similar to: Anyone knows how to receive a SIP call withoutregistering gateway?

Hello everyone, I am pulling my hair here because a carrier threw me curve early today. They want to send calls to my asterisk server using SIP. Then they said that their gateways don't have to register with my server, that all they have to do is send a prefix for validation. Whereas I can think of several ways to authenticate their incoming number string, I am only used to the orthodox

Hello, It's happens around 40 calls and above ? The **machine** accepts number of invites(we can see by tcpdump ) , but asterisk sees part of them (we can see by CLI log) , and when it does , asterisk accepting an invite it reply the initator. (as it should ) ? but the rest of invites are just ignored. it's seems like an O/S issue, because on asterisk level I can all going

Hi Guys, I've configured a fax support in my system with opensource app_fax/spandsp 0.6 - but there is a problem - if FAX transmission got interrupted - system permanently returns "The call dropped prematurely" - and no fax is submitted. This is sometimes very inconvenient, say if someone have sent a multipage fax with 20 pages, and it got interrupted on 19th page, the person will

Hello everyone, Even though a lot of it was a bit last minute, several of us from the commnunity made it to Baltimore to help Digium with their booth at ISPCon. It was a great time. Gregory Boehnlein, Brian Capouch, Christian Savinovich, Kristian Kielhofner (me), and John Todd (not pictured) were there (as well as others), and some pictures were taken (the up close ones of me were very

I looks like someone decided to bundle a patch that hasn't been merged yet. Good for testing, not so good for initial impressions. In /etc/asterisk/asterisk.conf add or uncomment this: [options] ;silence_suppression=yes And see if that helps. You need a timing source for it to work, which is why it is disabled by default, but the logging might be a bit chatty in any case. Dan

Hey all, I am getting this error code publishing and action. 346 API_EC_EDIT_FEED_BODY_LENGTH Feed story body is too long Looks like the limit is currently "200" , what do you guys think about putting validation into Facebooker for this kind of thing? So that it blows up right away. Dave

Nvidia previously announced open source LLVM CUDA compiler but now I couldn't find it anywhere. Can anyone help me to point out where I can download source code for playing with? Best Regards, Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20120920/45d097d5/attachment.html>

I know this is offtopic, but I hope someone here can give me some guidance. I'm attempting to build sleepyhead (not even on Centos, so it is really OT) and I hit this error: Project ERROR: Unknown module(s) in QT: help Makefile:44: recipe for target 'sub-sleepyhead-make_first-ordered' failed make: *** [sub-sleepyhead-make_first-ordered] Error 3 I don't know anything about

see subject... -- Die unaufgeforderte Zusendung einer Werbemail an Privatleute verst??t gegen ?1 UWG und ?823 I BGB (Beschlu? des LG Berlin vom 2.8.1998 Az: 16 O 201/98). Jede kommerzielle Nutzung der ?bermittelten pers?nlichen Daten sowie deren Weitergabe an Dritte ist ausdr?cklich untersagt!

Dear all. I am searching for KMeans ++ for R. I cannot find it. Do you know any package with it? Best regards, Rui

On 04/14/2011 07:26 PM, LiMaoquan2000 wrote: > Hi All, > Many Thanks to Underwood for her excellent review of our big trouble > which prevent LMS-based AEC algorithms to be used in most computer. > Maybe it can be summaried as follows: > 1. Different sample rate of sampling and rendering does exists in most > low-cost soundcards (In my experiments over more than 20 soundcards,

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com

I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and output have very fine tolerances on most standard audio

I thought as a means of preventing ssh-agent hijacking by remote hosts one could have a local process communicating with the agent, simply by having a term open with this sort of dialog: agent-mon: on host foo.elite.com requesting agent forwarding for host bar.elite.com (fingerprint matches known_hosts) allow? [yes/no]: I know concepts presented along with patches are prefered, but I'm

BJ, >BJ Weschke <bweschke@gmail.com> >Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom >SIP termination vs. DS3 >To: Asterisk Users Mailing List - Non-Commercial >Discussion <asterisk-users@lists.digium.com> >Message-ID: <79cf63305051908056c284cc9@mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 >Did I miss pricing/availability

Hi, I have 2 asterisk boxes as Gateway, in this arrangement. (PANASONIC PBX) - [ASTERISK1] - network - [ASTERISK2] - (ANALOG PHONE) everything works great, in both directions (receiving and making calls), but when i get a call on the (ANALOGPHONE), I haven't been able to transfer it to another extension of the PANASONIC PBX using the flash key. I've tried the using the t T options on

Hi, I just installed a fresh CentOS 6.6 desktop. It's a client's machine, it is physically installed on a testbench in my office. Usually, when I perform installations, I start with the base system on the testbench, and once networking is configured, I SSH into it and then do all the fine-tuning remotely. My configuration files, scripts and HOWTOs are all stored in a Github

The Asterisk Development Team would like to announce the release of Asterisk 13.29.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.29.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp <jcolp at digium.com> wrote: > On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote: > > I get the desired use case to run app_amd from within a Stasis > > application, but I’m not sure about app_queue. You have everything at > > your disposal within ARI itself to replicate all of the functionality > > of app_queue and

The first release candidate of Asterisk 1.2.0 has been released! It is available from the ftp.digium.com FTP servers, as well as the Digium CVS servers (under the 'v1-2-0-rc1' tag). This release includes a large number of improvements over beta2, including: * Many bug fixes * Documentation and sample configuration updates * New 'stack' applications Gosub/Return/etc.