similar to: Calling to users in other asterisk servers

Hi How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113

Hi, Plateform: gentoo + qmail My deliver line is in a .qmail-USER file: |/var/qmail/bin/preline /usr/libexec/dovecot/deliver -f ${SENDER:-<>} -d user at domain.tld I notice that sieve implementation in Dovecot use double ">" for the envelop sender when there is a redirect command in sieve script. (i've tried preline -f, without -f ${SENDER:-<>},... but nothing

Hello all, I haven't been able to solve an issue I am having when copying over data from a Windows box to a Samba share. I have found others with the same issue and I have made some configuration changes to try and solve the issue but the issue is still lingering. I want to migrate off my old Windows box but until I solve this issue I am I cannot move forward. Please let me know if

Aaron, Yes, I left the server string out intentionally because of how the mapped drive description showed in Windows. Could this be causing my problem? I was able to join this server to the domain and I could retrieive a list of users using the winfo command. Thanks! Will -----Original Message----- From: Aaron Kincer [mailto:kincera@gmail.com] Sent: Tuesday, May 29, 2007 12:21 PM To: Will

Hello, i want to use asterisk as a gateway for H323-Phones. But i cant get it work. I'm using a gatekeeper on another computer. My IP-phone is registered there. Does anybody can sent me an oh323.conf and extension.conf as examples? Thanks in advance Erik Bastian -- NEU : GMX Internet.FreeDSL Ab sofort DSL-Tarif ohne Grundgebühr: http://www.gmx.net/dsl

Hi all, i''m new at LARTC, and after reading the docs I found no solution to my problem ... On one side I have eth0 conected to the LAN, on the other side I have eth1 conected to a switch and to 3 DSL routers with 3 diferent providers, and also eth2 conected to a cisco 2600 conected to a LDMS line. I have readed the larct docs about multihomed conections to internet, but I''m

I have an asterisk server behind NAT - no audio on the test external calls I have tried making so far. Read http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - No solution evident from there, sounds like I have case 9. I would have thought that all I would have to do is port foward and have the external IP on the asterisk server, which I have done I have fowared 5060UDP, 8000UDP, and

Hello list, I have an Asterisk setup with the following details: 1. 3 x internal extensions / sip hardphones - Grandstream 2000 2. 2 x internal extensions / dahdi cordless phone 3. 1 x 2 FSX ports OpenVOX pci card 4. 1 x internal sip extension / sip softphone (linphone) 5. 1 x 800Mhz Asterisk + Linux server 6. Asterisk version is 1.6.2.13 7. 1 x IAX2 incoming trunk from phone provider for 1

Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan,

Somone in europe have had succes getting Callir ID showed on a phone screen conected to an Handytone 286 ? Adri? Vidal -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 235 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041124/e5514052/attachment.bin

I have a E1 conected to asterisk all zap channels are ok, but when calls come into Asterisk caller don't hear none ring, the call goes straight into the menu, how can i simulate 2 or 3 rings? here it is my conf. exten => s,1,Answer exten => s,2,Wait,2 exten => s,3,NoOp(${CALLERID}) exten => s,4,ResponseTimeout,45 exten => s,5,DigitTimeout,3 exten =>

On 28 Mar 2019, at 10.15, Arkadiusz Mi?kiewicz <arekm at maven.pl> wrote: > > error = 0x55e3e2b40ac0 "Fixed index file > /var/mail/piast_efaktury/dovecot.index: log_file_seq 13 -> 15", > nodiskspace = true, This was one of the things I was first wondering, but I'm not sure why it's not logging an error. Anyway, you're using filesystem quota? And this

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only

I''m trying to specify that an action should be rendered with a given layout one particular spec. What I''ve got at the moment is this. it "should render with the grabber layout" do controller.should_receive( :render ).with( :layout => "my_layout" ) do_get end This doesnt work even though this call to render is being executed. render :layout

Hi Users NUT, I want monitor a UPS of brand SMS (Sinus Double 8 KVA) using a raspberry-pi. In compatibility list, is listed to use the blazer_ser driver. I use a USB adapter to RS-232 conected in to the No-Breake. Follow the comands e confs. root at rasp:/home/pi# lsusb *Bus 001 Device 004: ID 067b:2303 Prolific Technology, Inc. PL2303 Serial Port* Bus 001 Device 003: ID 0424:ec00 Standard

Hello R developers. I encountered the same problem as Uwe Ligges with as.matrix.data.frame() in bug reports 3229 and 3242 - under section not-reproducible. Example I have is: > tmp level 2100-D 1 biological_process unknown NA 2 cellular process -5.88 3 development -8.42 4 physiological process -6.55 5

hello, i cant find a solution on this (might be) easy problem: i have a time serie by carlandar weeks, so for every carlendar week i have a value. now i would like to use the functions for time series, so i change structur to a time serie with cam <- ts(number,start=c(2001,1),deltat=7/365) or cam <- ts(number,start=c(2001,1),frequency=52) the problem on it is, that 2004 had 53 calendar

I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message

Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I

Hello All, Does anyone know why length(fit1$time) < length(fit2$n) in survfit.coxph output? Why is the predicted time length is not the same as the number of samples (n)? I tried: example(survfit.coxph). Thanks, parmee > fit2$n [1] 241 > fit2$time [1] 0 31 32 60 61 152 153 174 273 277 362 365 499 517 518 547 [17] 566 638 700 760 791