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  • AsteriskNow Migration / Shared Extension Space

    - by Aaron C. de Bruyn
    I am testing the possibility of migrating from an old Avaya phone system to AsteriskNow. The migration would cover several hundred phones--but spread out over several years. (Management wants to move buildings to the new phone system one by one as cables get cut or time permits.) Two other directive is that extensions must not change and they want a GUI that other admins (non-Linux geeks) can manage. They currently use 9XXX for all extensions. We linked the Avaya and Asterisk box via PRI card and they both are communicating. From the Avaya side, if we move (for example) extension 9001 to Asterisk, we forward the call over the PRI to the AsteriskNow box and the SIP phone rings. In AsteriskNow we have an outgoing rule '_9XXX' that routes all 4-digit extensions starting with 9 back to Avaya. Here's the trouble. Dialing 9001 (the extension moved over to AsteriskNow) causes the call to be routed out the PRI to the Avaya box, then the Avaya box routes the call back to Asterisk, and Asterisk routes it to the SIP phone. As we get more and more users switched over, it will use up more and more channels over the PRI card. Is there a way I can ask Asterisk to check it's local extensions first--then forward off to the Avaya system if it starts with '_9XXX'? (I know how I can do it when editing the raw config files, I'm just looking for a way to do it in the GUI so other admins can manage it if necessary.) As a last-ditch plan, I know I can specifically add '_9001' as an outgoing call rule and sent it directly to extension 9001--but I'd really hate to do that for several hundred phones

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  • How to enable CDR on AsteriskNow 1.5

    - by Michal Niklas
    I have upgraded PBX to Asterisk 1.6.2.7 and now CDR files are not created. It looks that such logging is disabled: Connected to Asterisk 1.6.2.7 currently running on pbx2 (pid = 5824) Verbosity is at least 3 pbx2*CLI> cdr show status pbx2*CLI> Call Detail Record (CDR) settings ---------------------------------- Logging: Disabled Mode: Simple Asterisk shows that CDR modules are loaded: pbx2*CLI> module show like cd Module Description Use Count cdr_manager.so Asterisk Manager Interface CDR Backend 0 cdr_csv.so Comma Separated Values CDR Backend 0 app_cdr.so Tell Asterisk to not maintain a CDR for 0 app_forkcdr.so Fork The CDR into 2 separate entities 0 func_cdr.so Call Detail Record (CDR) dialplan functi 0 cdr_custom.so Customizable Comma Separated Values CDR 0 6 modules loaded How to enable creating CDR csv files?

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  • How to enable CDR on AsteriskNow 1.5

    - by Michal Niklas
    I have upgraded PBX to Asterisk 1.6.2.7 and now CDR files are not created. It looks that such logging is disabled: Connected to Asterisk 1.6.2.7 currently running on pbx2 (pid = 5824) Verbosity is at least 3 pbx2*CLI> cdr show status pbx2*CLI> Call Detail Record (CDR) settings ---------------------------------- Logging: Disabled Mode: Simple Asterisk shows that CDR modules are loaded: pbx2*CLI> module show like cd Module Description Use Count cdr_manager.so Asterisk Manager Interface CDR Backend 0 cdr_csv.so Comma Separated Values CDR Backend 0 app_cdr.so Tell Asterisk to not maintain a CDR for 0 app_forkcdr.so Fork The CDR into 2 separate entities 0 func_cdr.so Call Detail Record (CDR) dialplan functi 0 cdr_custom.so Customizable Comma Separated Values CDR 0 6 modules loaded How to enable creating CDR csv files?

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  • asterisk send mute command to jukebox on incoming call

    - by Jona
    Hi, We're trialling a Asterisk Now server to take over from our ageing PBX system. One of the "nice to have" features would be the ability to pause or lower the volume on the office jukebox if an incoming call is detected. We currently run a linux jukebox which plays music out of the speakers using mpd and can be controlled by the mpc client. We can manually issue the following command to achieve this: mpc volume 20 Does anyone know how to get asterisk to execute this command or some action that we could hook into when a phone call is incoming to specific extensions?

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  • Server Fax Farm - Need suggestions, or advice

    - by Mike Curry
    We're Looking at creating a large fax farm via T.38 (Fax over Voip - hundreds of incoming and outgoing faxes) on linux servers, anyone have any suggestions on what is available? All my searches return using Asterisk 1.6.x with a commercial product from Digium called "Fax for Asterisk" (with required purchase of "channels" at $38.00 per channel). There must be an open source project out there I can't seem to find. Suggestions welcome! Here is some additional info: We're using Ubuntu 9.10, and planning to use T.38 If I have missed anything, let me know.

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  • Asterisk failing at startup after upgrading to asterisk18

    - by Supratik
    I was using asterisk16 and asterisk16-skypeforasterisk, which was working fine. I have recently upgraded to asterisk18 and asterisk18-skypeforasterisk, after that I am receiving the following error message. Asterisk ended with exit status 1 Asterisk died with code 1. Asterisk could not start! Use 'tail /var/log/asterisk/full' to find out why. When I checked the log I got the following messages. codec_g729a.c: == Found total of 11 G.729 licenses translate.c: empty buf size, you need to supply one Now, if I remove the /var/lib/asterisk/licenses folder it works fine. Can you please tell me what could be the issue here ? Warm Regards Supratik

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  • Skype Connect as SIP/Trunk for Asterisk

    - by Kaurin
    First off: I'm not sure if this should be on superuser or here. I have recently built a few Asterisk boxes with OpenVOX FXO/FXS ports little or no trouble. My current project is building an Asterisk box with SIP trunks. My current employer insisted on getting Skype Business/Skype connect for that purpose. After reviewing the Skype Connect plan, I agreed, because I thought it is going to be straightforward: Purchase G729 licences and setup SIP trunk/trunks. Boy was i wrong :) Here is the setup: The setup is for calling US numbers only via skype (we got skype US minute bundles in skype connect) AsteriskNOW - Asterisk 1.4 + asterisk-gui Trunks: SIP Trunk configured with Skype Connect - shows as registered Users: 2 test extensions. Both work fine when calling each other, voicemail etc works fine too The asterisk box is behind a Mikrotik router which i configured to forward all relevant ports: 5060-5090 UDP, 10000-20000 UDP. When trying out an extension outside of my LAN, it worked. I could make calls to the other extension. Outgoing rule: _NXXXXXXXXX Strip:0 Prepend:+1 Use skype trunk Inbound rule: Trunk: Skype Pattern: s Destination: Extension1 (6210) Here is the output of asterisk CLI (-rvvvvv) with outgoing calls: http://pastebin.com/eWVpL72e you can see the circuit-busy response when using trunk1 (skype) When calling my Skype Connect number from the outside, I get nothing in the logs. Can anyone with Skype Connect / Asterisk experience help out? :)

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