search for: jonsonplayer

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good

Hello, I have a strange situation with an SPA3102 FXO/FXS device. I'm in situation that when i receive a call from PBX line I must forward the calls to 2 VoIP numbers. Right now i have the following settings: (S0<:1010 at GW1>). I want to forward at 1020 too. I tested (S0<:1010|1020 at GW1>) and doesn't work. Did you have any other ideea? Thank you.

Hello, I have a problem with some variables in 1.8.6.0. I set on extension the following lines: exten => h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio, local_lostpackets)}) ; lost packets by local end ** exten => h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio, remote_lostpackets)}) ; lost packets by remote end exten => h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos,

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 guys, I recently observed that my asterisk sends me sms like messages on my phone (Nokia E71), I mean is SMS but is delivered some kind in-band though VoIP. Is strange because this messages contains informations about my voicemail and is sent by voicemail at mydomainxxx.com. I noticed that this messages appears every time when I logged in with my phone on my sip account. I'm interested

Hello ppl, I want to make a special dial plan for routing calls to a peer which has an pin protection. Normally if you want to call through that peer you must first enter pin for example 1234# and after that you hear the tone from line and after that you can dial desired numbers. I tried something like that, but doesn't worked. Did somebody have some clues? exten =>

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. --------------

Hello, I wish to make a simple system to limit peers at x minutes depending of buyer voip packet. Can someone help me with some directions? I intend to make a separate dial plan and every calls to be in cdr table in mysql. Is any chance to make some scripts to drop calls after peer used all minutes? I use asterisk 1.4.34 + mysql + cdr + asterisk-gui administration interface. I don't really

Hello I recently get a Cisco 7940G IP Phone and I try to make several things with it and I en counted many difficulties: 1.) I tried to unlock the phone and to set manually IP Address, Netmask, Gateway etc. I don't get any luck. 2.) I tried to upgrade firmware like they said with tftp server... I downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot directory. I don't get

Hello, i have a simple but annoying problem. I have the following entry in /etc/asterisk/externsions.conf file: ---<Cut Here>--- exten => 10100,1,Wait(4) exten => 10100,2,Playback(transfer,noanswer) exten => 10100,3,Dial(${PHONE30},30,t) exten => 10100,4,Background(extension) exten => 10100,5,Background(is-curntly-unavail) exten => 10100,6,Voicemail(9999) exten =>

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

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

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

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

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:

Hello, I got a SPA3102 and everything works fine except calling from voip to phone on fxo port. The phone ring but doesn't get any sound. I connected SPA at my asterisk server and i want to call from asterisk through SPA to fxo port where i have a regular phone. Thank you for support. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello PPL, someone have any idea for notifying users that they have voicemail waiting when they will register after weren't being registered on asterisk? I need this for nokia terminal e series users. I studied sms service but seems to be only for PSTN lines. I comes with idea to receive a call from asterisk and notified that you have a voicemail. Thank you. -------------- next part