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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

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

I've got the ztdummy kernel module loaded and seem 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 :

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

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

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

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

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

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

Anyone have an idea how to implement a phone number that can only be called once? The first time it will process normally and any subsequent calls will be rejected.

Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim

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:

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

Hi, What are the current best practices for running asterisk as SIP B2BUA? Are there any sample configs online or the books that detail this configuration for the newbies? I'm going to run it behind 1:1 NAT for the clients in the public internet so I will use the externip, localnet, and nat settings. Thanks, Andrew

Hi, all I want to add more than 200 sip accounts into sip.conf, username from 6000 to 6199, password is the same, i remember there is a better way to do this case, however, i have not searched the method yet. Anybody can tell me this method, TIA. BR Mike Li -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6. I make an attended transfer (asterisk feature), and I cant see the event. Any idea? Should I submit a bug report? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090313/eb5a7ea0/attachment.htm

i am asked to implement a call center of 50 seats for my company , and i was wondering if Asterisk can fit this as a relaibale and low price system is it mature enough for this task?? best regards Gers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090712/b5bab36d/attachment.htm

Hi, Would just like to inquire if anyone here has a setup of asterisk to send traffic to AS5400 connected to an SS7-PRI.? this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200? TIA Regards, Nhadie --------------

Hi All, Have any way to asterisk forward the 487 Request Cancelled in SIP TO SIP call? In a SIP to SIP call when the called peer B send 487 to Asterik, Asterisk return to calling peer A 603 PEER A ASTERISK PEER B | INVITE ------------>| | |<------------TRYING| | |

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