similar to: Using freeswitch and Icecast

what-he-said On 08/07/2013 06:48 AM, Basil Mohamed Gohar wrote: > On 08/06/2013 07:40 PM, Jorge N??ez wrote: >> Hi I am trying to use icecast to broadcast a realtime conference from >> freeswitch. But I am having a delay like 20 seconds then I reduced it to >> 12s. But I don't know if somebody can help me how to reduce it as lower >> as possible. >> >>

Thanks for your answer, well I changed this parameters on icecast.xml and the the delay reduce from 20s to 12s <burst-on-connect>0</burst-on-connect> <burst-size>4096</burst-size> Well I was trying to reproduce mp3 and ogg but both have 12 s of delay. How can I reduce to maybe 1 or 2 seconds. 2013/8/7 <icecast-request at xiph.org> > Send

Thanks for your answer, well I changed this parameters on icecast.xml and the the delay reduce from 20s to 12s <burst-on-connect>0</burst-on-connect> <burst-size>4096</burst-size> Well I was trying to reproduce mp3 and ogg but both have 12 s of delay. How can I reduce to maybe 1 or 2 seconds. -------------- next part -------------- An HTML attachment was

On 08/06/2013 07:40 PM, Jorge N??ez wrote: > Hi I am trying to use icecast to broadcast a realtime conference from > freeswitch. But I am having a delay like 20 seconds then I reduced it to > 12s. But I don't know if somebody can help me how to reduce it as lower > as possible. > > Thanks > > Jorge Jorge, first I'd like to know what you did to reduce the delay

what is the difference between FreeSwitch and Asterisk , whitch one is more scalable and reliable? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

This is slightly OT in that it isn't specifically *-related, but I was wondering what the members of the * user community felt about these two subjects. I've been perusing the OpenPBX.org mail list and the current hot topic is the fact that their project has come to a grinding halt. They are concerned that they don't have enough people working on their project. They feel that * has

Hi I am trying to deploy freeswitch with Digium TE121 card for my office setup, but it is continuously showing Signaling is up and channels are down except D channel. Our Architecture is like We have freeswitch installed with libpri1.4 and Dahdi. I am from India and here we are having E1 trunk. Dahdi Configuration is cat system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 7

We are working to develop a web based IVR Designer that will work with Asterisk as well as Freeswitch using Raphaejs library, Click following link for detail http://sourcecodemania.com/ivr-designer-using-raphaeljs-for-asterisk/ Looking for your valuable suggestions Regards Nasir Iqbal ICTBroadcast SMS, Fax and Voice broadcasting solution http://www.ictbroadcast.com/ -------------- next part

The only thing that has changed is the winamp version number. My radio streams are unaltered (to the point of the same process connecting to two versions of Winamp). What happens is that I begin to hear skips in the stream. Soon the stream is full on stuttering, as if a square wave was modulating the volume control. Anyone else experience this? ------ Dave Hayes - Consultant - Altadena CA, USA

Hello, One of our client company is providing hosted contact center solutions with Cisco IPCC. To keep the Call Recording cost at low, they are planning to use Asterisk / Free Switch. Can anyone integrate Cisco IPCC with Asterisk for call recording ? Regards, Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email:

Hi list, I'm trying to create an asterisk 1.8 rpm with SRTP. I found mention of a libsrtp rpm, <http://qutecom.ipex.cz/RPMS/srtp-1.4.4-1.i386.rpm > in these instructions, <http://www.voip-info.org/wiki/view/Asterisk+SRTP> but it is unreachable (by me, anyway). The libSRTP source is here, <http://srtp.sourceforge.net/download.html>. Has this already been packaged for

Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. Regards

Hi, Please excuse my ignorance but I'm having trouble with a very basic matter: I'm using vorbisfile to stream audio to the waveout device in Win32 (using waveOutWrite). I'm basically reading packets from an ogg file and streaming them using a simple buffering scheme. The thing is, this works great when the bitrate is more or less constant but the audio sounds garbled if there is a

Hi all, Today I got problem below and my domU become unresponsive and I should restart the pc to make it running properly again. [ 240.172092] INFO: task kworker/u:0:5 blocked for more than 120 seconds. [ 240.172110] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [ 240.172376] INFO: task jbd2/xvda1-8:153 blocked for more than 120 seconds. [ 240.172388]

Hi all, I'm new to VoIP, now we have a project that needs a PBX with client APPs. In our team we have argument for choosing PBX. By so far, we have following candidates: A: Open source 1) Asterisk PBX (http://www.asterisk.org) (with longest history that almost every one knows it, now the last version using the PJSIP stack) 2) FreeSwitch (http://www.freeswitch.org) (A lot people

Hi Please check out this product http://www.sangoma.com/products/hardware_products/data_networking/a301.html Does it work on Asterisk or Freeswitch ? Do Telcos provide an E3 connection ? One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does

I am using AMI to Originate a call. I have been able to get the caller id number to be passed through. However, I can't get the name to be passed through. A person I'm working with has a Freeswitch that is able to pass the caller id name and number through for their call. Comparing the Asterisk SIP messages to the Freeswitch SIP messages, I have narrowed the problem down to a single

All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and

Hello folks, I'm running 1.8.11 on a Centos 6 system with an adjacent Hylafax server using softmodems: Noticed this in the Asterisk log when trying to send a fax from Hylafax to Asterisk: [Apr 3 01:53:09] WARNING[29184]: chan_sip.c:8926 process_sdp: Multiple audio streams are not supported I've googled a few asterisk tickets that may suggest that yes, multiple audio streams are not