similar to: voip provider settings problem, please help

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

What is CFU and CFNR? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michel Koenen > Sent: Tuesday, August 30, 2005 1:46 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] How to use * and # as part of > numberindialcommand > > > From: "Damon

Hello List Next question where google did not spit out an unsable answer. When redirecting a call with Transfer, I would like to correctly indicate the reason. I did try this: exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)}) exten => XX,n,Dial(SIP/ZZ) exten => XX,n,set(REDIRECTING(reason)=cfb) exten => XX,n,Transfer(SIP/YY) I did try with 'reason'

Dear All, I can not transfer call in progress. What's wrong with my macro? I think tT flags is enough right? [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key exten => s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key exten => s,3,GotoIf($["${temp}" = ""]?5) ; If not existing, goto priority 5 exten =>

Hello my friends, I've a doubt, i want to be able to forward the incoming calls from PSTN to my cell phone...i mean, qhen i'm out of the office i need like aq macro that helps me to forward the incoming call that goes for example to my internal extension SIP 207, i 've this macro but i can make it work properly....i can't activate the forward in the phone, is quite confuse:

Hi chan_capi users, this thing is awesome, no delays like in modem_i4l! Plus, it got those nice ISDN features. Here's my question: Does my service provider (Deutsche Telekom) have to provide me with these Services (CD, ECT)? (the Readme in 0.2.5 says "does not relay on service CD") I know, that I don't have CFU,CFNR,CFBS (which I would have to order seperately). How likely

I am still getting these in my maillog: Jul 13 07:39:38 pop5.cfu.net dovecot: IMAP(breu at cfu.net): Corrupted index cache file /var/spool/mail/filer/storage//cfu.net/b/r/breu/dovecot.index.cache: invalid record size Jul 13 09:35:03 pop5.cfu.net dovecot: IMAP(breu at cfu.net): Corrupted index cache file /var/spool/mail/filer/storage//cfu.net/b/r/breu/dovecot.index.cache: invalid record size

I have a lattice plot that has 4 pages with 4 columns and 8 rows per page. I wish to have the rows use a separate x-axis since their ranges are quite different, but I wish to have those same limits used on each page. By setting an element of the scales list to something like x = list(limits = lim.list$CFU, lim.list$CFU, lim.list$CFU, lim.list$CFU,

Is it possible using asterisk to allow someone to dial in and remotely change where their call is forwarded to? For example, I'm working from home so I want my calls to go to 555 1234, now I need to go out for a bit so I'd like to phone the office and using DTMF tell the asterisk PBX to now forward my calls to my cell phone 555 3456 Has anyone implimented anything like this? R.

I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip

Hello everybody, First a big thanks for tinc-vpn I am still using it next to wireguard and openvpn. I am having a setup where the tinc debian appliance is at 100% cpu load doing about 7.5MB/s. Compression = 9 PMTU = 1400 PMTUDiscovery = yes Cipher = aes-128-cbc How can I pick a cipher that is the fasted for my CPU and don't create a CPU bottleneck at 100%. Kind regards, Jelle de Jong

Hi The only way to have CD service is using: DAHDISendCallreroutingFacility(<destination-5551212>, <original-my-number>, cfu|cfb|cfnr|unknown) or it is possible also with Dial() command before answering the call also ? Regards Babak -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have just installed a Sirrux PC140 card for the first time. Managed to build the drivers and get * to load them on FC4, but it does not work. It seems that layer1 in the ISDN is not even activated. The same ISDN lines connected to a Samsung DCS works so it is not the lines. I am including my sirrix.conf and the output of some of the * Srx commands below. Any pointers would be

For some reason, using the qcc package, I'm unable to use the plotmath notation in the title. Can anyone see what I'm doing wrong? library(qcc) a <- rnorm(100) qcc(a,type="xbar.one",title=expression(bar(X)),ylab=expression(CFU/ft^3) ) This seems to not let the expression be evaluated, so I tried:

Michel Send me the same output for a dial string that only sends the *31* Is this an ISDN line? What type of card/signalling/switchtype are you using? It looks as if the PSTN switch accepts the *31* and then hangs up so you can make the NEXT call with the *31* feature enabled. If so I assume the *31* feature will be enabled for the next call on the ENTIRE SPAN if it is an ISDN trunk group. If

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hello All, Upgraded a test machine this morning from beta9 to rc1 and lost access to messages and folders. an strace on the imap process shows it accessing the home directory, but not the mail directory. The two queries that I use are: password_query = SELECT email,crypt as password FROM users WHERE email = '%u' user_query = SELECT homedir AS home, maildir AS mail, 1000 AS uid, 1000

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

Yo all, As there has been some intrest, here's my updated version: I post it to "-dev" as well as "-users", as it may be of intrest to both. Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate set of features. Currently, my implementation supports call- forward unconditional, on no answer

All, Is anyone using iptables (recent module), or any other alternatives, to throttle the number of new imap or pop connections per minute? We have some applications that like to login every second to pull mail using imap, so we'd like to protect the entire dovecot server from these applications. We've already made the change over to high-perf mode, but we still need some type of denial