similar to: Digium IP Phones

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:

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

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",

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

Hello Asterisk users, We are having MANY but MANY problems configuring an analog fax machine to work properly on Asterisk, the first thing we do was to plug in the Fax analog machine to the FXS port of the Digium TDM410P, we set echocancel=no on zapata.conf and also faxdetect=yes on general section, but our Asterisk CRASH every time we try to send/receive fax! We are using Asterisk 1.4.21 and

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 have a repetative task in R and i'm trying to find a more efficient way to perform the following task. lst <- list(roots = c("car insurance", "auto insurance"), roots2 = c("insurance"), prefix = c("cheap", "budget"), prefix2 = c("low cost"), suffix = c("quote", "quotes"),

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

Hello, I've recently given a try to a Digium D70 phone. At the moment, I'm configuring them though config files with a DHCP server and not using DPMA. Of course, I'm connecting them to Asteris (PJSIP stack on 13.7.0). Which is the best place to: - read about past issues - open new tickets for remaining issues. Best regards -------------- next part -------------- An HTML attachment

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

Today I attempted to upgrade to S10_U2 and migrate some mirrored UFS SVM partitions to ZFS. I used Live Upgrade to migrate from U1 to U2 and that went without a hitch on my SunBlade 2000. And the initial conversion of one side of the UFS mirrors to a ZFS pool and subsequent data migration went fine. However, when I attempted to attach the second side mirrors as a mirror of the ZFS pool, all

Hello, I can do simple, "yum install asterisk18-*" and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, *"You do not appear to have the source for the 2.6.32-4-pve kernel installed".* * * 1- Based on above error and Google search I have

Howdy All, I'm considering Asterisk / Digium as a replacement to my existing phone switch. I need to continue to be able to push analog lines between multiple buildings in a campus environment. The Digium Analog 410 Card manual states it's not recommended to go beyond 1500 feet distance for an FXS card, and no line should leave the building or be bundled. The 2400 Series Manual does

Greetings all, After a long day of fighting with GTalk and having it finally working, I wanted to setup DPMA on my Digium phone. So first of all, I had to reinstall it all and reconfigure it all, since it works only on certified versions, and my installation was not from the certified branch. It took a long time of recompiling, testing, adding missing stuff, but I got it straight. Now, I

Hello, is there a mailinglist where I can post questions regarding Digium IP-phones ? I have the following question : I'm trying to provision a Digium D70 IP-phone from a https provisioning server. The Digium D70 contacts the provisioning server correctly but seems to log in with the wrong credentials : /var/log/ssl_access_log : XX.XX.XX.46 - - [22/Jan/2014:12:15:09 +0100] "GET

Hello everyone, I see that the "yum install freepbx" from Digium repository actually installs the latest FreePBX which is nice. However, I don't see the old FOP in FreePBX anymore. Is there a way to install FOP or FOP2 through repository? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dears; I need to order Digium card and not able to know which one is the best quality? Is it that of AEX with the end E or EF or P or B? I saw those card that its slot is small (I think those that end by EF), are they the best card? Really I am caring to have a card that has echo cancelation and the voice volume is high enough (because previously I faced the problem that the voice volume was

Hi all, I'm trying out a Digium D70 phone with Asterisk 11. My voicemail messages are listed in the "visual voicemail" app on the phone, but they do not successfully play back. The correct duration is shown, but the progress bar just jumps back to zero when I press the "Play" softbutton. I can hear my messages fine if I "manually" dial into my voicemail

Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance > > Message: 10 > Date: Thu, 18 Mar 2010 00:21:06 -0400 > From: Zeeshan Zakaria <zishanov at gmail.com> > Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE >

I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2.