similar to: Google Voice receiving call problem

OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can't seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes

Dear all, First off, I am very new to asterisk so forgive me if any of my comments or questions seem trivial. Thanks to [this post](http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/) and [this post](http://www.davidvossel.com/?p=28), I have GV set up on asterisk through jabber.conf and gtalk.conf. I can successfully dial out from asterisk. I'm trying to set up an

Hi there, I'm in the process of switching from using Samba4 internal DNS to using BIND as my backend DNS. However, I'm currently running into some issues with the transition. Here's an example of the messages I'm getting from /var/log/messages logs: Nov 9 15:34:26 pho-dcpvl-01N named[27524]: samba_dlz: starting transaction on zone DOMAIN Nov 9 15:34:26 pho-dcpvl-01N

Hey Rowland, Below is a cutdown version of my DHCP. As you can see, I haven't really set anything up for ddns-update. While using Samba4's internal DNS I had the setting 'ddns-update-style interim;' and it seemed to have worked fine. But with bind I'm not sure what else is needed. Thanks for taking a look at it. Philip # # DHCP Server Configuration file. # see

Thanks a lot. I'll take a read through it and see if I can get it working. ________________________________________ From: samba [samba-bounces at lists.samba.org] on behalf of Rowland Penny [rowlandpenny241155 at gmail.com] Sent: Monday, November 09, 2015 5:02 PM To: samba at lists.samba.org Subject: Re: [Samba] Samba_dlz: canceling trasaction on zone domain On 09/11/15 21:28, Philip Banh

This is a mail in response to streamripper being threatened by legal action from Live365. The DMCA strikes again. jack. ----- Forwarded message from Fred von Lohmann <fred@vonlohmann.com> ----- Delivered-To: jack@localhost.cantcode.com Delivered-To: jack@icecast.org X-Authentication-Warning: penguin.onehouse.com: majordomo set sender to owner-pho@onehouse.com using -f X-Sent: 31 May 2001

I am banging my head over a simple asterisk trick I was seeing on one asterisk server. An extension dials an international premium number, the called number answers, then the extension hangups, but the call continue to run on the international number side, generating an high profit for the premium number company and a big loss for the asterisk owner. I think some sort of "transfer"

No, there is no Music On Hold starting and the bad thing is the call duration reported by asterisk was just few seconds while the call duration reported by the provider was few thousand seconds, the max allowed. So they will be able to terminate the call on the asterisk side and have it run on the provider side. Leandro 2016-09-15 19:18 GMT+02:00 Max Grobecker <max.grobecker at

> From: "JD Conley" <jdc@malibuboats.com> > To: "'vorbis@xiph.org'" <vorbis@xiph.org> > Subject: RE: [vorbis] new MS codecs > Date: Thu, 14 Dec 2000 10:49:58 -0800 > X-Mailer: Internet Mail Service (5.5.2653.19) > Reply-To: vorbis@xiph.org > > Oh yeah, they have some samples on their site. Interestingly enough, they > don't

What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I just discover an "hidden" problem with AEL macro I want to have your feedback. If you use a macro to dial out, like &dialout(${EXTEN}), the leg extension will became ~~~~s~~~~ and if it happens you transfer the call, that will be the callerid appearing on the other phone display. I am just rewriting all the dialplan getting rid of the macro and using gosub, even if asterisk is

Hello, almost any useful CLI command has an analogue on Asterisk Manager Interface, but I cannot find a way to get the list of subscriptions using AMI. Which is the command, if any? The CLI command is "sip show subscriptions" Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'd like to dial two extensions (or external number) and ask for confirmation to accept the call. Dialing an extension, asking for confirmation and then dialing a second extension if the call has not been accepted is easy by using the dial option U(...), but if I dial two extensions at once, when the first answers, the other stops ringing. Any idea to make the first continue to ring until

Hello, I was wondering if there is a tool that can create the realtime database structure for latest Asterisk version or a web resource/file containing the sql scripts. Hope I haven't missed obvious things, I had no luck searching on the web, in the wiki I found few pages with bits of sql or table structures, like:

Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then

You can use "Command" command, and "sip show subscriptions" as a parameter -- Alex Epshteyn email: alex at thirdlane.com web: www.thirdlane.com phone +1 415.261.6601 ----- Original Message ----- > From: "Leandro Dardini" <ldardini at gmail.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at

Hey, all. I'm trying to do some fun with auto-provisioning of Polycom phones, and one thing that would make life easier for me would be if I could have a per-phone sip.conf file. If not, no biggie -- but if there's a way to do an include (as per extensions.conf) or something, that would be great. I've gone through docs, and an older version of "Asterisk: the Future of

Hi, On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest. Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly 1-2 concurrent calls. No other activity on server. Top shows asterisk on top. Its quad xeon server with 4 gb ram. Any suggestion where should I start and how should I go about with my investigation. Thank you and have a great weekend. Sans

When you set sendrpid=yes in sip.conf, a very nice feature is activated. When dialing an extension, the callerid of the dialed extension is returned back on the display of the calling phone. So if you call extension 100, you can see you are calling Ann (for example). I want to selectively disable the transmission of this information back to the caller. How can I do it? I tried setting

Hello, I was trying to use a queue in linear order and to provide the exact order of members to dial by adjusting the uniqueid value. Obviously it doesn't work and it seems an old problem: https://issues.asterisk.org/jira/browse/ASTERISK-18480 Realtime configuration can't identify "orders" in the list of results, so the members for the queue are returned in random order.