search for: jonson

...itef?view=vs-2019 the header to include is float.h also the ymath.h file contains the comment /* ymath.h internal header */ so probably shall not be used. I do not know how compiler-rt works only tried to compile rustc that is using compiler-rt How shall I go forward with this problem? BR/Andreas Jonson -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20190404/a7ffce47/attachment-0001.html>

Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking about SPA3102. What you guys think about it. Is ok, is working with asterisk, can i use it like voip peer. Thank you for your advice. Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070117/93bc7fdb/attachment.htm

Hello, I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to configure voice part on it. I cannot get it if I can use like peer for my asterisk. Please help me with some tips. Thank you guys. Regards, Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070126/a49e3bdb/attachment.htm

Hello, I want to limit calls per sip account user. How may I realize this setting? For example I want to limit to 10 min all possible calls from an account or to limit external calls to 10 min and local call remain unlimited. Thank you for support guys. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...un? Does it help to copy dlls from a naitive NT system (which?). Is a wine snapshot more stable than daily builds? Please, any hits are greatly appreciated. I've heard rumors on people running Pagemaker rock stable under wine and that makes me curius and hopefull. TIA and regards, -- Fredrik Jonson

...x numbers instead of 00xxx numbers. Some users of main use + instead of 00 for international dial. Is there any solution for this problem? As far as i readed in asterisk is some kind of replacement of characters in dial plan command. Could i use that for archiving this option? Thank you for help. Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130611/9505a49e/attachment.htm>

Hello, I just want to put all my sip accounts in mysql and asterisk use it from mysql. How can I do that, could you be more specific because I readed alot on wiki and i'm lost... I don't know what to modify in Makefile from channel directory. I use asterisk 1.4.4, that is already compiled and i also have CDR in mysql. I must create manny accounts and I want to realize that from mysql.

Hello, i hev a subscription to a international voip provider and I want all calls for numbers _001xxxxxxxxxx to go through my voip provider. I tried many settings in sip.conf , extensions.conf and iax.conf. Please give me some simple example for how can i transfer the specified calls to my external voip provider. What may I put and where in witch file. Thank you for your support. --------------

...ns.c: 221 sip_acf_channel_read: Unrecognized argument 'rtpqos, audio, remote_jitter' to CHANNEL [September 5 22:39:33] WARNING [14432]: func_channel.c: 393 func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio, remote_jitter' Any idea how I can fix? Best regards, Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110905/5f0e26d1/attachment.htm>

...ne. 4.) On the other side my phone doesn't have ringtone either. Any clue how may I put ringtones on it? I know is a lot of questions for you guys, but I browse on cisco.com web site and google for hours and I don't get it any clue to make work this phone in any way. Thank you for help. Jonson.

Okay, i'll move my discuss to asterisk-users. Thank you. On 1/17/07, Tzafrir Cohen <tzafrir.cohen@xorcom.com> wrote: > > > On Wed, Jan 17, 2007 at 04:39:03PM +0800, ??? wrote: > > Jonson Player wrote: > > > Hello, I intend to buy a FXO/FXS device from Linksys. > > > I'm thinking about SPA3102. What you guys thik about it. > > > Is ok, is working with asterisk, can i use it like voip > > > peer. Thank you for your advice. > > Generally y...

Hello, i just installed asterisk 1.2.15. I got this error message. Somebody can help me? Thank You. Feb 27 11:47:43 NOTICE[17086] cdr.c: CDR simple logging enabled. Feb 27 11:47:44 WARNING[17086] pbx.c: Already have an application 'Pickup' Feb 27 11:47:44 WARNING[17086] loader.c: app_directed_pickup.so: load_module failed, returning -1 Feb 27 11:47:44 WARNING[17086] loader.c: Loading

Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. -------------- next part

...1.4.18.1 built by root @ h-gw on a i586 running Linux on 2008-03-23 00:26:44 UTC) and everything is ok but when i call an extension i cannot hear anything. I don't get any visible error on sip debug... i changed the codecs... everything is the same... Can someone help me with that? Thank you. Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080323/89da28fc/attachment.htm

Hello, I lookin' for a call in number from UK or USA. Can somebody offers me a peering for this or specify any sip provider that offers this thing? Thank you very much, Jonson.

Hello, I want to use an Huawei stick model K3765 which support voice with asterisk. I'm begginer with this kind of interaction from asterisk with external devices. Can someone guide me what should i configure to use this device? Thank you for support, Regards, Jonson. --- www.Mobile-Wi.Fi

...till equal to 2.000000e+00 and 4.559372e-33, respectively.  I am concerned that these will affect my estimates since my study is only interested in obtaining the orthogonal main effects of the factors. How can I improve my design so that I can be ensured of the main effects? Thank you very much. Jonson M. Javier  [[alternative HTML version deleted]]

Hello, i just want to activate SMS service between my asterisk local sip accounts and between asterisk and local sip accounts. How can i do this thin? Also i tried smsq to an account but all i obtained is a error message: ---<Cut Here>--- May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to open /var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1: Permission denied, deleting May 22

Hello if someone found some method to authentificate to asterisk with nokia push to talk clients please send me all your documentations and the tests results, I really need this for a project of main and i wanna dig deeper to solve this mister. Thank you guys for your cooperation. Alex i put you at cc because i know you find this interesting too and maybe meanwhile you already know more about

Hello, I'm wandering how can I make voicemail notification when i got a messages in asterisk mailboxes. For the moment i have e-mail notifications, but I readed that I can do also a sms notification to local sip accounts. Also I'm wandering if i can make something like callback from asterisk to sip account, and play voicemail check, when the user log in. Is there someone that use this