similar to: Call Transfer Fails - Not a Valid Extension

We are trying to add voice mail to our hotel rooms. Our current phone instruction cards say 'to reach voice mail dial ext 456". Replacing those instructions is not feasible at the moment. We have Feature Code *97 that takes them directly to their voice mail box. Question - What is an easy way to have exten 456 dial *97. We are using AsteriskNow distro, version11. Phil Ledon

Hi all, Just installed Asterisk 13 on CentOS 7 and have run into a problem. The Scenario is this: Asterisk is on the internet the Phone, a D40, is behind NAT So someone calls the number and Asterisk routes the call to the D40 Everything works fine and the call is established, but then after 5 min. the caller stops getting audio from the D40 but there is still audio to the D40. using both

Hi Is the keep alive activated on the phone? On Thu, Aug 11, 2016, 5:36 PM Dovid Bender <dovid at telecurve.com> wrote: > 1) Does it happen every time at the 5 minute work? > 2) Have you done a dump on the client side to see if the NAT device is > dropping the packets? > 3) Is the phone behind a load balance internet connection and is the RTP > port changing? > > >

Would it be possible to set the ringtone based on the number that was dialed? Example of what the goal is: Dial Denver number Incoming calls ring with ringtone 1 Dial main number Incoming calls ring with ringtone 2 We are currently using Digium D40, D50, D70 phones. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All; Any one used Digium IP Phones D40? I need to know if they are stable with good voice quality? Comparing to Polycom 330, which is better? Let us talk frankly although I know that we have to support Digium. Regards Bilal

I'm trying to solve a mystery for the last couple of days. I have a mix of D70, D50 and D40 behind NAT. Server is in a colocation, not a VPS. For several years, everything was working fine, no issues. A few days ago I started having problems at one particular site. NO CHANGES have been made to this office network - same router, switch and internet provider. No new equipment added or

Hi, We have a number of 4200''s setup using a combination of an SVM 4 way mirror and a ZFS raidz stripe. Each disk (of 4) is divided up like this / 6GB UFS s0 Swap 8GB s1 /var 6GB UFS s3 Metadb 50MB UFS s4 /data 48GB ZFS s5 For SVM we do a 4 way mirror on /,swap, and /var So we have 3 SVM mirrors d0=root (sub mirrors d10, d20, d30, d40) d1=swap (sub mirrors d11, d21,d31,d41)

Hello, I use Brendan''s sysperfstat script to see the overall system performance and found the the disk utilization is over 100: 15:51:38 14.52 15.01 200.00 24.42 0.00 0.00 83.53 0.00 15:51:42 11.37 15.01 200.00 25.48 0.00 0.00 88.43 0.00 ------ Utilisation ------ ------ Saturation ------ Time %CPU %Mem %Disk %Net CPU Mem

I'm writing a function and keep getting the following error message. myfunc <- function(lst) { lst <- list(roots = c("car insurance", "auto insurance"), roots2 = c("insurance"), prefix = c("cheap", "budget"), prefix2 = c("low cost"), suffix = c("quote", "quotes"), suffix2 = c("rate",

This is a continuation of the thread: "yum update fails: CentOS 5.2". I installed a fresh Centos 5.2 and ran yum update. It ran correctly, except for these issues: 1. As previously mentioned, after deciding what it is going to do, It doesn't ask if its ok. But if I type y<enter> it reports: Is this ok [y/N]: Downloading Packages: 2. During the long period it

I'd like to have Asterisk log useful messages during operation. Is there any way in extensions.conf that I can manually log messages to a file, say via syslog()? The console output is ugly, with all the extra "Executing NoOp("SIP/pstn.voip.com-08a28bd0"," crud at the front of each line. I'm not sure how to save console output anyway. Thanks, Doug.

Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? Many thanks -------------- next part -------------- An HTML attachment was

I'm trying to understand why vendors keep making 100Mbps integrated 1-port switches in their hardware SIP phones. Even the recently-announced D40 and D50 Digium phones are limited to 100Mbps. Only the more expensive models (like the D70) can run at 1000Mbps. However, you can't expect a firm with hundreds of extensions to buy the most expensive model... And gigabit speed is important when

----- Original Message ----- | Not tried automount with Centos 7 nor with selinux. | | With that said autofs relies on nfs mounting to work, so have you | started there by attempting to manually mount /home? | | Another place to look is at the hostname. I've had problems where auto | mount doesn't like the short name and insists on using a FQDN, to get | around that you could try using

Ok, I can't quite figure out how to make this work. I want to simultaneously log everything for facility local5 in a local file and a remote syslog-ng server. local7 is working fine getting the boot.log log entries transferred over to the syslog-ng server, but not so much with local5. Local logging of local5.* on the client is working fine now. SERVER: /etc/syslog-ng/syslog-ng.conf

Sipura has implemented auto-answer in version 0.9.5 of the SPA-841 firmware. However, it is implemented via the Call-Info header, which Asterisk stable doesn't currently support. The attached patch implments a quick hack to support the Call-Info header from the Dial() application by way of setting the CALL_INFO variable. For example, the following macro can be used to dial up a single

So I'm working with our Digium D40's and we're not using DPMA. This video ( http://www.youtube.com/watch?v=zcuocp01pfM#t=35s ) shows presence information being displayed in the Contacts application. Obviously the video is showing DPMA in play. Is it possible to enable this functionality without it? Is this status information only available on higher-end Digium phones? In the contacts

I have a really long functions, and at the end of the function, I am using a if statement to tag certain keywords based on whether they have certain values contained in them. However, the if statement doesn't seem to work. When I had split up the commands into various functions, it worked fine, but I'm not sure what going on now that it's combined into a single function. myfunc

I'm using Rsync 3.0.5 between two Debian boxes. I have an rsync script setup that uses --log-file=mylogfile.log and also I have specified --log-file-format=whatever. But I can't seem to get the --log-file-format to be recognized. I don't get any sort of errors, but the log format that I get is just the default, no matter how I change the format option. Just for simplicity's

Is there any chance to running new games under Wine (because I've seen some new videogames running in Youtube) and, what things can be made to try to get those games working. I don't know much about wine so I don't know how it works but, Colud we make some kind of guide about problems, solutions and wine moddings to encorauge users to get working new games?