similar to: SIP IP Authentication - Socket or Via?

Hi, I've been doing some testing on moving from 1.2 to 1.4 and one issue I've encountered is re-transmits whenever an INVITE is cancelled. I have a stateless SIP proxy in fron of my asterisk servers (all it does is direct requests to one asteisk server or another) and the re-transmits do not occur on 1.2.17 which is the current verion I have in use on my production servers. The

I recently upgraded to the Gutsy beta, and this may just be a side effect of it being beta, but thought I'd raise it here in case I am doing something obviously wrong. I'm not familiar with how the configure script works and looking at the config.log didn't help me. Just to note, I was successfully compiling on ubuntu amd64 before I upgraded to the gutsy beta. I have raised a bug

I don't seem to be able to find the necessary hardware specs for an Asterisk server. What I have in mind is a dedicated server to serve 50 or so people. All users will use SIP phones and there will be an ISDN gateway for outgoing/incoming calls. Do you have any suggestions about the server specs (CPU, RAM, HD, etc)? Also, has anyone used Epigi Quadro ISDN gateway with Asterisk? If so, what is

I looking for a simple way to use cortado java applet on my website, help --------------------------------- Sick of deleting your inbox? Yahoo!7 Mail has free unlimited storage. Get it now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/theora/attachments/20070915/2df326af/attachment.htm

I found the details: 44100hz, 16bits, stereo I am looking at rewriting the program to record at 8Khz/16Khz/32Khz mono only should I record at 44100hz (and convert down) or record at the required hz level? ----- Original Message ----- From: <speex-dev-request@xiph.org> To: <speex-dev@xiph.org> Sent: Thursday, June 14, 2007 3:44 AM Subject: Speex-dev Digest, Vol 37, Issue 19 >

I am looking at writing a program using php that after X number of days (eg 60 days) a speex file is created that has all the speex file in a set folder with a bit of text to speech between the file for id reasons the speex file in the folder are deleted once done based on what I have read (corrent me if Im wrong): I need to convert the speex files to wav files (executable program), then I can

found this at the php website http://au2.php.net/manual/en/ref.oggvorbis.php I have nearly completed the protype version ----- Original Message ----- From: "Conrad Parker" <conrad@metadecks.org> To: "Tom Sparks" <tom_a_sparks@yahoo.com.au> Cc: <speex-dev@xiph.org> Sent: Monday, June 18, 2007 3:48 PM Subject: Re: [Speex-dev] audio stitching in php > On

what you are looking for is a Speech_recognition (http://en.wikipedia.org/wiki/Speech_recognition) program to convert to mp3 use lame (http://lame.sourceforge.net/index.php) (thats if the tape recorder encodes it file in wav (uncompressed) file witch is unlikely get yourself a sound editor like Audacity (http://audacity.sourceforge.net/) to do the convertion between the compress wav file and

Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on

I really hope that someone can help me with this one. DTMF tones are not working for certain places that I call, specifically 1-800-882-8880 which is the AA advantage line. It works for almost everyplace else. If I bypass asterisk, the call works fine. Network looks like: <SPA-2000> --SIP-- ASTERISK --SIP-- <AS5350> --PRI-- PSTN sip.conf entries [VGW01] (this is the AS5350)

Do anyone has an idea about an open source SIP API written in C# that can communicate with Asterisk, to call out? Regards, Sanjay.

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Does anyone know if the e164.org ENUM service is still active? If anyone who has anything to do with the e164.org ENUM site monitors this list could you check your signup page as the Captcha's (the test to see if you are human) fails for both the text and audio tests every time. I'd post a message on the e164.org forums but the signup page there has the test missing altogether. Greyman.

Hey there. I've been working on a project to integrate Asterisk with Exchange Unified Messaging via sipX using large parts borrowed from: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html ... and everything works surprisingly well. The one problem I have is MWI, or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so I've been looking into

> The easy answer is yes .. glibc requires so many things to be restarted, > that is the best bet. Or certainly the easiest. > > Note: in CentOS 7, there is also a kernel update which is rated as > Important .. so you should boot to that anyway: > https://lists.centos.org/pipermail/centos-announce/2016-February/021705.html > > Here is a good link to figure out what to

What does this mean? I've finally managed to install icecast with SSL support but there are some issues with ices2. [2020-10-23 00:26:03] INFO ices-core/main IceS 2.0.2 started... [2020-10-23 00:26:03] INFO signals/signal_usr1_handler Metadata update requested [2020-10-23 00:26:03] INFO playlist-basic/playlist_basic_get_next_filename Loading playlist from file

Hi, re-posting this with a more appropriate subject for my reply; > The easy answer is yes .. glibc requires so many things to be restarted, > that is the best bet. Or certainly the easiest. > > Note: in CentOS 7, there is also a kernel update which is rated as > Important .. so you should boot to that anyway: >

Hi I have a question if there was a major change in CDR? Few days ago I have upgraded to 1.4.12.1 from 1.4.4 and something bizarre happened. After the upgrade I have no call details in the cdr table when the call did not go through because of for example: Unable to create the channel of type Sip - no route to destination. In such situation the call does not exist in the cdr table while it was

Hi, Unfortunately this is not the case. For some reasons (separation, fraud detection) they separate the customer using their source IP address. So I am still looking for some solution. Thanks, Attila From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Glenn Geller (VDOPh) Sent: Wednesday, May 25, 2016 11:47 PM To: Asterisk Users