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...peer to each other, than we get soon also a great amount of users together, and than our service becomes more valuable, ... Let's discuss advantages and disadvantages! bye Ronald -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com +886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmat...

The system log (/var/log/messages) of a CentOS 7.2 system has frequently-repeated message line pairs like: Jul 18 14:00:01 localhost systemd: Started Session 307 of user root. Jul 18 14:00:01 localhost systemd: Starting Session 307 of user root. where the session number increases each time. Looking around on this, e.g. Red Hat Bugzilla bug 727315, it looks like it's when crond starts a

I have a very simple question . I have 2 internal extension 301 and 300 sip phone . I want to these extesion can call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal. How can I do that ? [x1] exten => 300,1,Dial(SIP/300) include => pstnlocal [x2] exten => 301,1,Dial(SIP/301) include =>international [pstnlocal] exten =>

Hello All, Yep I'm a newbe. I'm just started to play with asterisk. What I have Redhat Fedora Core 2 (New install) 3 X100P cards. I installed zaptel-1.0.3 libpri-1.0.3 asterisk-1.0.3 Where should I start?? -- Thanks, David -- This message has been scanned for viruses and dangerous content by KE6UPI, and is believed to be clean. KE6UPI thanks MailScanner for their support. Please

hi, i am using Asterisk CVS-05/31/04. i have the problem that sip clients can make calls over asterisk without registering befor. the xlite is not loged in with any username/secret bit still can make calls over asterisk. how can that be? thx for help. thomas

How do you make the page http://hostname/cgi-bin/astcc-admin/astcc-admin.cgi secure ? , so that only the person administering the calling cards can see the page and make changes to the calling cards, I was thinking of using .htaccess to restrict the access to the page by requiring a password, however since it is a cgi script that does not seem to be posible. Any ideas, any suggestions ?

I have setup flash pannel, ... looks nice, but so far I could not configure it to get more than 4x7 buttons. I tried to make the buttons smaller, but than just the entire picture is smaller. The description says you can have a hundred buttons, .... Can I have multiple flash pannels? E.g. for each department? bye Ronald

Hey, I currently have the voicemail set up so I can dial out from a voicemail box. I seem to not be able to place them. I have put the number in with and without a 1, and then hit pound and it says "Please wait while I connect your call" and then disconnects. Any ideas as to why?

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 =>

Moving from CentOS 5 to CentOS 7, discovered that a test in a SysV initscript for whether it is running interactively, instead of at boot time, no longer works. Under CentOS 7 / systemd, is there some way for the initscript to know this?

...NING[19176]: chan_zap.c"3452: zt_handle_event: Ring/Off-hook in strage state 6 on channel 4 Spawn extension (incoming_88097074, s, 2) exited non-zero on 'Zap/4-1' Hungup 'Zap/4-1' -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com +886 (0) 939--77-55-16 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirma...

Good Day Has anybody here looked closely at the call cost calculation in ASTCC? Can you duplicate the way the cost of a call is calculated? I believe that there is an error in the code. I have fixed it, I think and submitted a patch but we need user comments. I would appreciate if anybody involved would slip over to chech out this link on the bugtracker and provide feedback.

Hello everyone, Version 1.0.7 of Asterisk, Zaptel, libpri, and Asterisk-addons has now been released. Libpri and -addons have not changed, but have been updated anyway to keep the version numbers consistent. All of the tarballs are available on the ftp site. ftp://ftp.asterisk.org/pub/asterisk/ I have posted the ChangeLogs for easy viewing at the following address.

Earlier this afternoon I had this working exten => 2890,1,Answer exten => 2890,2,GoTo(12) exten => 2890,12,Wait(1) exten => 2890,13,Festival('I can say numbers like') exten => 2890,14,SayNumber(1230001,f) exten => 2890,15,Wait(1) exten => 2890,16,HangUp I was so very proud of myself... All of a sudden after a reboot.... I get the following from the same call plan

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the

ok so my table looks like this... REATE TABLE `sip` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `accountcode` varchar(20) default NULL, `amaflags` varchar(7) default NULL, `callgroup` varchar(10) default NULL, `callerid` varchar(80) default NULL, `canreinvite` char(3) default 'yes', `context` varchar(80) default NULL, `defaultip`

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

Over the past 2 weeks I have been able to compile and get an asterisk system up & running on a debian Linux box. I have setup 5 internal sip clients on the lan and all works great! I can also call from outside (PSTN) into the system and reach extensions and services no problem. All is up & running behind a nat firewall with proper ports forwarded and locked down on each device to work

Hi all, a noob here, I am trying to get outbound calls through asterisk working with Broadvoice. I have consulted the following two online tutorials: http://www.broadvoice.com/support_install_asterisk.html http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice in an effort to make outbound calls. My current settings are as follows: sip.conf register =>

...,NoOp exten => s,3,NoOp exten => s,4,Dial(${PHONE_601}&${PHONE_603}&${PHONE_604},30,tr) exten => s,5,VoiceMail(u${DefaultVM}) ; no one answers so send to exten => s,6,Hangup -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com +886 (0) 939--77-55-16 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirma...