similar to: asterisk with ser project, , , , here we go! ready or not!!!

Got an email this morning with the subject "Welcome to Gizmo Project". I didn't sign up with those yokels. Anyone else got spammed by them?

Hello! For comparing two non-normally distributed samples, Leech (2002) suggested to report nonparametric effect size indices, such as Vargha & Delaney's A or Cliff's d. I tried to search the R-sites, but could not find related procedures or packages that include nonparametric effect sizes. Thank you for your help! Citation: Leech (2002). A call for greater use of nonparametric

I've just gotten off a skype conference call and it pisses me off that the quality of skype is higher than my asterisk calls. Is there such a thing as a super high bandwidth codec? In a situation that you have the bandwidth to share is there something that I can use for important calls when the situation warrants it? TIA, Dean -------------- next part -------------- An

Hello I am using a Junghanns QuadBRI ISDN card - the module name is qozap. If I like at my interrupt assignment, qozap is sharing interrupt 10 with libata and uhci_hcd. I think libata is the IDE hard drive module and uhci_hcd is a USB module. linux:~ # modprobe qozap linux:~ # cat /proc/interrupts CPU0 0: 12634579 XT-PIC timer 1: 10 XT-PIC i8042

Hi, I know the following is mostly the issue of SER and I already posted the same content to SER User list. Just for more input, I posted it to this list. Sorry for the cross post for some people. I've set up SER for UA to UA call. I'm thinking of setting up SER to relay to Asterisk PBX to use conference call and voicemail of Asterisk. I will employ this system for client connection

Dear ALL: My scenario lists below: Assume: UA1 with sip id "1011" And dial number to PSTN is "0939749xxx" There is no modification rule at my CISCO. (It will not change any dialed number) UA1 ==> SER ==> UA2 (SIP to SIP) UA1 ==> SER ==> Asterisk ==> CISCO 5300 ==>

I have several Asterisk servers installed and one SER server which will act as a gateway to PSTN, en redirect server. I was thinking to implement it the following way: - Register all the * servers at SER (is this neccessary?) -> this works via register=>asterisk:password@serbox in sip.conf - Setup aliases in SER for the telephonenumbers to the appropiate * server: serctl alias add

hello all, i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the requests without a problem. When I updated to 1.23 requests from SER to asterisk die quietly, no

Hey gang, I'm trying to work out all possible scenarios using SER & Asterisk in our upcomming deployment. The example scenario is 50 different customers, all with different numbers of SIP UAs. All UAs would register with SER; This will help keep any inter-office conversations off our bandwidth since SER doesn't handle the RTP stream. Calls from PSTN to UA are easy to handle.

Hi everybody, I am trying to make up call flow diagrams for for a setup which include ser as a sip proxy/registrar and asteriks as a voicemail server. Is my sequence correct?: UA 1 send an invite to SER. SER forwards this invite to UA2. UA2 sends back a sends back a 100 trying and 180 ringing message. SER forwards these. However UA2 doesnt answer the phone,so what happens then?...is there a

How can I install a softphone on my USB flash drive like Xlite and have it ready to go when I plug it in at any Windows XP computer? (Same for a Linux softphone, both on one USB flash drive). bye Ronald Wiplinger

Hi, I'm trying to transfer an incoming call from the PSTN to another PSTN number through a SER - Asterisk system. SER doing only routing.. pstn call-> SER -> asterisk (call forward) -> SER -> pstn Logic for SER: If something comes from the pstn, send it to asterisk. If something comes from asterisk, send it to the pstn. Every time I am getting a "Got SIP response 481

Hello i am trying to use this exmple with SER-0.9.3 but still NATED Clients are not working any other requirement http://www.voip-info.org/tiki-index.php?page=SER+example+NAThelper ----------------------------------------------------------- # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters

Hi all I am in this strange situation: we had ser configured to relay calls to numbers to asterisk extensions and all used to work nicely, with both ser and asterisk running on the same machine with public ip (ser on port 5060 and * on 5061). We had to move temporarily our server to another provider which put our server on a dmz, so that now we have our server with private ip but reachable from

Heya, I'm trying to get SER up and running as a front-end for a couple of Asterisk boxes for SIP clients. I'd like clients to register with the SER platform. However, I'd like clients to authenticate with Asterisk when they try to make outgoing calls via Asterisk. Otherwise it seems that users could bypass my SER box and register directly with the Asterisk boxes and bypass

Hello all! I googled lists.digium.com and ser mailing list, but did not find any working configuration of asterisk used as voicemail for SER. This is my config if (uri==myself) { if (method=="REGISTER") { save("location"); log (1, "Registered\n"); break; };

Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original

I have a really simple question about a fairly complex problem: I have a Cisco 7960 behind a NAT. I have an Asterisk server behind a different NAT. I have a SER server (with rtpproxy installed) on a public IP adress. I've opened ports with static NAT to * and the Cisco. Without using SER, I can register the phone to *, I can complete calls, I just can't move audio. Reading the

Hi everybody, I have a SER Server (Sip Proxy / REGISTRAR) and a Asterisk Server (PSTN and other services). I've got some clients that make calls to each other through my SER Server, that's to say, non external or international calls. I would like my clients to make external and international calls through my server but for that they must authenticate at another server to have a valid

Hello, I have several * servers behind a SER server (in a local ip range). The SER server is also publicy reachable. On the other site, I have SIP clients that are behind another NAT or in the same NAT range as the * server. Can someone give me some directions/hints etc. on how to make this work. I think I should be using MediaProxy with SER. But do the SIP clients need to register at the SER