similar to: voicemailmain

Is there a way to combine these lines into one? exten => s,2,GotoIf($["${CALLERIDNUM:0:3}" = "800"]?s|108) exten => s,3,GotoIf($["${CALLERIDNUM:0:3}" = "866"]?s|108) exten => s,4,GotoIf($["${CALLERIDNUM:0:3}" = "877"]?s|108) exten => s,5,GotoIf($["${CALLERIDNUM:0:3}" = "888"]?s|108) Thanks --John

I have 1 x100p. Caller id works fine with the beta1 release. Cvshead releases fail with a combination of checksum and ss_thread errors? I'm concerned when beta2 or the 1.2 release comes out it will not work. I have been through the configs I can't find and changes that need to be made to get CVSHEAD to work. Thanks John Hill

Hi, We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones. Most things are running fine ;-) But, when you are calling and you want to Transfer, you need to press first on the 'more' button (4th), then you have the label 'Trnsfr' to Transfer. these are the lables on the softkeys when having a phone call: "Holt / EndCall / Confrn / more" press more and you get

>1- Call comes in without callerid >2- AGI script answers line >3- AGI script asks to record name >4- Park the call and get the parked extension number >5- Ring all the phones in the house (exec Dial) >6- If phone is picked up, play recorded name >7- Wait for DTMF to accept or decline call >8- If accepted, bridge parked call and current call. Mike, I am wanting this

I just saw the S100-FX v2.0 on eBay. I was wondering if anyone has tried it out and what their opinion of it was. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060126/3da2e4e5/attachment.htm

Op 2013-07-13 om 19:58 schreef Michael Szerencsits: > Hi, > > I tested the following config: > > DEFAULT AutoInstall > PROMPT 0 > LABEL AutoInstall > KERNEL http://server/vmlinuz64 > APPEND initrd=http://server/initrd64.img > root=live:http://server/LiveOS/squashfs.img quiet lang=en > keymap=de-latin1 systemd.unit=rescue.target > system.unit=rescue.target

I want to include a new context in my exensions.conf I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort of follow it?! I have a context [local] that I know zapata.conf points to, I have edited extensions.conf and put in my phone, sip and iax extensions. I want to add an sms context. I understand that all calls go through my [local] context and I have

Do functions like call forwarding, do not disturb and so on work with SIP phones? I had these features working with the S100 USB device but can't seem to get them to work with the phones that are plugged into the ATA186s. Also, how do I get an extension that's plugged into an ATA186 to present caller ID? Thanks...

I've spent more than a day trying to write a script to backup my remote server to a local machine using ssh. The script works perfectly when I execute it directly (and I've gotten a number of variations to work okay), but I can't get cron to successfully execute it. There is no result output, so it seems cron is hanging on something, but I don't know what. I'm running it as

Op 2013-07-12 om 14:06 schreef syslinux-owner at zytor.com: > Reason: Message body is too big: 1780404 bytes with a limit of 512 KB So previous not on the mailinglist. > Date: Fri, 12 Jul 2013 23:05:05 +0200 (CEST) > From: Michael Szerencsits <szerencsits.michael at gmx.at> > To: syslinux at zytor.com > Subject: efi64 boot fail during download from kernel and initrd

Hi... I've been fighting this for a while now, trying clean builds of Asterisk 1.14.18, 1.14.19rc3, and then 1.6 Beta 6 today. No workee. :-( Here's the results for various calls made off-hook (push the blue Speakerphone button on the Polycom 430): 988852700 - Phone waits for me to either hit the soft-key "Send" or "EndCall". If I hit "Send",

Hi all, I am setting up my second server with Samba 3.0.21c with LDAP as described in Samba3-ByExample. The server OS is Slackware Linux 10.2, without PAM. All seemed to work perfect until I decided to try folder redirection. I followed the instructions given in the "happy users" chapter, but now new users cant log in. When trying to log in with a new user (umgrtest16) from a WinXP SP2

Hi all! I'm trying to use my first-ever firewire device, and just OOB I'm not getting very far, so advice would be appreciated. When I plug in the device I see some entries in /var/log/messages: Jul 27 14:50:30 fcshome kernel: firewire_core: phy config: card 0, new root=ffc1, gap_count=5 Jul 27 14:50:31 fcshome kernel: firewire_core: created device fw1: GUID 0003f300118123f9, S100

Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls. Can this be because I nowhere use the Answer() application in my dialplan when dialing out? -----Original Message----- From:

Hi, I am trying to read the "credit.lisp" file of the Japanese credit database in UCI repository, but it is in lisp format which I do not know how to read. I have not found how to do that in the foreign library http://archive.ics.uci.edu/ml/datasets/Japanese+Credit+Screening <http://archive.ics.uci.edu/ml/datasets/Japanese+Credit+Screening> Could anyone help me? Best

Hi I'm looking for some help with Vicidial, If you have experience with it and could help with some consulting please contact me off list. Cheers, Joel Hill Asterisk IT jhill@asteriskit.com.au

I have a system running Asterisk 11.2.1 that has had a couple calls between internal extensions get "stuck open". I didn't catch the verbose log for the first one, since I generally don't verbosely log to file, but the second one shows that the call that got stuck was dialed, but the caller hung up before the called device answered. This server is running a hotdesking

I have a extension 981 setup for entering VoiceMailMain: exten => 981,1,VoiceMailMain,([mailbox]@usvm) exten => 981,2,HangUp() I want to pass the calling extension to the context (extension and mailbox numbers are the same). This dosen't seem to work. I get this in the console: Asterisk Ready. *CLI> -- Executing VoiceMailMain("SIP/2504-ba66",

Hi everybody, I'm in a middle of a Asterisk learning period. I am at a very good point except I'm not able to use VoiceMailMain. This Is my simple dialplan regarding VoiceMail ;Number that the IP Phones dial to access voice mail exten => 22999,1,VoiceMailMain (s${CALLERIDNUM}) exten => 22999,2,Wait(3) exten => 22999,3,Hangup Why do I get Forbidden 403 and one console display

Hello, On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using classical File module (in modules;conf and voicemail.conf): cd asterisk-17.3.0 ... make menuselect.makeopts menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; done menuselect/menuselect --enable app_voicemail_odbc