similar to: asterisk codec sizes, data plus overhead

Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A)----> Cisco AS5xxx ----> ASterisk---->PSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains

Hello, I'm trying to receive faxes with asterisk. My configuration is like this: PSTN fax -> ISDN -> Cisco router with VoIP module -> Asterisk When I try to send a fax from PSTN fax I got the standard fax signal, Asterisk starts rxfax application and then call ends and there is no tif anywhere. On the fax display there is still one message: Calling... Part of my extensions.conf:

We appear to be having a problem with our asterisk setup. We have a cisco AS5300 with pri lines coming in and passing the calls onto asterisk then too the sip phones. the phone call from the sip phones (7960's) appears to be ok nice and clear including the user who has called in. but if your the user who has called in its all crackley sounds really bad when they speak. i believe this

Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM

Hi sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun

All, I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is.... One pri terminating into a Cisco 2431 router Sip messages from the Cisco get sent to a asterisk server linksys ata's a each remote end. I can receive the calling name if the call originates

I would like to know how to calculate the amount of bandwith I would need to host X number of calls. For example, if user A in San Francisco with an ATA 186 calls user B in New York with an ATA 186 and Asterisk is being hosted in a PC in Miami. How much bandwith do I need to have in Miami? Do I just need bandwith for the setup of the call (ie the SIP part) or are there any instances where the

What kind of stability / reliability are people currently experiencing with the Linux / Asterisk combination? We will be running 3-10 SIP phones from India to US using nothing more than regular cable / dsl connections from both locations. Also, what make / model SIP phone do you recommended that would allow us to configure the phones to work on alternate ports (or is this a standard

Hi,guys: I am interesting to separate the code by the clang . such as: void a() { ... } #ifdef Test void b() { } #endif void c() { .... } Is there some method for me to get the code in the region of TEST ? Thanks, Yao. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20150724/2c1435fa/attachment.html>

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

Hello, Has anyone experienced a segmentation fault in asterisk for using G729 against an AS5300 in SIP ? I'm having this problem. Also, any other codec I use gives me incompatible media (can be read in SIP DEBUG messages). AS5300 configured for multiple codecs, so is Asterisk. Tried G711u/A G723 and G.729. Any clues ? Regards, Jorge A. Info: Asterisk ver 1.0.7 stable Using AMPortal

I'm am currently using autuvb4 at q-2 mono 44100. This produces roughly 28 to 34kb/s. It's ok but I've heard AACPlus at 32kb/s stereo and it's definately better, unfortunately. Regards, Ross. >> I'm hoping that Monty and others can improve Vorbis to start >> competing >> with AAC+ at low bitrates. Monty said he had some ideas but I wonder >> if

Apologies if this is a bit of a commercial post (I've not been lurking long), but you might like to know that the world's most listened-to online radio station has just launched an Ogg stream. (As of today!) http://www.virginradio.co.uk/thestation/listen/ogg.html (or alternatively www.virginradio.co.uk/thestation/listen from a non MSIE+PC browser). Any ideas you have to help promote this

List users, It's been a while since I've posted here, but I've been hard at work pushing toward our large scale Asterisk goal and keeping up with this list can be a full time job by itself (I have19,543 unread list messages!!). This Friday, September 16th 2005, my team will be at the MCI Development Lab in Richardson, Texas testing our setup. We have a three server system

Gautham Kasinath Software Engineer Arkin Systems Pvt Ltd T. Nagar Chennai Ph. (91) (44) 8216686 Extn 14

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've added lzo compression support for tinc 1.0pre8. Lzo is a very fast compressor (see http://www.oberhumer.com/opensource/lzo/). I've implemented it by using two new compression levels. Compression level 10 is for fast compression using lzo1x-1 algorithm. Compression level 11 is for slow compression using lzo1x-999 algorithm.

> hmmm.. very interesting.. any info on how they've done the licensing ? No idea how the licensing works. If it's free for the streamer then that may destroy Vorbis's inroads into streaming. I'm hoping that Monty and others can improve Vorbis to start competing with AAC+ at low bitrates. Monty said he had some ideas but I wonder if anything is being worked on presently.

Hi, I am experiencing some unfortunate problems when encoding WAV files to spx using version 1.0.4. A "click" which is not present in the original WAV file is added to the end of the spx file. Is this a known problem, and if so, which version of the encoder should I switch to... and if I have to switch the encoder, will I also have to switch the version of the decoder? Sincerely,

On Solaris 10 If I install using ZFS root on only one drive is there a way to add another drive as a mirror later? Sorry if this was discussed already. I searched the archives and couldn''t find the answer. Thank you.