similar to: Slightly OT: OpenPBX.org and Freeswitch

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>

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

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 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, What do you think of this project www.openpbx.org ? Something like ser and openser ! Kinds Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

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. >> >>

Hello there! During some research on Internet I found the following comparison on site Voip-Info (see, "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ"): The main points listed on Asterisk's "CONS" that concerned me were: * Conferencing on Asterisk depends on Zaptel hardware and/or kernel modules for timing; * Lack of built-in STUN support for SIP NAT

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

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

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

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]

Hallo list I've been running fedora for donks as my production laptop os, but now I want to set up one of those old laptops to run as a home server running a number of home type vm appliances. I don't want to risk having to tear down and rebuild the setup every 6 months to a year - so, figure centos is my best canditate to run as a stable server. The sort of home type vms I envisage

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 interested in getting Openpbx up on Centos. The install information is at http://wiki.openpbx.org/tiki-index.php?page=Getting+Started And it is quite a bit beyond my current skills, so I am looking for some one that has done it and can help step me through it. I want Openpbx over AsteriskNow or Trixbox to get T.38 support.

at http://wiki.openpbx.org/tiki-index.php?page=Easy+route+to+building+OpenPBX.org there is the following adduser command: adduser --no-create-home --ingroup openpbx --disabled-password --disabled-login openpbx This does NOT seem to be the right format for Centos. So far, using man, I have come up with; adduser -M -g openpbx What else do I need?

at: http://wiki.openpbx.org/tiki-index.php?page=Easy%20route%20to%20building%20OpenPBX.org A number of dependencies are listed, with a Debian slant. I have found most of these in our standard Centos repos ( snarfed spanDSP from trixbox repo). But the following I have not figured out: * linux headers for your kernel is this glibc-kernheaders? * Libssl-dev

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

Hello Experts. Im working with Asterisk PBXand freeswitch PBX. I have a challenge with FXO card in Asterisk and i could not solve it yet. I hope you could guide me in this regards. When i want route the call to FXO channels, Before the callee answer the phone (pick up phone), The channel is answered with FXO card. How can change this treat so that the callee dont answer the phone, the channel dont

Hi. I am using asterisk 1.8 and everything was working fine when I was at svn 342661. I then upgraded to vrsion 349339 and discovered the following problem -- one of the end points is a freeswitch box which offers a number of codecs, including PCMU. However, when I tried to make a call I got a 488 response and a message "multiple audio streams not supported" in the log. Is this by

> no, it only has a theoretical impact on the number of open > filehandles, theroretical means in that context surely not relevant > for a fileserver > > if you have a preforking service and expect many thousand forkers than > it may become relevant Thank you Would you give the same advice in the case of a Samba 4 AD Domain Controller?