search for: sampietro

hi, i installed asterisk and works very well, the only problem is that when i try to call a direct number of a company that has a normal PBX i got this error: to 10.8.210.153:5060 == Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro) -- Executing Goto("SIP/a.sampietro-f7be", "doisdn|BYEXTENSION|1") in new stack -- Goto (doisdn,00115601992,1) -- Executing StripMSD("SIP/a.sampietro-f7be", "1") in new stack -- Executing DigitTimeout("SIP/a.sampiet...

hi, can some one tell me a good IP phone (not software, but a "real" phone :) that work well with asterisk? how mutch does it cost a good IP phone? i made a VoIP network for my company, but now we are using a client for PC phone... i'd like to buy a IP phone, can someone tell me witch model i should buy? thanks, Angelo

hi, i need to do a demo of asterisk in a office that doesn't have an ISDN line, can someone tell me if asterisk works also if i use a modem card mapped on /dev/ttyl0 insted of use a isdn card? did someone try to do something similar before? thanks for your help!! Angelo

Hello guys, Have you guys experienced problem with digit 0 from x-lite. I cannot access asterisk services with number containing digit 0. For example i have voicemail with mailbox number 6100, if i login from x-lite login always fail because asterisk only read number 61.

hi, i have to do a demo with asterisk, unfortunately i don't have yet an x100p card, so i need to use a 56k voice modem on my motherboard... could someone tell me how i can configure asterisk to use this modem to call? thanks a lot for the help!!! Angelo

i read that quite all of you are using the driver for the isdn that you are using is the digium... i use I4L and i got a problem with some call through normal PBX, can someone tell me where i can download the digium drivers for the ISDN card? i didn't find it... but i'd like to try if is a driver issue... (i use a digium isdn card...) thanks a lot, Angelo

hi, my asterisk has some problem when i call a destination number and the phone call is through a normal PBX... you can see from the trace that there is a "Forbidden" error... but i don't know why... someone could help? thanks, Angelo this is the trace: 9 headers, 0 lines Sip read: INVITE sip:00115601992@asterisk SIP/2.0 Via: SIP/2.0/UDP 10.8.210.147:5060 From: Giorgio