search for: x2n

...rmittently and some users on every call. Note all the PAP2-NAs are running the same and latest firmware. Am I missing something completely obvious? Is there a way to see why Asterisk is sending 488 (i.e. what is not acceptable?). sip debug peer and sip.conf is below: Sip read: INVITE sip:800@sip.x2n.net SIP/2.0 v: SIP/2.0/UDP 216.252.155.227:11288;branch=z9hG4bK-8079155a;rport f: Bode Dar <sip:3255@sip.x2n.net>;tag=faf9295d7d8c85e6o0 t: <sip:800@sip.x2n.net> i: 1db23be9-a4adb4ca@192.168.2.3 CSeq: 103 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="3255",real...

...advice ll1<-function(rho,theta,beta1,beta2,beta3,beta4,t,Szenariosw5,Testfaellew5,X1,X2) { n<-length(t) t<-cumsum(t) tn<-t[length(t)] Szenn<-Szenariosw5[length(Szenariosw5)] Testn<-Testfaellew5[length(Testfaellew5)] X1n<-X1[length(X1)] X2n<-X2[length(X2)] n/rho-(tn^theta)*(beta1*Szenn+beta2*Testn+beta3*X1n+beta4*X2n) } p0<-c(rho=1,theta=1,beta1=1,beta2=1,beta3=1,beta4=1) optim(p0,fn=ll1,t=t,Szenariosw5=Szenariosw5,Testfaellew5=Testfaellew5,X1=X1,X2=X2) i always got an error message: Argument "theta" is missing,...

Hello All, I am looking for a sip phone that is capable of automatic nat. The Cisco ata186 for example works fine for natting with iconnecthere, but as for asterisk, both my 7960 and polycom ip600 require you to set the nat ip on the tftp. Does anyone know a good phone (or ata) that can do this automatically? For example, I want to give a phone to my brother, who is going to europe. His ICH

...o voicemail. Any ideas on what to do? The only thing I came up with is to also create local voicemail boxes that email the voicemail to the user when the 'net connection comes back. I was wondering if there were any more creative ideas. Thanks for reading. -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990

Guys. What do I need to configure in order to let my Asterisk receive calls from sip phones, etc not registered with my server on my extension? For example, let people use their asterisks or sip phones to call blah111@server.com?

Hi all, I need all your help on this. I have the next part of code:............e1=x1-mean(x1)e2=x2-mean(x2)n1=length(x1)n2=length(x2N=(n1 + n2)nu2=sum( c( ( x1 -mean(x1) )^2 , ( x2-mean(x2) )^2 ) )/Nss=c(e1,e2) b3= N*sum(ss^4)/ (sum( ss^2)^2)............what do lines 6-8 (mathematical notation)?Also the, what means part?............for(j in 1:B){ ss11=sample(x1, n1, replace=TRUE) ss12=sample(x2, n2, replace=TRUE) ............}....

Greetings All, I have a number of projects in the works at the moment and for one of them, I need to locate an inexpensive and reliable service that can provide small-office virtual services: 1. FAX to Email 2. Toll Free number with voicemail boxes for Tech Support, Billing Inquiries, Customer Service, Abuse Reporting, etc... I have been looking all over the Internet and there seem to be a LOT

I have seen on eBay adapters for Cisco 7940/7960 phones, to use cell phone headsets. They were about 12-15$. I think original manufacturere was at http://www.ciscoheadsetadapter.com. >Started a Wiki page here: > > http://www.voip-info.org/wiki-Cisco+Phone+Headsets > > >Jim > >James H. Thompson >jht at lava.net > > ----- Original Message ----- > From:

Can anyone please help with an astcc problem. I just got it going, but "billed cost" stays 0. The test route is setup with "Inc. Seconds" = 6 and "Cost per additional minute" = 10000. What can the problem be? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.16 - Release Date: 24/05/2005

...NTCODE}) and the CLI shows: -- Executing NoOp("SIP/x.x.x.x-0821e058", "") in new stack indicating a blank AccountCode. I tried username-based authentication and have the exact same issue. Anyone have any ideas on why this is happening? -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990

I have been hearing a lot about the new Gnet SIP phones. Is anyone using them? How do they perform? Sean

Is it possible to terminate (or forward) lets say 800 DID number to FWD number. -- #Joseph

I've got a project where I need to sell a voip QOS product from Australia to US resellers. I don't suppose anyone here knows where I can find a list of a whole heap of US resellers do you in either VOIP or IP space? Regards, Dean Collins Cognation Pty Ltd dean@cognation.net +1-212-203-4357 +61-2-8307-3503 (Sydney in-dial) -------------- next part -------------- An

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

Is there any SIP or IAX phones that can be configure in french instead of english. I tested Cisco 7960 phones but I can't change the language it's only available in english with the SIP firmware. I have a customer that's located in France and he wants french phones if possible. So I'm wondering if there's any one out there that found a phone that can be change to

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 2775 bytes Desc: image001.gif Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070527/cb333f7e/attachment.gif

I'm putting together an Asterisk server for a local charity to use as an announcement system. I've been thinking about how to write the dialplan to allow different options for different groups' announcements, as well as mailboxes for the various groups and the charity's administrators. Of course, this would also need to include an option for the heads of the different groups to

Would anyone mind emailing me, or maybe posting somewhere their music on hold .so file? thx -ben

Guys. Anybody using sipuras 2002 knows if there is a way to make the phones connected to it to autodial an extension when the phone is picked up? For example, if the phone is on a police booth (building entrance) and you want the guys to just pick up the phone and make the phone auto dial the receptionist extension without the guys having to dial anything (ala batphone). Is this possible with

Hi all, My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote an AGI script that does the following: 1) If it's a toll free number (800|888|877|866), set the CallerID name to "TollFree Caller" 2) Use curl to look up the number in Google phonebook 3) If a business listing, set the CallerID name to business name, as is. 4) If it's a residential