similar to: Bandwidth Consuming

Hi Oddsock, Like Clement, I am sure Nullsoft is still "offering" AOL's bandwidth since I think Nullsoft is not part of AOL anymore. About the new broadcasting methods, is the multicast technology already available? I have heard only few providers are equipped with multicast enabled routers. What about p2p streaming, is it really reliable? When I see Peercast's statistics, only

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

Dear, Our iax clients's ip and port in the database were removed automatically, after 30 secs. the iax info is saved in odbc and postgresql . asterisk=# select * from iax_buddies where username='9706015'; name | username | type | secret | md5secret | dbsecret | transfer | inkeys | outkeys | auth | accountcode | amaflags | callerid | context | defaultip | host | language

hello is any web based sip phone? for example: a user after logining in, view a configured sip phone, and ...... best MAni ____________________________________________________________________________________ Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains. http://farechase.yahoo.com/promo-generic-14795097

Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 | 2 | hangup | | 455 | DID | 14193016880 | 1 | Dial | H323/1169#989181310524@66.152.61.66|60 | didx.org for

Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.

hi After compiling and installing pwlib and openh323 , the asterisk, give the folloing error. please tell me where the problem is ? Best Mani *CLI> -- Executing Dial("SIP/2.2.2.2-086f5ac0", "H323/652#150388590962@1.1.1.1|60") in new stack Mar 28 14:17:23 WARNING[11985]: channel.c:2576 ast_request: No translator path exists for channel type H323 (native 4) to 256 Mar 28

Hi Everybody, It works with Peercast because Peercast receives the headers from Icecast2 and sends its own headers back to the client (Winamp, WMP, Real...). Remember that Peercast is a client AND a server... I guess Peercast doesn't send back the headers in the same way than Icecast2. Maybe Peercast doesn't use ICY/X-audiocast. Talking about Real (v9.0 and 10), I already got the

At 08:30 PM 4/7/2004, you wrote: >On Thursday 08 April 2004 01:40, Allan wrote: > > Hi > > > > Can anyone tell me how to get Metadata (Artist & Title) to display in WMP > > and RealOne > > > > It seems to work in WinAMP5 and if a rebroadcast the stream via PeerCast > > > > > > Im using SAM2 and Icecast2 > > > >They should

Hi Stephan, I am talking about the meta-data: Artist - Song. It's weird because you are the second person on the list who doesn't get it working in Real... I just tried again with "BEK_dns" and "Radio Alcanzando Naciones" the 2 first MP3 radios I found in the stream directory and everything worked perfectly for Real (but not for WMP). In RealPlayer, I clicked

Hello :-) I had a look at the servers that can stream Theora. I saw that ffmpeg was greatly appreciated. What's your opinion ? Is there any chance that IceCast can do it (What source would be used? Ices?) ? I really want to support Xiph products. I saw that page http://www.linuxmafia.com/faq/Apps/AV/video-streaming-servers.html but it's a bit old now, may be things have changed :-) Well,

Hi Can anyone tell me how to get Metadata (Artist & Title) to display in WMP and RealOne It seems to work in WinAMP5 and if a rebroadcast the stream via PeerCast <p>Im using SAM2 and Icecast2 Allan --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to

Dear list, I agree with the main points that have been stated here lately -- icecast2 appears not ready for prime time (because of lack of web presence / tarball and rpm downloads / some documentation on line) and icecast1 community needs a support forum if this list no longer is one. It would be so simple to spell out the statuses on www.icecast.org -- Status of icecast1, status of

Dear for having an stable system which limit option is good for ulimit comand ? 2-is any option for making asterisk crash-free? Best -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110810/365d9d56/attachment.htm>

oddcast DSP is what you want, it supports live input (called Advanced Recording) and does support streaming to Peercast...It currently does not support Vorbis metadata for live recording (since metadata for vorbis streaming is inserted into the ogg stream, as opposed to how it's done for mp3 in which the metadata is updated on the server via URL calls)... anyway, it's the closest thing to

Hi, I've had a few problems with broadcasting an icecast2 playlist with peercast. The problem is that when getting the m3u file, the response isn't terminiated with a newline (\n or \r\n). peercast doesn't particularily like this. I've submitted a patch to peercast for this, but I wonder if icecast2 should add the \r\n to the end. Maybe there's some more buggy software around

OK if I go with say HE-AAC or Opus, can I still encode as mp3? I am using Butt as my source encoding. Can Butt be configured to encode using these codecs instead of their codecs its defaulted to? Still a bit unclear on the method to calculate a 128k stream with 6,000 GB available bandwidth over 30 days to give an estimated average user connections possible. Mit freundlichen Gr??en Scott

Dear, because of using dial(local/...) each incoming calls (_12X.) makes 4 ports on asterisk. I can not use goto , because of some limitations. is any way to decrease it? Best, [MAIN] exten => _12X.,Dial(LOCAL/${EXTEN}@TEST/n,60) .... [TEST] exten _X.,1,Dial(${EXTEN}@next_gateway,60)

Hello Nicolas, Sunday, April 6, 2003, 2:42:36 PM, you wrote: NB> Hello EISELE, NB> Sunday, April 6, 2003, 11:32:59 AM, you wrote: EP>> Hi, EP>> I'm CTO for a new Techno/Trance Radio named Let's Go Zik EP>> (http://www.letsgozik.com) and i'm looking for the cheapest solution to EP>> broadcast. For the moment, we're using Live356 for relaying our