similar to: Asterisk, GalaxyVoice and Humble Pie

I got a call from GV on Monday evening telling me they wanted me to move my Asterisk server over to a new IP address (216.229.127.40) by this saturday. Why the couldn't tell me this in an email is beyond me but anyways .. So I done changed the number and so far its all ok but whilst testing I noticed that I could no longer accept incoming phone calls. I swapped back and still no inbound

One of my buddies signed up with GV yesterday with a view to using them on his * server. Problem is that the settings they gave him don't work with asterisk. They do however work with X-Lite. Any ideas? He's using the settings outlined on my web page. Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hi Folks, I've been playing with call recording for our support department which was kinda going ok until I spotted something odd in the CDR. None of the support calls are being entered into the CDR properly. I'm using mysql as the back end and Areski's web based front end and all was going fine. The problem seems to be that the CDR doesn't get populated with the destination

Hi. A friend of mine came asking about this VOIP provider. I haven't heard of them so I thought I might ask the list. Anyone has any experience using them (http://www.optonline.net/Home, http://www.optimumvoice.com/index.jhtml)? Not just with Asterisk, even with their supplied HandyTone and a regular analog phone. Thanks -------------- next part -------------- An HTML

Dear Asterisk Community, I would like to offer my most humble apology for the enormous stream of mail messages currently battering at your mailboxes. I was honestly trying to help a worthy cause. Please blame me and only me. Neither Russell nor Josh are at all to blame for this. Nor, for that matter, is Olle. I am the guilty party. Please direct all anger/outrage/hatred/etc. towards

Look, I pay nothing for an incredible operating system with enormous features and stability, so it feels a bit awkward asking for more. [Sarcasm On] Now, get going and build me up a 5.3 kernel-vm's would ya? ;) [Sarcasm Off} -- Humbly, John Thomas

Do any providers support speex codec? I just searched for a while and couldn't find anything. I'm looking for both SIP vendors (such as galaxyvoice) and IAX / IAX2 vendors (such as voipjet). I'm looking for a company that would be cheap for low-volume use. Thanks, -- Daniel JB Clark | http://pobox.com/~dclark

Dear all, I have used Markdown via Drupal a few months now, with a customer site and my currently my own blog and based on that practical experience I wanted to add some extra functionality to PHP Markdown Extra, so out of this grow my (extended) weekend project https://github.com/egil/php-markdown-extra-extended The big missing item right now that I would like to implement is the ability to add

Let's say you were wanted to terminate calls onto your Asterisk system but your only available codec was G.729 and you had no control over the remote SIP proxy sending you the traffic. What would you do? Does anyone have an update on Asterisk supporting T.38 with SIP? Thanks! chris -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am thinking of dumping broadvoice so I need some other VoIP providers that have a las vegas DID and a service better than broadvoice. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050131/bf4b1c31/attachment.htm

On 2020-03-04 10:21, Kees Cook wrote: > On Wed, Mar 04, 2020 at 10:21:36AM +0100, Peter Zijlstra wrote: >> But at what cost; it does unspeakable ugly to the asm. And didn't a >> kernel compiled with the extended PIE range produce a measurably slower >> kernel due to all the ugly? > > Was that true? I thought the final results were a wash and that earlier >

On 2020-03-04 10:21, Kees Cook wrote: > On Wed, Mar 04, 2020 at 10:21:36AM +0100, Peter Zijlstra wrote: >> But at what cost; it does unspeakable ugly to the asm. And didn't a >> kernel compiled with the extended PIE range produce a measurably slower >> kernel due to all the ugly? > > Was that true? I thought the final results were a wash and that earlier >

On Wed, Mar 4, 2020 at 10:45 AM H. Peter Anvin <hpa at zytor.com> wrote: > > On 2020-03-04 10:21, Kees Cook wrote: > > On Wed, Mar 04, 2020 at 10:21:36AM +0100, Peter Zijlstra wrote: > >> But at what cost; it does unspeakable ugly to the asm. And didn't a > >> kernel compiled with the extended PIE range produce a measurably slower > >> kernel due to

On Tue, Mar 03, 2020 at 01:19:22PM -0800, Kees Cook wrote: > On Tue, Mar 03, 2020 at 01:01:26PM -0800, Kristen Carlson Accardi wrote: > > On Tue, 2020-03-03 at 07:43 -0800, Thomas Garnier wrote: > > > On Tue, Mar 3, 2020 at 1:55 AM Peter Zijlstra <peterz at infradead.org> > > > > But,... do we still need this in the light of that fine-grained > > >

On Tue, Mar 03, 2020 at 01:19:22PM -0800, Kees Cook wrote: > On Tue, Mar 03, 2020 at 01:01:26PM -0800, Kristen Carlson Accardi wrote: > > On Tue, 2020-03-03 at 07:43 -0800, Thomas Garnier wrote: > > > On Tue, Mar 3, 2020 at 1:55 AM Peter Zijlstra <peterz at infradead.org> > > > > But,... do we still need this in the light of that fine-grained > > >

I've complied the OH323 .so successfully and can easily receive calls from my H323 gatekeeper (using 711u), however it seems that all outgoing calls are refused and I'm getting "reason 23 (Temporary failure)" as an error code which I can't find documented everywhere. My H323 gatekeeper needs a 001NXXNXXXXXX to dial out to the PSTN even if I'm in north america (Montreal)

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on host it's comming

Greetings all! I'm facing a puzzle I have not been able to solve. I need to make an EPS of a pie-chart (Yes, I know; please don't bother to tell me! I just need to ...). I'm trying to do it with pie(), and I want to have just the plain pie-chart with no annotations. So far so good: "labels=rep(NA,...)" will do it. But I want to have it output to an EPS file with the

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