search for: evariste

Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work

...to have all the desired prerequisites in place, but Asterisk never seems to compile with MeetMe() application support enabled, nor does there appear to be a module I am failing to load that would contain this application. Is there something really obvious I am missing? Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

Dear mailing list, I have a C function that gives me a wrong result when I run it under Windows 7. This is the code under Linux (RHEL5): > library(phenoTest) > data(epheno) > sign <- sample(featureNames(epheno))[1:20] > score <- getFc(epheno)[,1] > head(score) 1007_s_at 1053_at 117_at 121_at 1255_g_at 1294_at -1.183019 1.113544 1.186186 -1.034779 -1.044456

...active in Asterisk-user > mail list, and would like to offer you buy signature > in your messages for some monthly price. > > Is it interested for you? > > -- > Thanks, > Pavel Ismailov > skype: pavel.ismailov > www.flashphoner.com > -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u

So, is there anyone out there that provides rather generic but comprehensive CNAM-style directory services via SIP, to end-users? So I can put names to my calling numbers? Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

I have a customer installation with an Adtran DS3 mux. The DS1's go into my Asterisk servers that run IVR/Call recorders. The DS3 provider is Qwest, and they tell me that they routinely drop the DS3 service to redundant back-up's and that this is a common practice that happens thousands of times to DS3 lines daily across the US without any service interruptions. They say that the

http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/ FUD? Interesting? Boring? New news? Old news? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

hello all, i have 2 asterisk boxes on that 1 have public IP Address and another is only have local IP address now on public IP there are some 7 DID forwarded , now i want to forward 3 DID out of 7 DID to local machine we called server B , I know there are DIal , and Switch statement in asterisk , but is there any other convenient way to do this. because if call ratio is high then my call legs

Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus

This is a global message to all to announce our callcenter / dialer / predictive dialer / vicidial program is now open. Codecs: G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike

...a static IP (host=) endpoint defined. Then, Asterisk can accept registrations from your users. Where to route the call is determined entirely in the dial plan (extensions.conf), where you can send calls to particular SIP peers. So, for example, here is a regular user defined in sip.conf: [Alex_Evariste_2] type=friend host=dynamic canreinvite=no username=Alex_Evariste_2 secret=xxxxxx nat=yes allow=ulaw qualify=yes mailbox=1000 at evariste context=default-user-dial And here is a dedicated trunk to a provider: [my_sip_provider] host=xxx.yyy.zzz.www insecure=very type=peer qualify=no canreinvit...

I made a little write-up that attempts to synthesise a lot of the information out there about how to get HylaFAX working with Asterisk by way of IAXmodem for inbound faxing: http://blog.evaristesys.com/?p=24 Of course, there are bound to be some things I've left out or are grossly in need of correction. So, before I link it off the voip-wiki I am extremely eager to solicit the input of the community. If you get a chance and take a look, I would appreciate it. Thanks! -- Alex Bala...

Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

Just an FYI for Digium. I received a mailing today from you guys which was nice. The price of mailing was ~$1.60 and inside was an inflatable beach ball. Cool, but I tried to blow up the beach ball and the the seam where the part opens to inflate the ball was not connected to the ball whatsoever, so it went right in the trash. I wonder if the sick heat had anything to do with it, was mine just

Does anybody know if SetCallerPres works on calls via SIP through a Cisco gateway? We are trying to mark outbound calls as anonymous and we set it to prohib, but calls still show outbound callerid. We are SIP from * to the Cisco gateway and then PRI outbound. If we strip the callerid num, then the first number on the PRI gets added as teh callerid, so we can't do that. We need to make

...the dial plan wouldn't work because > the far-end switch would simply pass that onto the subscriber, rather > interpreting it to mean that the B channel is unavailable and it should > go on to other T1s in the trunk group. > > Any ideas? > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : +1-678-954-0670 > Direct : +1-678-954-0671 > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE...

...address in the SIP binding, and expects that's what the Contact: reachability information will be too. Any way to overcome this in Asterisk? I thought about the externip= option but it did not seem to work from an internal test box that is not behind NAT. Thanks, -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

I have to connect a Meridian S1 to Asterisk for a slow migration to VoIP. What is the best way to connect them? 1. Is a T1 the best solution? 2. Can I pass Caller and Callee information across the link? If a T1 is best, I recall a standalone T1-SIP device a year ago on this list. Does anyone recall what the device was? How well it works? Thanks, MD -------------- next part --------------