similar to: Mailinglist Digium IP-phones : provisioning Digium D70

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

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

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, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. -------------- 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

It looks like logrotate changed behaviour and started to delete old logs instead of compressing them. As you can see it just wiped away the one from 20160327, and that with all configs on default. Does someone know how to fix this?? -rw-r--r-- 1 root root 357256 Dec 14 03:15 ssl_access_log-20151214.gz -rw-r--r-- 1 root root 354106 Dec 20 03:38 ssl_access_log-20151220.gz -rw-r--r-- 1 root root

Adrian, I?ve had huge problems with log rotate recently. For some reason, since the period you have mentioned, I have to restart rsyslog on my CentOS 6.7 servers after logrotate runs either through a cronjob or manually (even on the standard config). I?m not sure if you?ve experienced this to or whether they are linked - but there?s definitely been issues since January with logrotate.? Kindest

Hi all I just saw that I miss some log files in /var/log/httpd. There is a gap in January/February2016, no logs are available for this period. It seemed to work fine with the earlier logs as you can see. Has someone an explanation for this? It's a Centos 7.2 VM with Apache 2.4 running ownCloud only. Default logrotate config. -rw-r--r-- 1 root root 3367604 Apr 20 16:12 ssl_access_log

Hello, I've defined some new musiconhold classes in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [908001] mode=files directory=/var/lib/asterisk/moh/908001 random=yes ; [101001-1] mode=files directory=/var/lib/asterisk/moh/101001/1 random=yes ; [101001-2] mode=files directory=/var/lib/asterisk/moh/101001/2 random=yes But the new classes never show up

Dear members of the list, a customer of mine has some questions and I would like to pose some of them further to you guys. What is the situation with Asterisk and fax over IP ? Can Asterisk receive a fax over a POTS or ISDN line ?? Do I then need a Digium TDM-card and an FXO-module or a T38-gateway ? What phones should I use to automatically configure them from a central place via HTTP or FTP ?

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, I have Snom, Cisco, Grandstream & YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101202/ea25cd7e/attachment.htm

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Hi all, I've been modifying the ringtone on a group of Snom phones like this, depending on certain dial-plan conditions: Exten => s,1,SIPAddHeader("Alert-Info: <http://192.168.0.200/tel_ring01.wav>") exten => s,n,Dial(SIP/mjc_office&SIP/mjc_home&SIP/mjc_lab&SIP/mjc_server,20,trj) Now, I'm migrating slowly to Digium D70 phones, which have a different

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:

Hello list, google returns a discussion on the dev-list when I search for how to mail a voicemail to multiple mail addresses. Is there yet a seperator that actually works to define multiple mail addresses ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

On 11-08-16 18:03, Matt Fredrickson wrote: > On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens at telenet.be> wrote: >> My main reason not to upgrade to Ast 13 is because I'm afraid of losing >> functionality as there are certain functions deprecated/replaced. This can >> also cause headache :-) >> >> I will do so if there is no other option.