similar to: Freeswitch with Digium T316 timed out, T316 timed out

Hello, I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as "busy". The telco has been working with their switch manufacturer and took the output of "pri show span 1" from me and came back with this: ----quote--- Please check your timers

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

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

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 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

I have a 'CONGESTION' Status with R2 protocol. While testing this scenario sip GW--?Asterisk ?Digium E1 R2 Protocol?Cisco E1 R2 protocol?sip Gw Find below my error and configuration ,where are the errors in my configuration ? ========================================================================= Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on

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

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 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:

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. 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

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

The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but

Hello, I'm researching the possibilities of multiple communication platforms like Asterisk and FreeSwitch for handling a dynamic sequence of applications to execute, like Playback, Read, etc. This only applies to originating a call from an external application by using the AMI Manager and the Originate action. I need to know the following: 1) Does the Originate action support multiple

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. >> >>

Hi Joshua If i put the default_user option per endpoint would it work?? So what exactly does the contact_user option do? I know that in freeswitch there is the option extension-in-contact.We ?basically need to achieve the same functionality? Thanks<div> </div><div> </div><!-- originalMessage --><div>-------- Original message --------</div><div>From:

Everyone, I will be attending Astricon in Phoenix and would like to have a little get together to discuss Open Source Telephony and the challenges we as developers and system integrators face. Exchange ideas and go over some use cases and see how we can all work together to improve our understanding of the dynamics of how everything works together. * Scaleability * Reusability of code

Hi All, Do you'll have any recommendations on a Linux based Customer Management and Pre-paid Billing system for Asterisk, Freeswitch or Kamalio? The system should also allow customers to register, login, buy more credit, view call records, etc. Commercial or Open-source are ok as long as they run on Linux. Thanks, A. -------------- next part -------------- An HTML attachment was scrubbed...