similar to: FreePBX 2.1.0: Manually rewriting

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

So I took a chance with an X100P knock-off on eBay. I'm running Asterisk + FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and kernel 2.6.16.16. Everything has been fine up until now. I compile the 1.2.5 Zaptel drivers without a problem, get the udev configuration in, modprobe zaptel, and finally modprobe wcfxo. At this point, I get the message: ZT_CHANCONFIG failed on channel

My mission is to get one * box to dial another * box' extensions. I have set this up previously without any issues by making a simple IAX trunk/extension pair on the two boxes and create a dial plan with a prefix like 9|XXX to select an extension on the other box. My problem is that I now have to do this with extremely restrictive firewalls thrown into the mix - firewalls I have no control

Situation, simple home setup: * Trixbox 2.0 * Feature Codes installed * GNet PA-168V based ATA * Cheesy cordless analogue phone >From what I gather, dialing *70 from the handset should activate Call Waiting. All it seems to do is change the message "The person at <extension> is on the phone" to "<ring> <ring> The person at <extension> is

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 Daniel, Am 27.03.20 um 09:24 schrieb Administrator: > Hangup is h extension. your macro will never be executed. Solution: > > same = n,Dial(whatever) > same = n,[...]) > same = n,Hangup > > exten  = h,1,1,DumpChan() >  same = n,System(/home/asterisk/bash_test) I don't really understand your code… I think I don't have to edit the first part of the conf file

Hello- I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls per day max. I used 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

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,

can someone overthere help? the server specs are as follows HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM, running fedora core 3 asterisk-1.2.5 ss7-0.8.3d. using sip as advised to receive calls from another gateway in US. using g729 in transcoding way. however, I noticed the call hit the 51 active calls which is 102channels, I run "top" to check the system resources usage

I've gotten nothing from http://bbs.atcom.cn on this so far. Perhaps someone on the list has experience with this. Is there a way to get MWI support for PA168V-based ATAs? Apparently some IP phones based on the PA168V chip has this support already (Atcom AT-320 for example) by configuring Asterisk with 'mailboxdetails=yes' in iax.conf. On my ATA, however, it does nothing. Any

Hi All After lots of try I was successfull in connecting to PSTN to make and recevice calls , I used AMP for this purpose , now I wanted to try out this Asterisk server answers the call , ask for the extensions and then after the extension entered the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3 I did some configuration for my

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

Hi, guys. May be someone could give me advise? I am trying to find good VOIP provider ONLY for OUTGOING calls with low per channel cost and cheap rates on Eastern Europe, Turky and xUSSR. Should support g729 or g723 codecs, SIP or IAX connectivity. -- ========================================================================= = Best regards, Nikolay Pavlov.

Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With SIP 3.2.X firmware (available on the Polycom download site) and Asterisk 1.6.1, Polycom phones now support a full featured BLF showing statuses of Ringing, Inuse and Online and one touch directed call pickup. On the asterisk side all that needs to be done is to add a hint to the extension and enable directed pickup.

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.

Greetings *`s, I am having what appears to be a small problem, but the frustration is erally getting to me, what am I doing wrong here ? I used AMP to set up a custom menu, so if caller presses 1 it goes to ext200, if caller 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

r> <h2>Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010= ) </h2>With SIP 3.2.X firmware (available on the Polycom download site)=20 and Asterisk 1.6.1, Polycom phones now support a full featured BLF=20 showing statuses of Ringing, Inuse and Online and one touch directed=20 call pickup. <br>On the asterisk side all that needs to be done is to add a hint

Hi, Have looked around for info about this: <http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com> http://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

Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I

Is there a configuration difference for clone X100P cards versus "compatible"? I have a similar problem to what 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"