similar to: Difference between Asterisk and FreeSwitch

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, 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:

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. >> >>

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

Sorry. I forgotten the words volume or loudness. But it is know as microphone stroke too, i think. If something can tell me something about that procedure it would complete my pleasure. To bring back memories, i only wanted to know wheather i can change a variable that holds the sound intensity (loudness) needet to start "encoding >> sending" if the speex codec is in voice

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, I'm currently working on a server whitch use samba and openldap, The OS used is Debian squeeze 6.0.1 64 on the server, the previous was fedora 5 My Samba is the domain Master of the network, the users of the ldap are link with the samba, and i try to join computer XP to this domain, so the user present in the ldap could (with login and password) log on in the domain, access shares

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 mounted by smbmount a partition on a winNT machine without problem... but when i tried to copy a file whitch have size more than 2 Gb i have the following error : Overflow size allowed for a file (translated from french ;-) :D?bordement de la taille permise pour un fichier ) PS. the file system of the harde drive is NTFS PS. The command of smbmount : /usr/sbin/smbmount

Hello, I'm trying to sum up current options for doing small (up to 20 agents) inbound-only CC. I've found: astguiclient, maybe there are some other CC solutions? And on the other side: witch is better to have 20 PC w/ softphones or one T1 channel bank and normal phones with hands-free sets. (whitch set whould you recomend) ? kd, -- Krzysztof Drewicz Affordable 2/4 span E1/T1

Hello List, as an long time macintosh user , musican/producer/programmer , i am very upset that another great technology (DSS ) vanished because of http streaming so i turned my interest towards icecast, whitch seems an fantastic and evolved media streaming server. I am very interested in Midi, especialy the possibility to *sync Audio with Midi*. So my question , would it be possible to stream a

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

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. I have been trying to check, how llvm works on Solaris recently. First I have tested lli, whitch seems to execute the bytecode generated on Linux without any problems. However, llc has failed to generate valid SPARC assembler code even on the helloworld example. Here is the generated code: sakharov at trillian:~$ cat ./test.s .text .align 16 .globl main

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