search for: phone5

...aviour? - at least fake some '***' in the password fields... Asterisk also had me chasing my tail - it never mentioned anything such as 'SIP Registration password is incorrect' - I got one.. chan_sip.c:7231 handle_request: Failed to authenticate user "Phone Five" <sip:phone5@asterisk.mydom.com;user=phone>;tag=fjhgkjhgkhjlk (OK - "failed authentication" - but something about the password would have been better) and got lots of... chan_sip.c:7588 handle_request: Registration from '<sip:phone5@asterisk.mydom.com;user=phone>' failed for '19...

...out hangup... -- Hungup 'Zap/4-1' If I change to [channels] context=pstn group = 1 signalling = fxs_ks callprogress = no channel => 4,3 then when dialing out as soon as * realizes the zap channel is available it signals back call success.. -- Executing Macro("SIP/phone5-4313", "dialout|93053486215") in new stack -- Executing Dial("SIP/phone5-4313", "ZAP/4/93053486215") in new stack -- Called 4/93053486215 then this happens immediately -- Zap/4-1 answered SIP/phone5-4313 It does dial out and eventually it is connect...

Hey, how can I implement a dial plan like the following: incoming call: 1. ring phones 1,2,3 monday to friday between 9:00 and 20:00; if no answer after 15 sec also ring phones 4 and 5 2. ring phone 1 monday to friday between 0:00-9:00 and 20:00-24:00; if no answer after 20 sec also ring phones 2 and 3 3. ring phone 1 saturday and sunday all day I do not need a in detail answer for each of the

...essage (in * CLI) that particular extension does not exist in a given context Here are my contexts definitions: [from-sip] exten =>101,1,Dial(SIP/phone1) exten =>102,1,Dial(SIP/phone2) exten =>103,1,Dial(SIP/phone3) [iax-user] exten=>201,1,Dial(IAX2/phone4) exten=>202,1,Dial(IAX2/phone5) If I try to call from IAX2 phone to say ext 102, I get "request '102@iax-user' does not exist" I have tried to include iax-user in from-sip and I can make calls from SIP phones to IAX2 ones, but not the other way around. Now for an interesting bit. If I include "from-sip...

Hello guys, I ask you to share your experience with your BudgeTone 100.... I have my asterisk @ work and I've bought a GrandStream BudgeTone (SIP phone) and I usually use X-Lite I have plugged my BudgeTone into my home network because I want to be called even at home. I succeed to register my X-Lite with Asterisk from home but I can't do that with my BudgeTone. (I don't know

...calls. Illustration: (hope it don't get messed up) Incomming Call --> My Company --> Group1 --> UserA-phone1 --> UserA-phone2 --> Group2 ---> UserB-phone3 ---> UserB-phone4 --> Group3 ---> UserC-phone5 ---> UserC-phone6 So what I want per group level is that only one user (phone) is active at the time. And if all of the groups are busy, I want to send the caller to voicemail. Everyone can call out at the same time, but it must update the group count the phone belongs to. The main p...