search for: extensions_addit

Struggling with an IP650 and 7 IP335s this morning. I have the following hints defined (courtesy of FreePBX 2.9): extensions_additional.conf:exten => 300,hint,SIP/300 extensions_additional.conf:exten => 301,hint,SIP/301 extensions_additional.conf:exten => 302,hint,SIP/302 extensions_additional.conf:exten => 303,hint,SIP/303 extensions_additional.conf:exten => 304,hint,SIP/304 extensions_additional.conf:exten =&g...

Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the file, changing its ownership...

...nge that line to p->subscribed = DIALOG_INFO_XML; On Tue, Jun 14, 2011 at 8:36 AM, Jeff LaCoursiere <jeff at sunfone.com> wrote: > > Struggling with an IP650 and 7 IP335s this morning. I have the following > hints defined (courtesy of FreePBX 2.9): > > extensions_additional.conf:**exten => 300,hint,SIP/300 > extensions_additional.conf:**exten => 301,hint,SIP/301 > extensions_additional.conf:**exten => 302,hint,SIP/302 > extensions_additional.conf:**exten => 303,hint,SIP/303 > extensions_additional.conf:**exten => 304,hint,SIP/304 > e...

...have to re-enter everything. From: "Lachek Butalek" <lachek@gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Date: Thu, 8 Jun 2006 12:24:24 -0400 Subject: [Asterisk-Users] FreePBX 2.1.0: Manually rewriting extensions_additional.conf Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is...

...ed the AMP Digital Receptionist to make a simple voice menu: "Thank you for calling xxxx". I did this for both Normal times and After Hours times. It worked fine. I then went to the AMP Maintenance window, Config Edit, got the "phpconfig for Asterisk PBX" page, and selected the extensions_additional.conf page. On this page were the entries for the Normal and After Hours greetings. The initial greeting phrases were expressed in terms of statements like: "exten => s,n,Background(custom/aa_num)". It was easy to extend the greeting (for instance, "Office hours are 7-7"...

Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or A@H, there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs, which is all fine and good, unless you...

...:<br><blockquote class=3D"= gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-= left:1ex"> <br> Struggling with an IP650 and 7 IP335s this morning. =A0I have the following= hints defined (courtesy of FreePBX 2.9):<br> <br> extensions_additional.conf:<u></u>exten =3D&gt; 300,hint,SIP/300<br> extensions_additional.conf:<u></u>exten =3D&gt; 301,hint,SIP/301<br> extensions_additional.conf:<u></u>exten =3D&gt; 302,hint,SIP/302<br> extensions_additional.conf:<u></u&...

...s purpose , I added following in "extensions_custom.conf" file [from-pstn-custom] include => ext-local exten => s,1,Answer() exten => s,n,Background(enter-ext-of-person) exten => s,n,Wait(2) exten => s,n,Goto(ext-local,s,1) the "ext-local" context is in "extensions_additional.conf" Now what happens is that , I the call from pstn get answered , then it ask for the extension , but after enering the extension no , I find the call get hung up The asterisk CLI message is as Starting simple switch on 'Zap/1-1' -- Executing Answer("Zap/1-1&qu...

How can i activate "ChanSpy" to spy on a dedicated extension? I see the following in "/etc/asterisk/extensions_additional.conf" [chanspy] include => chanspy-custom exten => 501**,1,Chanspy(801) exten => 501**,n,Hangup exten => 502**,1,Chanspy(802) exten => 502**,n,Hangup But when i try to call "501**", it doesn't give any response. Thanks. Torintino __________...

Hi All; Using the FreePBX, after I added the extension from the GUI, I discover that it is automatically added in the extensions_additional.conf in the context [ext-local] and [from-did-direct-ivr] How I can change these context name? I need to determine this. How? Regards Bilal

Hi, I'm struggling to get busydetect to work. I'm using asterisk 1.2.1 and a digium TDM04B (4 port FXO) card. I've set busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've modified zondata.c with a busy setting of 620+480, 300/200 which is the busysignal received from Korea Telecom. Asterisk isn't detecting the busy signal and doesn't hangup.

...aller presses 2 it goes to ext201 etc etc... Now I have created a third option that when the caller presses 3 it must play a sound and hang up. No rocket science yet. When adding the details in AMP for when caller dials 3, I have referenced it using 'custom-myapp,s,1', and if I go to 'extensions_additional.conf' I see the following line under the rest of menu item info that was created : "exten => 3,1,Goto(custom-myapp,s,1) ;" and in the extensions_custom.conf file I have [custom-myapp] exten => 3,1,SayDigits(1234) exten => 3,2,Hangup() But when you call and press...

...p://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail If we are using 5 digit extensions (10102: 10 for the company, 102 for the extension), where can we put something so that "102*" goes straight to voicemail without waiting while the extension rings? Here is what we have in extensions_additional.conf: exten => 100,1,Goto(ext-local,10100,1) exten => 101,1,Goto(ext-local,10101,1) exten => 102,1,Goto(ext-local,10102,1) exten => 103,1,Goto(ext-local,10103,1) Would something like this in extensions.conf work? exten => _XXX*,1,Voicemail(u${EXTEN:1}) Where would be the...

...… I think I don't have to edit the first part of the conf file (" same = n,Dial(whatever) "), you just mean the second part of the code is executed by "n,Hangup"? Then I have to add the second part to extensions_custom.conf, context [macro-hangupcall-custom]? (I cannot edit extensions_additional.conf where're the other settings/it doesn't make sense, because FreePBX overwrites it.) Probably you mean h,1,DumpChan() instead of h,1,1,DumpChan()? last line: same → same code like in the upper line up to "h", so it's "exten  = h,…"? Kai

I have a little confusion about the general settings (other than the register values) in the SIP General area. I understand that for examle in a SIP context like [FWD] or [BROADVOICE] the entries in those areas are ths settings that take effect in any communication woth FWD and/or BROADVOICE. However, I'm confused as to the purpose of the "general" settings -- to what or which

...it, you would put in something like: [from-pstn] exten => s,1,Dial(YourInternalExtension,15) 'Dial whatever your internal extension is for 15 seconds exten => s,2,Hangup() 'Hang up the line if nobody answers. You could put in a goto to fire the call to the [from-internal] context in extensions_additional.conf so it can have voicemail logic. I found that the best part of AMP is they have a really really good extensions.conf you can use as a template to make a customized dialplan. Starting from the base AMP extensions.conf and extensions_additional.conf, I have modified my dialplan *way* beyond...

...at David Shaw posted earlier today. 0.5 installed OK, but mine just with one X100P clone. Default config files, edited zapata.conf per the FAQs so it includes the line channel => 1 without the semicolon. Any outgoing call attempt returns "all circuits are busy" announcement. Edited extensions_additional.conf so the outgoing macro calls channel 1 instead of group 1, no change (except the CLI reports it's dialing Zap/1 instead of Zap/g1) Console reports that it's executing Dial/Zap/g1/2345678 where that's the number I dialed, and then "everyone is busy/congested..." z...

...keep it as '/var/run/asterisk, my DID phone will work with stanaphone (in which i'm crapping in my pants if they'll exist cause they never return emails). Though CLI won't work. if i do '/var/run', my DID won't work, but CLI will... I've tried just coping over the extensions_additional.conf and sip_additional.conf files from my old setup to my new one, and that didn't work. Maybe I should just install my previous version. Are there QoS differences though? I'd rather not regress if that were the case. -- Anything else, let me know. - Dominic "It is not the...

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

...ux: I'm unable to capture a log (or perhaps it's captured and I'm just not aware where).. Asterisk@home seems to dialout and receive calls using Macros.. I suspect it's a clever way of managing the setup, but I'm not sure where the various portions of SIP.conf, extensions.conf, extensions_additional.conf, extensions_custom.conf or indeed oh323.conf. - are relevant. I mention this because I realize no one will be able to help my "specific" problems unless I put up the confs.. but I have not added anything related to oh323 at all (excepting working on oh323.conf which is not mea...