similar to: PBX selection

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

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:

Hi Guys We are using the wizard to configure our pjsip trunk(see below) How do we get this setting to work contact_user=username We want to change the contact field in the sip invite to display the username of the trunk [trunk_defaults](!) type = wizard transport = transport-udp endpoint/allow_subscribe = no endpoint/allow = !all,g729 aor/qualify_frequency = 30

Hi, I have to connect 3 asterisk servers,each of them being TLS server for his clients and connected in both way in TLS with both others asterisk, each having hi own Common Name. Is this possible? I set up 2 asterik's , one server and the other client, this is OK. But I can't deal with certificats generated on both servers. I tried to put tlscertfile ans tlscafile in the peer

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which

Thanks for the feedback. I do agree with having multiple smaller servers. When I was first approached with this task I mentioned as much. However, the current desire is to work with already existing hardware. That is out of my hands at the moment unless it just can't be done. I will explore Freeswitch a bit soon to compare it as well. I am struggling to find what the bottle neck is in

Hello List, In our lab asterisk has crashed due to some unknown reason and it has been restarted by safe_asterisk service. But before crash we can see lots of below log entry (asterisk version 1.8.9.3). Sep 3 07:55:21] WARNING[16287] res_rtp_asterisk.c: RTP Transmission error of packet to [2002:c117:a683::c117:a683]:20940: Address family not supported by protocol chan_sip.c: Purely numeric

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

Thanks for the suggestion Tony, I installed each codec for MoH, core sounds, and extra sound packages. Unfortunately the tests produce the same results. [Sep 1 20:36:45] ERROR[10081][C-00007fe5]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x20380b0 ( continuously for a while followed by a [Sep 1 20:36:46] WARNING[7761][C-0000770d]:

${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)} I use the above command to get the system date and time it returns 20110321-034329 but it is exactly 8 hours early than the system time when I type date in linux terminal Mon Mar 21 19:43:35 HKT 2011 I am looking for help. CK -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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, 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 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 everyone, There are a number of issues with the current name of the Clang project: * It is prone to incorrect type setting, typically as CLang, CLANG, or most commonly c̦҉̫̘̺̹̖̗͒͆͋̈̃̇߯l߲҉̷̡̰̖͈̤̺͒҆̾̚͡͝a̺̹͍̔߭͠ͅn͋́͡g̱߫̉ * The C++ compiler ends in the string "g++", which causes problems for compiler wrapper scripts * The name has been used by a kickstarter project

Hello all, I need a help. I am modeling a disease and a create a R function like that: Lambda<-function (x,date1,r,h,a){ ndate1 <- as.Date(date1, "%d/%m/%Y") t1 <- as.numeric(ndate1) x[order(x$i),] t <-x[,"t"] i <-x[,"i"] CONTAGIEUX <-x[,"CONTAGIEUX"] while ( t1 < min(t) ){ for (i in 1:length(i) ){ {for (j in

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]

hello all! i am about to deploy an app that needs to connect to different schemas... uhm. let me explain. the app is located on a server (of course), and on the same server is a database with many schemas called new_york, cicago, barcelona, hongkong, singapore and so on. people from this citys need to use the app, and the app should connect to the correct schema depending from where they

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