similar to: Can asterisk work with a dynamic IP?

Hi, We have zabbix running and would love to be able to monitor our asterisk box with it. I believe that some sort of SNMP is build in 1.4+ correct? Where do I find more info or a how to on what is supported and how to use it? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have setup asterisk 1.4.22; so far everything good. Except, I am still searching for voIP phones. Which grandstream phone should I buy, this is going to be for small office for testing purposes. I am on a budget, hoping to find someone here who has some used to sell or point me in the direction of a seller. I am in the US. thanks, Mike

Hi, My apologies for the OT. My googling came up empty and hopefully there are some members in the community that could give me a hint how to solve this issue: Cisco 7961 with SIP firmware 8.3.3. Needed to downgrade it to 8.0.4SRS2. The downgrade process started off good. The 7961 got it's IP address via DHCP, found it's SEP<mac>.cnf.xml file and started to "upgrade"

I have the following 3 commands which are supposed to run every night to back up my web server: rsync -avve ssh --numeric-ids --delete --ignore-errors myuser at myisp.com:/usr/home/myuser /Volumes/Downloads rsync -avve ssh --copy-links --numeric-ids --delete --ignore-errors myuser at myisp.com:/usr/home/myuser/mail_boxes /Volumes/Downloads/myuser/mail_boxesBU rsync -avve ssh --copy-links

Hello, I am attempting to implement load balancing on a firewall to allow me to use two ISPs. I have followed the instructions in section 4.2 of the LARTC HOWTO but have got stuck on the split access section I have the options CONFIG_IP_ADVANCED_ROUTER and CONFIG_IP_MULTIPLE_TABLES in my kernel (2.4.24) and have added routes to the /etc/iproute2/rt_tables file but when I try a command in the

hi all, I have a successful (self-compiled from source code) Dovecot v2.2.16 setup active on my Solaris-10 server, with mails stored in user's $HOME/Maildir The mails are fetched from the ISP with "fetchmail", using a "cron" job fetchmail -f $HOME/etc/pop3.myisp where the file "$HOME/etc/pop3.myisp" contains set logfile fetchmail.log poll pop.myisp.be with

before I accuse somebody to "overbill" I would like you to calculate with me: Rate: 0.0189 for calling Taiwan via NuFone Duration: 930 seconds Lets vote for the answers: 0.7269 or 0.2929 ??? bye Ronald Wiplinger

I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger

In the past I used SetGroup and CheckGroup to figure out if my allowed providers lines are all used or not. Since most of my provider have given me a single line anyway, I wonder if there is a way to check if this (provider) line is taken already. How can I do that? Same is with the phone. How can I see in CLI if a phone is now in use or not? "Sip show peers" shows me just if it is

Hi - OK not really an Asterisk question but it is affecting one of my favorite features - emailing voice mail! I've posted on some Linux forums and sendmail.org but no response so I'm hoping someone will take pity on me ;-) My ISP requires SMTP authorization and I'm having a heck of a time getting it to work. I've included the following below: Asterisk 1.4.21 CentOS 5 Sendmail

I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger

I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) exten => ronald,2,Voicemail,u615@office exten =>

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it?

I tried to use message waiting indicator, by "Subscribe for MWI" in the web menu of the phone. However, it does not light up / flash, even if a voice mail is waiting. Where is the switch to turn it to? bye Ronald

Hi all, Has anyone made up custom ring tones for the Polycom SIP phones? We use different rings for different lines, but the ones it comes with are all very similar. In the interesting of sharing, here's one I made up for paging: <PAGE_BEEP se.pat.ringer.13.name="Page Beep" se.pat.ringer.13.inst.1.type="chord" se.pat.ringer.13.inst.1.value="12"

If a user calls and hangs up before the destination party rings, than the in-use flag remains set! This is one case, but maybe there are many other cases. I have created a number the user can dial to reset this flag. However, that is written in the manual!!! Who reads a manual anyway!!!! I want to make to reset all in use flag with a program. Has anybody done it, or has a better idea? My idea

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;

We often find ourselves reading through all sorts of contests on the Internet that never seem to echo our own personal skill set or interests. Perhaps you've even fantasized about a type of contest with the types of prizes and goodies that YOU'D actually enjoy. Maybe you've wished there were something along the lines of a asterisk phone system diagram contest? With prizes ranging from