search for: freeswitch

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

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 ---------...

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.con...

...heir project. They feel that * has improved since the fork but they still have the same complaints: the Asterisk core is a "pig," the core is huge and messy with ugly code, and finally that Asterisk development is throttled by the "Digistapo." Some there have suggested that Freeswitch is going to be the wave of the future. They hyped it up pretty well. However, after visiting the Freeswitch site and signing up to get the source and docs, I'm not all that impressed. It's a great idea but it is months away from being a truly useful product. My questions to the Aster...

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 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: http://lists.xiph.org/pipermail/icecast/attachments/2...

...t at xiph.org > > You can reach the person managing the list at > icecast-owner at xiph.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Icecast digest..." > > > Today's Topics: > > 1. Using freeswitch and Icecast (Jorge N??ez) > 2. Re: Using freeswitch and Icecast (Basil Mohamed Gohar) > 3. Re: Using freeswitch and Icecast (Thomas B. R?cker) > 4. Re: Using freeswitch and Icecast (TheDarkener) > > > ----------------------------------------------------------------------...

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 from 20 to 12 seconds. Secondly, t...

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 --...

...172388] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [ 240.172610] INFO: task flush-202:0:421 blocked for more than 120 seconds. [ 240.172619] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [ 240.172946] INFO: task freeswitch:1030 blocked for more than 120 seconds. [ 240.172957] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [ 240.173215] INFO: task freeswitch:1033 blocked for more than 120 seconds. [ 240.173225] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" d...

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 value. In the Invite, the following line is essentially identical with the exception of the screen=no. Freeswi...

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 >> >> Jorge > Jorge, first I'd like to know what you did to reduce the delay from 20 &...

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

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:

...ect 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 recommended it to us) B: Commercial 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now acquired by a HongKong company now 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It also includes VoIP SDK, WebRTC and offer rebranding app f...

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 not look feasible ? Thanks for any input you may...

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

...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 running include: LibreELEC https://libreelec.tv/ Vortexbox https://wiki.vortexbox.org/available_images ownCloud https://owncloud.org/download/#owncloud-server-appliance freeSwitch https://freeswitch.org/pages/vm.html Etc... Without wanting to start a very competative discussion - what would be a suggested minimal centos 7 install to provide such a server? My main area of confusion is whether I need a gui on the server for kodi? Many thanks for any answers to this wildly o...