Re: [asterisk-users] Asterisk manager
Voipers Portugal wrote: I know that, that is why I asked if there was any tool that would do something like that, but by acessing the Manager API? Anyone? Our interface uses ARI and MySQL. There is no reason that you could not manage a secure box with the interface app, with MySQL replication of the master table out to a slave-only table on the exposed machine. That way, there is really nothing on the exposed machine to compromise. Go the extra step and SSL the replication channel and the only thing you'd have to have in the clear would be the SIP connections themselves. -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any Suggestions for Election Polling Application?
Gerald Drouillard wrote: Looking to set up an outbound only Asterisk installation for 5 to 10 attendants that will cold calling phone numbers in a database. The customer would like the server to call the numbers as needed and transfer the call to an open attendant if a voice response is detected. The customer called this call banking but it does not seem to translate directly into what Asterisk calls it? Would Asterisk be able to do this? Anybody have good experiences with softphone software? Would Asterisk able to tranfer the person's name/phone number back to the softphone once the connection is made? Any suggestions for SIP phones? Any trouble with using ITSP like Vonage if the user has a good internet connection? This is exactly the application we are building right now. Contact me offlist and I can put you in touch with one of our sales team. -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quadbri + tdm400p + modem-fax
Paco Brufal wrote: On sep/28/2006, Steve Underwood wrote: Lots have tried it. it doesn't work. With Sangoma cards it will work? Thanks. Every time. -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WAS: 64 analog phones NOW: Selection criteria and recipie for a good Asterisk install [long]
Jeronimo Romero wrote: Has anyone tried RedFone?? It is supposed to offload a lot of that bus overhead to the external unit doing TDMoE. We're going to be deploying it within the next month. We're mostly looking at the fault-fail-over aspect of it, but certainly having a dedicated external unit for managing PRI - TDM is also of interest / use to us. I'll let you know how it goes. -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Book on Asterisk
Norbert Zawodsky wrote: Hi everybody! I have some Linux experience but I'm completely new to asterisk. I bought a small VoIP-PBX which has Linux (Kernel 2.6.13) Asterisk (1.2.12) preinstalled and some basic configuration (Wiht a few extensions). Now I want to implement something more, fox example voicemail (storing voicemail data in an extern mysql DB) and so on. And since I don't want to waste your time with stupid questions ... can someone of you recommend a really good book on Asterisk? (To buy or for download) ... or another online source of information which would be helpful for someone like me? I searched Amazon with Asterisk and got 21 hits.. Thanks Norbert Hi, Norbert ... The O'Reily Book for Asterisk: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 Enjoy! -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztcfg / X100P question
Tzafrir Cohen wrote: On Tue, Sep 26, 2006 at 08:45:49AM -0400, Michel Vaillancourt wrote: Tzafrir Cohen wrote: what's the contents of /etc/zaptel.conf ? pbx1:~# cat /etc/zaptel.conf # # Zaptel Configuration File # This file is parsed by the Zaptel Configurator, ztcfg # loadzone = us defaultzone=us # fxsks=1 # channels=1 This line is unnecessary. Just remove it. Oh, *Duh*. I am so used to setting up PRIs... Thanks. pbx1:~# ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) ... Now I need to figure out why its not seeing the card here. However, at least the Bizzare Error(tm) is out of the way. Thanks again. -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztcfg / X100P question
Tzafrir Cohen wrote: what's the contents of /etc/zaptel.conf ? pbx1:~# cat /etc/zaptel.conf # # Zaptel Configuration File # This file is parsed by the Zaptel Configurator, ztcfg # loadzone = us defaultzone=us # fxsks=1 # channels=1 This error comes from ztcfg . Look at ztcfg.c at the tarball of zaptel. Though from the word tones I gather that this is related to the tonezone library. Unfortunately I am not a C programmer, and thus the file in question is largely shrapnel to me. However, from what I can glean, the function in question is static int rad_chanconfig(char *keyword, char *args) and has something to do with struct zt_radio_param. I am puzzled as to what is going on that it thinks a radio is involved. -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztcfg / X100P question
Hi, folks. I've got an X100P Wildcard here. I get an odd error when running ZTCFG on it. === pbx1:~# asterisk -V Asterisk SVN-branch-1.2-r43509 pbx1:~# lsmod Module Size Used by wcfxo 13184 0 zaptel202148 1 wcfxo crc_ccitt 2208 1 zaptel pbx1:~# dmesg | grep -i zap Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-branch-1.2-r1468 Echo Canceller: KB1 pbx1:~# ztcfg -vv Notice: Configuration file is /etc/zaptel.conf line 10: Cannot get number of tones for channel 1 line 10: Cannot init tones for channel 1 2 error(s) detected === I've run google on the errors, but all I turn up are Asterisk source code hunks that really don't explain to me what *triggers* that error. Could someone suggest to me what the issue could be? -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime madness
Scott Pinhorne wrote: Hi All I have 2 sip users setup in the database for realtime and they also have their extension setup in the database. When I register user 1 fine and can make and recieve calls. As soon as i register user2 user1 is then unable to make any calls?? If i put the config fr both users in the flat config files and register them both it works fine, its only when they are running in realtime from database. anyone knwo whats going? a comand line output doesnt shown anything for user1 when user2 is registered. thanks scott Not really sure what is going on here. We use ARI for everything. There 40 phones defined in our office set up, for example, and call routing never hitches up. Can you post a sanitized SELECT * of your SIP user table? -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip configuration using mysql
Arkaitz wrote: Hi, I'm trying to use mysql for sip users management and i'm a bit stuck with a problem. I use asterisk-1.2.12.1 and res_config_mysql from asterisk-addons-1.2.4. The fact is that i've put a row in the mysql sip table for my linksys phone and i can make calls and receive calls with it, but it doesn't appear in sip show peers, and asterisk is unable to find files when I use that phone configured from mysql. Try: /etc/asterisk/sip.conf [general] rtcachefriends=yes -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip configuration using mysql
Arkaitz wrote: Hi, Thanks, now i see the phone in show sip peers, I've been reading about rtcachefriends and now i understand what was the problem. But the other problem is still here :(. It seems that asterisk is unable to find any file in the system, not gsm file nor codec... nothing. It's strange since i provide the same options in sip.conf than in mysql row, but still it fails. i don't understand why. Thanks for your time Suggest you check file permissions vs the user that Asterisk is running as. -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
