similar to: included context not being prioritized properly

Hi list, I am still a newbie and struggling with tweaking the dial plan to my requirements. I have tried googling for this specific problem, and apologies if I have overlooked the obvious answer already. If you could please be so kind as to point me in the right direction, that would be most appreciated. What I am trying to do, is get rid of the initial "+" in phone numbers coming in

Given the following SRV records: _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 sometimes.sip-happens.com. _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com. Why is asterisk (1.4.17) not honouring the priority and not failing over to using other records when a connection fails? For a given call to tollfree.sip-happens.com ares.sip-happens.com was chosen

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

Hello All, I was wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB? hyperion*CLI> show dialplan app-blacklist-add [ Context 'app-blacklist-add' created by 'pbx_config' ] '1' => 1.

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Has anyone implemented "hot seating" in any neat way? This where people can log in to any phone in the company and have their calls/voicemail come to that particular handset.....

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm)

When I first to the "add extension" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by

Can someone refresh my memory how to backlist caller ID in asterisk 1.8? I had it working in ver. 1.4 but in 1.8 it changed. -- Joseph

I have a TMD400 card installed in a PC with one fxs (installed in slot 2) and two fxos (installed in slots 3 & 4).? fxos work fine but I am unable to get a dial tone for any devices connected to the fxs.? I have correctly connected the power supply to the card and I have even tried moving the card from slot 1 to 2 on the board. Below is from the console when I try to route a call from FXO on

I have a [start] context that all my inbound and '0' calls are routed into. Because of the way I want to set my system up, I want to prompt the user to enter a 1 if they know the extension, or a 2 for a directory and nothing else. It works, however there is a 5 to 10 second delay after enter the 1 or 2 before the system responds. I have read over the wiki on how asterisk handles digit

Hello Asterisk users, Can someone thwack me with a clue stick please? I am following the Asterisk TFOT book Dial() example trying to get the limit and announcements to work as per below. These settings seem to have no effect. There are no warning messages after 4 minutes or every 30 secs thereafter and the call lasts longer than 5 minutes. gunner*CLI> show dialplan [ Context

Greetings, Running CVS HEAD about 3 weeks old, I have been beating my head trying to get this to work properly.. Or at least figure out what's going on. Maybe I have done things wrong... I have created a 'catch all' extension at the end of our last context where all phones & voicemail extension exist. This catch all is included in all and works quite nicely except when voicemail

Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING, TWILIO)). It does not work and NO error message in CLI. I have also tried http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ since I first emailed this group, but that does not seem to work either. Here is my log: [Dec 2 15:09:28] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: Call from

I was wondering why *foo at default should match '_*[0-9a-zA-Z].*0.' in 1.6.13. Who is making the parse error, * or me? CLI> dialplan show *foo at default '_*[0-9a-zA-Z].*0.' => 1. NoOp(${EXTEN}) [pbx_config] 2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config] 3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config]

Asterisk isn't playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory. Asterisk just plays the "The person at extension..." message, not the greetings I have recorded. Thanks -- asterisk*CLI> show dialplan macro-stdexten [

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.

Before I play around with this again in 1.2.1, regexten is still essentially broken, correct? The misconception seems to be that it allows you to execute a command upon registration from a SIP UA. Even the O'Reilly TFOT book erroneously states that this is what it is for. After reading the developer discussion though, it definitely seems to be broken. Is it fixed yet? Doug.

i have a box running debian sarge with asterisk installed from distribution packages: CLI> show version Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by kk@nyx on a x86_64 running Linux I have managed to configure a simple dialplan (the PBX task is quite simple as this is a small office with just a few phones) I have one Zap (PSTN) line connected to it and one SIP to a local provider. After

Hello, I just can't seem to get the MusicOnHold function to work out ok. I' managed to get the MP3Player app to work out fine, but when I run the MusicOnHold all i get is siliece. I can see that Asterisk executes mpg123 properly (I think) #ps axuww|grep mp gk 4383 0.0 0.4 3736 552 pts/4 S 15:06 0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 sample-hold.mp3