednesday, July 19, 2017 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] corosync and Asterisk 13 Did you installed the dev package? corosync-dev Marcelo H.

I want to use corosync and installed it via ubuntu repository. I guess there is a version 1 and 2 of corosync. For some reason ./configure for Asterisk (13) isn't recognizing I have corosync installed. I can't enable the res_corosync module in menuselect. Any ideas? Thanks! Travis -------------- next part -------------- An HTML attachment was scrubbed... URL:

Humm, if I put a filter in this lines, maybe back? 2016-01-05 12:36 GMT-02:00, Ryan, Travis <RyanT at oscarwinski.com>: > >> -----Original Message----- >> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- >> bounces at lists.digium.com] On Behalf Of Vitor Mazuco >> Sent: Tuesday, January 05, 2016 9:21 AM >> To: Asterisk Users Mailing

> -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Joshua Colp > Sent: Wednesday, September 23, 2015 6:38 PM > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU > 14.04 > > On 15-09-23 07:36 PM, Ryan, Travis wrote:

I have an 14.04 server with Asterisk 13.5 that will recognize it. It has corosync and corosync-dev version 2.3.3 on it. The one I can?t get to recognize it is 16.04 ubuntu, with corosync and -dev version 2.3.5. Won?t be recognized by ./configure for Asterisk 13.17, 13.16, or Asterisk Certified 13.13. I REALLY need some help figuring this out. ? Thanks! From: asterisk-users-bounces at

If you want to use dahdi dummy driver inside asterisk for timer then this is possible with openvz based container virtualization. We have tested vicidial in this mode for 5-10 agents and it worked well. Mitul Limbani On Apr 8, 2016 8:52 AM, "Pete Mundy" <pete at fiberphone.co.nz> wrote: > List, > > Might as well throw my hat in the ring! > > I can't say