[Asterisk-Users] asterisk hardware
Hello folks, anyone using hardware IAX phones with asterisk ? I've googled on this issue and found several hardware phones which support IAX protocol, but before paying money I'd like to know more about what people experiencing with them. Thank you, Tofik Suleymanov ___ --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] asterisk hardware
Give idefisk a try. It works very well for me, its free, and does not crash all the time like Cubix (formerly Firefly). Tofik Suleymanov wrote: Hello folks, anyone using hardware IAX phones with asterisk ? I've googled on this issue and found several hardware phones which support IAX protocol, but before paying money I'd like to know more about what people experiencing with them. Thank you, Tofik Suleymanov ___ --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] asterisk hardware
Steve Totaro wrote: Give idefisk a try. It works very well for me, its free, and does not crash all the time like Cubix (formerly Firefly). Hello Steve, As far as i know 'idefisk' is a softphone, but i need a hardware phone. Thank you for reply. Tofik Suleymanov ___ --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] asterisk hardware
He asked about hard phones not soft phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, May 07, 2006 12:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk hardware Give idefisk a try. It works very well for me, its free, and does not crash all the time like Cubix (formerly Firefly). Tofik Suleymanov wrote: Hello folks, anyone using hardware IAX phones with asterisk ? I've googled on this issue and found several hardware phones which support IAX protocol, but before paying money I'd like to know more about what people experiencing with them. Thank you, Tofik Suleymanov ___ --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 ___ --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] asterisk hardware
anyone using hardware IAX phones with asterisk ? I've googled on this issue and found several hardware phones which support IAX protocol, but before paying money I'd like to know more about what people experiencing with them. I have had three of them for neary two years. Here's an executive review: They're dirt cheap You get what you pay for. Some (not all) firmware has clicks and hums. The later phones look and sound better and are more robust. You can get the firmware and tweak it (see yahoo group) Many languages are available since users have the firmware They do both SIP and IAX, (one at a time via firmware change) They will speak the ip address, server address etc, good for unsighted users. Many don't like this and have disabled it (by a firmware tweak) Conclusion: these phones are excellent for use on the road (assuming a real Internet connection and not through a proxy as many hotels do). I wouldn't recommend them for daily intensive use as they aren't built for it. All of my phones have been continuosly online and working for over one year. They are perfect for setting up asterisk, tinkering with firmware and settings, giving a phone to distant firends or relatives. Google for yuxin,atcom, yahoo for PA1688 mailing list and here http://www.voip-info.org/wiki-VOIP+Phones ___ --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] TDM4xxP
On Sun, May 07, 2006 at 12:21:53AM -0400, Roger Gulbranson wrote: On Sat, 2006-05-06 at 19:43 -0400, Steve Totaro wrote: Roger Gulbranson wrote: On Sat, 2006-05-06 at 07:42 -0400, Steve Totaro wrote: I have a TDM4xxp card with no modules. My question is, will this card be sufficient to provide timing or does it need to have modules? Yes, it will be -- I use such a card in one of my SPARCs. IIRC though, you need to tweak the source to allow it to load wctdm without a module. Can you give me a little clue about what needs to be tweaked? A very nice clue would be very specific and line by line ;-) It's been a year and a half since I worked on this so my memory is a bit hazy. My guess is that the line is: static int timingonly = 1; The original value was a zero. timingonly Try 'modinfo wctdm' . This is a parameter of the module (at least now). So it can be set at modprobe time or through modprobe.conf . -- Tzafrir ___ --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] FOP flash panel: how to reload config files when running
No, you have to kill the op_server app and restart it This is incorrect. You can just send it the HUP (Hangup) signal and it will reload it's configuration files. Isn't that what HUP does? :) ___ --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] asterisk hardware
Wilson Pickett wrote: anyone using hardware IAX phones with asterisk ? I've googled on this issue and found several hardware phones which support IAX protocol, but before paying money I'd like to know more about what people experiencing with them. I have had three of them for neary two years. Here's an executive review: They're dirt cheap You get what you pay for. Some (not all) firmware has clicks and hums. The later phones look and sound better and are more robust. You can get the firmware and tweak it (see yahoo group) Many languages are available since users have the firmware They do both SIP and IAX, (one at a time via firmware change) They will speak the ip address, server address etc, good for unsighted users. Many don't like this and have disabled it (by a firmware tweak) Conclusion: these phones are excellent for use on the road (assuming a real Internet connection and not through a proxy as many hotels do). I wouldn't recommend them for daily intensive use as they aren't built for it. All of my phones have been continuosly online and working for over one year. They are perfect for setting up asterisk, tinkering with firmware and settings, giving a phone to distant firends or relatives. Google for yuxin,atcom, yahoo for PA1688 mailing list and here http://www.voip-info.org/wiki-VOIP+Phones ___ --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 Wilson, thank you very much for extensive and useful answer ! Tofik Suleymanov ___ --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] asterisk hardware
Tofik Suleymanov wrote: Steve Totaro wrote: Give idefisk a try. It works very well for me, its free, and does not crash all the time like Cubix (formerly Firefly). Hello Steve, As far as i know 'idefisk' is a softphone, but i need a hardware phone. Thank you for reply. Tofik Suleymanov Oooops, sorry its late. My favorites in order, Polycom, Snom, Cisco. ___ --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
[Fwd: Re: [Asterisk-Users] asterisk hardware]
Tofik Suleymanov wrote: Steve Totaro wrote: Give idefisk a try. It works very well for me, its free, and does not crash all the time like Cubix (formerly Firefly). Hello Steve, As far as i know 'idefisk' is a softphone, but i need a hardware phone. Thank you for reply. Tofik Suleymanov Oooops, sorry its late. My favorites in order, Polycom, Snom, Cisco. Ps, again, you were asking IAX. Never tried an IAX hardphone since I have never heard good things about quality or reliability. SIP just works. NAT issues are usually simple to fix if that is your reason for going IAX. ___ --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] Assterisk prompts
Does Asterisk have voice prompts for the following. 1. The number you dialled is not available. Please try again later. 2. The number you dialled is not recognised /Obelix ___ --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] more one asterisk hardware
Hello list, We are going to build and deploy telephony-system for approx ~1000 users with ASTERISK as main PBX.I was googling through this mailing list and found a lot of useful information.Some of my questions solved ,some other are still on agenda.Below I will formulate short questions that are still unanswered: 1. What are pros and cons of using SER with ASTERISK ? 2. Should we place SER as a registar server or only as a load balancer (or maybe both) ? 3. What hardware SIP phones are known to work flawlessly in production system with ~1000 users, and where can we buy this amount of phones with the best possible discount ? More about features of those phones please. 4. What hardware (ram,cpu , ... etc etc) should be used in a such environment to make ASTERISK happy ? 5. What if we run ASTERISK on FreeBSD instead of Linux ? Cons and pros please. Thank you, Tofik Suleymanov ___ --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] app_rxfax problem on 1.2.6
Hi! I'm using freepbx, with * 1.2.6, everything is working nice, except fax handling. the incoming faxes got received: May 6 23:24:39 DEBUG[12505] app_rxfax.c: == May 6 23:24:39 DEBUG[12505] app_rxfax.c: Pages transferred: 1 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Image size: 1728 x 1116 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Image resolution7700 x 3850 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Transfer Rate: 9600 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Bad rows3 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Longest bad row run 3 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Compression type2 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Image size (bytes) 0 May 6 23:24:39 DEBUG[12505] app_rxfax.c: == May 6 23:24:42 DEBUG[12505] app_rxfax.c: == May 6 23:24:42 DEBUG[12505] app_rxfax.c: Fax successfully received. May 6 23:24:42 DEBUG[12505] app_rxfax.c: Remote station id: T-Info2001 May 6 23:24:42 DEBUG[12505] app_rxfax.c: Local station id: May 6 23:24:42 DEBUG[12505] app_rxfax.c: Pages transferred: 1 May 6 23:24:42 DEBUG[12505] app_rxfax.c: Image resolution: 7700 x 3850 May 6 23:24:42 DEBUG[12505] app_rxfax.c: Transfer Rate: 9600 May 6 23:24:42 DEBUG[12505] app_rxfax.c: == May 6 23:24:42 WARNING[12505] app_rxfax.c: Unable to restore read format on 'mISDN/2-1' May 6 23:24:42 WARNING[12505] app_rxfax.c: Unable to restore write format on 'mISDN/2-1' May 6 23:24:42 VERBOSE[12505] logger.c: == Spawn extension (macro-faxreceive, s, 3) exited non-zero on 'mISDN/2-1' in macro 'faxreceive' May 6 23:24:42 VERBOSE[12505] logger.c: == Spawn extension (macro-faxreceive, s, 3) exited non-zero on 'mISDN/2-1' but as seen, macro-faxreceive exiting before converting the fax. I think it's becaus unable to restore format, but not sure. The strange is, same freepbx (amp) works nice on 1.0.x I have installed 1.2.6 from source, but spandsp and appfax got from package. I've tried to install newest spandsp, and appfax, but appfax got failed on compilation. Anyone knows the problem? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --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] Assterisk prompts
-BEGIN PGP SIGNED MESSAGE- Hash: SHA256 Obelix wrote: Does Asterisk have voice prompts for the following. 1. The number you dialled is not available. Please try again later. 2. The number you dialled is not recognised Take a look at the following URLs for a good list of the sounds available: http://www.voip-info.org/wiki-Asterisk+sound+files http://www.voip-info.org/wiki/view/Asterisk+sound+files+additional -BEGIN PGP SIGNATURE- Version: PGP Desktop 9.0.5 (Build 5050) iQEVAwUBRF3UajXfYB1SJBTyAQhhMQf+MHMROt2Ag6BNWB+0T173Nw39wAW5WOEq NkBVF7SYWfSsnkUFNbzDtFdwaJbdvwpgjoPbBJpZWuvw4RPm0DigSUSLYVKqwHlq 0QCUSGWqZMQy6bSVo2qTzhDJeGCqGjyT3XWsD7vPgwvuOXqzaFgg3x3kKeqRSR6n Xt4pFoz0D8uBR2AqV7+VAUTyWL7zvxtP7jF99Cd6XPUZs/XynYvKBexXHEjVVnzp 908Nm+wKV+9B22oY0NbW3OV/fynm7ae589uNV7JmDbBMQL+toauHFd7FMLVTd+0W w2jInIsBEIC6GN5rMMBEFTH/pvjCEF1kEGmxMxIjP5ctqkz4a0pZig== =oA6N -END PGP SIGNATURE- ___ --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] asterisk hardware
On Sunday, May 07, 2006 10:07 AM Steve Totaro wrote: Oooops, sorry its late. Obviously. :-) My favorites in order, Polycom, Snom, Cisco. Since when do these use IAX? He asked for IAX hardphones... If I am mistaken let me know since I am looking for good reliable SNOM-like IAX phones as well! :-) Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --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] www.SavaJe.com
Small java applications seem to be quite common among many phones. I tend to stay away from such phones. I'd first like phone vendors to get their acts together and provide me decent standard interfaces. Currently I need an expensive cable just for the pleasure of connecting my mobile phone to a PC, and I can only sync to it using a proprietary protocol. My phone has no such Linux software and only a very buggy windows software for that. I'm not an expert in that field, but from what I understand, this is basically possible with many of the newer mobile phones. -- Tzafrir Sorry, I just realised Tzafrir isn't our target market..sorry for wasting all of your time..for those of you that actually use half of your phones 'basic' functionality like data connectivity this could be a cool thing for your Asterisk application development. See you at JavaOne show. Dean Collins ___ --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] FOP flash panel: how to reload config files when running
Wilson Pickett wrote: No, you have to kill the op_server app and restart it This is incorrect. You can just send it the HUP (Hangup) signal and it will reload it's configuration files. Isn't that what HUP does? :) No, HUP sends the Hang UP signal, causing an application to reload/re-read it's configs without ending the application. A TERM causes will KILL it. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Unable to Make Asterisk-addons
Turns out there is a fault due to the new way that asterisk installs It isnt feasible to fix it until the asterisk installation methodhas settled down a bit. Does anyone know a time scale on this? Dan Journo ___ --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] H323 to SIP
Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --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] H323 to SIP
You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --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 ___ --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] H323 to SIP
I dont have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --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 ___ --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 ___ --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] H323 to SIP
You could begin with: http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation http://www.voip-info.org/wiki/view/Asterisk+H323+channels http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels and much more. You need to install chan_h323 module and configure as well as you need in your application, (if you need gatekeeper functionality maybe you need to try before GNUGK), and later via extensions make wherever you need. Its a little complicated and you need how to work with asterisk before doing all this things. Regards Farhad Ibragimov escribió: I don’t have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --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 ___ --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 ___ --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 ___ --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] H323 to SIP
Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could begin with: http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation http://www.voip-info.org/wiki/view/Asterisk+H323+channels http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels and much more. You need to install chan_h323 module and configure as well as you need in your application, (if you need gatekeeper functionality maybe you need to try before GNUGK), and later via extensions make wherever you need. Its a little complicated and you need how to work with asterisk before doing all this things. Regards Farhad Ibragimov escribió: I dont have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --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 ___ --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 ___ --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 ___ --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 ___ --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] another question about hardware for using with asterisk
On Sun, 7 May 2006, Tofik Suleymanov wrote: Hello folks, firstly, thank you for your useful and fast answers ! Is there anybody using D-Link SIP phones ? Are D-Link SIP phones ok to install in production environment ? Give your comments please. Tofik Suleymanov I'll pipe in on this one. We got a few D-Link's in to test, and for some strange reason, they're the only ones that won't hold a steady IP address, and haven't been stable at all. You're better off getting cisco's or polycom's, those are designed much better and seem to work really well. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] FOP flash panel: how to reload config files when running
On Sun, May 07, 2006 at 08:44:41AM -0400, Doug Lytle wrote: Wilson Pickett wrote: No, you have to kill the op_server app and restart it This is incorrect. You can just send it the HUP (Hangup) signal and it will reload it's configuration files. Isn't that what HUP does? :) No, HUP sends the Hang UP signal, causing an application to reload/re-read it's configs without ending the application. A TERM causes will KILL it. Actually the name Hang UP is misleading here. SIGHUP basically tells a terminal application that its terminal has been hung up. Normally such a program should terminate, because you wouldn't want to leave a stale program running (see also nohup(1) ). By convention X programs behave basically the same on SIGHUP. Although their hangup is really the closing of the connection to the X server. Daemons have no controlling terminal normally. So SIGHUP is meaningless for them. For some strange reason, that signal was abused to tell programs with no terminal to re-read their configuration. /me wonders what should happen when you run 'asterisk -c' in a an xterm and then close that xterm. ;-) -- Tzafrir ___ --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] Announcement Haiku
This extremely useful dialplan requires the standard Asterisk sounds, plus the additional ones in the asterisk-sounds package. Scott. [haiku] exten = s,1,Playback(privacy-please-dial) exten = s,n,Playback(letters/a) exten = s,n,Playback(high) exten = s,n,Playback(letters/q) exten = s,n,WaitExten(60) exten = 01,1,Playback(a-collect-charge-of) exten = 01,n,Wait(1) exten = 01,n,Playback(attention-required) exten = 01,n,Wait(1) exten = 01,n,Playback(shall-i-try-again) exten = 01,n,Wait(3) exten = 01,n,Goto(s,1) exten = 02,1,Playback(at) exten = 02,n,Playback(midnight-tonight) exten = 02,n,Wait(1) exten = 02,n,Playback(doing-enum-lookup) exten = 02,n,Wait(1) exten = 02,n,Playback(do-not-disturb) exten = 02,n,Wait(3) exten = 02,n,Goto(s,1) exten = 03,1,Playback(message-from) exten = 03,n,Playback(detroit) exten = 03,n,Wait(1) exten = 03,n,Playback(chicago) exten = 03,n,Playback(houston) exten = 03,n,Playback(letters/l) exten = 03,n,Playback(letters/a) exten = 03,n,Wait(1) exten = 03,n,Playback(for-english-press) exten = 03,n,Playback(plugh) exten = 03,n,Wait(3) exten = 03,n,Goto(s,1) ___ --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] H323 to SIP
On Sun, 7 May 2006 19:58:26 +0500, Farhad Ibragimov [EMAIL PROTECTED] wrote: Thanks Try reading this URL (spanish language): http://www.ecualug.org/?q=2006/02/28/comos/asterisk_1_2_4_agregando_soporte_para_el_protocolo_h_323 With the page instructions I can call from and to H.323 to every registred SIP/IAX2/H.323 device with my Asterisk box. Good luck, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could begin with: http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation http://www.voip-info.org/wiki/view/Asterisk+H323+channels http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels and much more. You need to install chan_h323 module and configure as well as you need in your application, (if you need gatekeeper functionality maybe you need to try before GNUGK), and later via extensions make wherever you need. Its a little complicated and you need how to work with asterisk before doing all this things. Regards Farhad Ibragimov escribió: I dont have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 -- Guillermo V. Salas M Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 1er Piso Teléfono: 262 8071 Celular : 09 985 5138 Manta - Manabí - Ecuador ___ --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] another question about hardware for using with asterisk
On 7 May 2006, at 16:16, Aaron Daniel wrote: On Sun, 7 May 2006, Tofik Suleymanov wrote: Hello folks, firstly, thank you for your useful and fast answers ! Is there anybody using D-Link SIP phones ? Are D-Link SIP phones ok to install in production environment ? Give your comments please. Tofik Suleymanov I'll pipe in on this one. We got a few D-Link's in to test, and for some strange reason, they're the only ones that won't hold a steady IP address, and haven't been stable at all. You're better off getting cisco's or polycom's, those are designed much better and seem to work really well. I like the elmeg 290 (cheaper clone of the older snom) It looks, feels and sounds like a 'real' office phone but doesn't dominate a desk. I only have 2, so I can't comment on mass deployment ! I also don't know if they sell outside Europe. Tim. Tim Panton [EMAIL PROTECTED] ___ --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] asterisk hardware
Keyboardot ragadtam, hogy va'laszoljak Koopmann, Jan-Peter osszedobalt bytejaira: Since when do these use IAX? He asked for IAX hardphones... If I am mistaken let me know since I am looking for good reliable SNOM-like IAX phones as well! :-) I'm sorry if i recommend some foolish (i've just joined the maillist) but have you tried PA168 chip based hardphones and ATAs? www.areadfox.com for example -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --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] asterisk hardware
At 12:55 PM 5/7/2006, you wrote: Keyboardot ragadtam, hogy va'laszoljak Koopmann, Jan-Peter osszedobalt bytejaira: Since when do these use IAX? He asked for IAX hardphones... If I am mistaken let me know since I am looking for good reliable SNOM-like IAX phones as well! :-) I'm sorry if i recommend some foolish (i've just joined the maillist) but have you tried PA168 chip based hardphones and ATAs? www.areadfox.com for example I believe you mean http://www.aredfox.com/eindex.htm . Tom ___ --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] asterisk hardware
I'd rather shoot myself in the head! other day we had a site that flashed the PA168 chipset phones with new firmware and they all ended up with the same MAC address!! I thought that shouldn't happen normally ...And talk about nasty cheap effects, sidetone, distortion and the list goes on. RobOn 07/05/06, Woodoo People .pGa! [EMAIL PROTECTED] wrote: Keyboardot ragadtam, hogy va'laszoljak Koopmann, Jan-Peter osszedobalt bytejaira: Since when do these use IAX? He asked for IAX hardphones... If I am mistaken let me know since I am looking for good reliable SNOM-like IAX phones as well! :-)I'm sorry if i recommend some foolish (i've just joined the maillist)but have you tried PA168 chip based hardphones and ATAs? www.areadfox.comfor example --WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com[EMAIL PROTECTED]]iCQ#33118021[wpeople.on.iRCNet [EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] asterisk hardware
Well, to tell the truth, the phones, what available in Hungary, is 90% working. The other 10% is sometimes bad as you get out off the box, sometimes it's noisy, echoing, crappy sound, rebooting, etc. Is i asked so many folks on Cebit (who resells this phone) most of them, told me, there are two kind of this phone. One is cheap and crappy, other is not as cheap but at least working :-) Either way, i think the chip itself is working nice, has gpl source (look on voip-info.org) so you can go for it, and don't undertake the chip because of a nasty manufacturer. I'd rather shoot myself in the head! other day we had a site that flashed the PA168 chipset phones with new firmware and they all ended up with the same MAC address!! I thought that shouldn't happen normally ... And talk about nasty cheap effects, sidetone, distortion and the list goes on. Since when do these use IAX? He asked for IAX hardphones... If I am mistaken let me know since I am looking for good reliable SNOM-like IAX phones as well! :-) I'm sorry if i recommend some foolish (i've just joined the maillist) but have you tried PA168 chip based hardphones and ATAs? www.areadfox.com for example -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --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] asterisk hardware
Since when do these use IAX? He asked for IAX hardphones... If I am mistaken let me know since I am looking for good reliable SNOM-like IAX phones as well! :-) I'm sorry if i recommend some foolish (i've just joined the maillist) but have you tried PA168 chip based hardphones and ATAs? www.areadfox.com for example I believe you mean http://www.aredfox.com/eindex.htm . yes :-) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --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] Need a Service that allows me to call Toll Free Outbound numbers
Simple as that please email me direct. [EMAIL PROTECTED] Also looking for a U.S. DID provider as well as orig provider. ___ --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] Voicemail indication for analog phones
I have 20 or so users with analog cordless phones connected via a 24port FXS box (vegastream). The vegastream supports voicemail indication via a studder tone, which is great but I have some users asking for a more positive (or proactive) indication of voicemail. I had a couple ideas; 1.If an extensionhas a voicemail then ring the extension once per hour for one ring, then hangup. 2.If an extension has a voicemail then ring the extension once per hour and if answered, play something along the lines of You have X voicemails, please dial *97 to listen to your messages. Something that I have to keep in mind is that some of these phones are in residential areas, so either of the above actions would need to be time limited (active from 8:00am to 9:00pm for example). Has anyone done something like this before, anyone have any better ideas? Will this have to be executed by Cron (probably because the above examples are time based)? Thanks for any input. Garth ___ --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] Need a Service that allows me to call Toll Free Outbound numbers
Bob's Leaky News Service schreef: Simple as that please email me direct. [EMAIL PROTECTED] Also looking for a U.S. DID provider as well as orig provider. FWD (FreeWorld Dialup) will allow Toll Free outbound... So will some of the Finarea services (check the wiki for their various services. According to http://www.voip-info.org/wiki/view/Finarea+SA you get free outbound USA calls (not quite free, as you have to pay a small amount to add call credits to your account which have to be replenished every 120 days) from: internetcalls; sipdiscount; voipcheap; voipdiscount and voipstunt) HTH! -- Francesco Peeters PIII-450 - 512 MB RAM - 2x HFC-PCI - BRIstuff Florz patch AMD Duron 1GHz - 512 MB RAM, 2x HFC-PCI - vISDN ___ --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] Re: Re: Re: Voicemail error
At 04:33 PM 5/6/2006, you wrote: All I need is a way to uppercase a string, which from everything I've read so far isn't in the code. Then again, I could just use all uppercase for my SIP/IAX device names even if it *does* look ugly. ;) What if you just prefix all names with the number one? 1dave 1dave-cell Ira ___ --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] SSH from System() ?...
Hi, I would like to execute a command on a different system using ssh. When I execute the command from the CLI on the asterisk machine, it works fine (I set up RSA keys on both sides) When I execute the same command from System() inside the dialplan, the log shows it is being executed, and a session is established, but the remote host never receives the command. I *think* it has to do with the command shell environment in which the system command is opened... Any suggestions on how to set this up would be appreciated... -- Francesco Peeters PIII-450, 512 MB, 2x HFC-PCI, BRIstuff Florz patch AMD Duron 1GHz, 512 MB, 2x HFC-PCI, vISDN ___ --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] SSH from System() ?...
Francesco Peeters schreef: Hi, I would like to execute a command on a different system using ssh. When I execute the command from the CLI on the asterisk machine, it works fine (I set up RSA keys on both sides) When I execute the same command from System() inside the dialplan, the log shows it is being executed, and a session is established, but the remote host never receives the command. I *think* it has to do with the command shell environment in which the system command is opened... Any suggestions on how to set this up would be appreciated... Never mind, I had wrong permissions on the authorized_keys file on the target machine! -- Francesco Peeters PIII-450, 512 MB, 2x HFC-PCI, BRIstuff Florz patch AMD Duron 1GHz, 512 MB, 2x HFC-PCI, vISDN ___ --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] Chanspy Specifying Agent not Working
I use chanspy to listen to agents for QA. It works great for listening to random conversations. * works to switch between calls. I cannot however specify the agent I want to listen to. I dial the agent's extension and press # but get connected to some random conversation, not the one I dialed. Anyone know what I am doing wrong? Thanks, Steve ___ --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] Enter CDR Account code during call
Does anyone have any suggestions as how to enter a CDR Account code during a call? I know it can be done in the extension logic before the answering the call, but I wanted to optionally enter an account code on certain calls without prompting on every call before or after the call? ___ --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] SSH from System() ?...
On Mon, May 08, 2006 at 12:40:55AM +0200, Francesco Peeters wrote: Hi, I would like to execute a command on a different system using ssh. When I execute the command from the CLI on the asterisk machine, it works fine (I set up RSA keys on both sides) When I execute the same command from System() inside the dialplan, the log shows it is being executed, and a session is established, but the remote host never receives the command. I *think* it has to do with the command shell environment in which the system command is opened... Any suggestions on how to set this up would be appreciated... My guess is that the host key is not in the known_hosts file for the user asterisk . Add it manually (just copy the line from your file, or your whole file) and see if it helps. -- Tzafrir ___ --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
