similar to: Still no CLI in 1.4 branch (OSX)

Probably just me being dumb, but I am trying to update my asterisk to the latest (1.2.5) When I go to my /usr/src/asterisk I type: make upgrade make install This seems to be doing it's thing, but when I type show version from the console (after a restart) I still get: Asterisk SVN-branch-1.2-r7231 built by root @ notdeadyet-imac.local on a Power Macintosh running Darwin on 2006-03-04

Hi Friends, I have successfully implemented Intercom, Voicemail and International dialing using Asterisk. Now I want to connect my PSTN Lines to Asterisk server. I have 3 PSTN number (lines) to connect to Asterisk. For this, I want to use Sipura SPA-2100. Is my decession is correct or not? Is there any disadvantages with this Sipura SPA-2100? Please tell me. Thank you. Regards, Chandramouli

Hi, I have a spiffy new gateway which seems quite promising. It's the Audiocodes MP114 FXS_FXO (2 of each). I have got it configured and working reasonably well, but have a couple of issues. 1) Asterisk 1.2.13 voicemail seems to be hung up on by the gateway after 10 seconds. This isn't asterisk saying it's quiet for 10 seconds, it's the gateway deciding it's time to go

I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight. I took the additional step of nuking my modules directory first... When I used the command asterisk -v to start asterisk, it seemed to go along and get to the point where asterisk is running(ie Asterisk Ready). At that point it was eating all available CPU. I went ahead and tried to register a softphone to it via IAX2, which

Ok, I am back from my thanksgiving holiday, and I find there was a big snow storm here in Seattle. Apparently during the storm there where multiple brown out/black outs. I have struggled since day one to get a high quality PSTN gateway configured with my very long loop and Mac based asterisk. I originally tried the HT-488, which had multiple issues, and was unacceptable. I then purchased

I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a

Martin: I had seen your other post and sent you a message off-list, but I never got a response. What do you feel is the most lacking that does not make it ready for a production enviroment. - I've been using a SIP deskphone in my office and usually some sort of ATA at my house, both as the primary phone. I've also had mobile phones from almost every carrier. Each one of these devices

Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want

Hi, I'm currently using EMA package to make clustering and heatmaps. The online doc concerning the package gives the following example code: data(marty) c<-clustering(marty, metric="pearson", method="ward") clustering.plot(c, title="Hierarchical Clustering\nPearson-Ward") which is working perfectly, However, when I'm changing the method to

Hi anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ? Actually my HandyTone 488 are connected to: wan port to my lan line FXO port are connected to my local analogic line i want that when a call in by my analog line, it's sent to my asterisk for other voip post can answer .. it's possible ? thanks bye

Hi there, been using NUT for ages and recently upgraded to a new server; while I'm waiting for a USB->Serial adapter for Solaris to arrive, I thought I'd play with the newhidups, as the APC UPS I have has a USB port and the new server doesn't have a serial port.... I've got Solaris 10 & newhidups working with the ugen driver up to a point, and that is a read

I need to recommend some backup options for a web server running CentOS 4.1. The client prefers using a tape drive as their backup device and has access to safe offsite storage. I was thinking of system backups weekly and differential backups nightly but don't know what software to recommend for the differential b/u's. For full backups I can just schedule a tar and compress using

Since upgrading to 7.2, I've noticed a random problem where I dial a number and hear all the correct tones in the handset, but the display won't show all the numbers I dialed. So you sit there waiting for the dialplan to kick the call off (b/c you heard the proper amount of tones played and think it's all good) but the phone is just sitting there b/c it somehow "missed"

Hey Guys, I''m having trouble with getting my draggables to go over the top of other items on my page. For instance, if I drag an icon from ''lower'' in the page to a ''higher'' point then it slides underneath it visually. I have set the z-index to a very high number and that doesn''t seem to have any effect. Are there other things I need to

I did a quick check between what I have set and your settings below. You can try the following and see if it helps In your endpoint: bind_rtp_to_media_address=yes With best regards Florian Floimair Innovation - Software-Development - VoIP & DevOps COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstra?e 51 Tel: +43-662-85 62 25 Fax: +43-662-85 62 26 http://www.commend.com Security

I know that the prototype library was paired with Rails but I''m curious if anyone has used the ajax part (of the libraries) with java servlets? If so could you please point me to some examples or documentation. Cheers, Marty

Hey Guys, I''ve just encountered this error in Firefox will I pulled the latest scriptaculous rc into project. Sadly, I don''t have a linkable page for you to look over at the point. My question is what is this error exactly? My code before this never got it but now it does. My few searches suggest it''s the browsers way of keeping infinite loops from happening

I just noticed, the calling device isn't even sending the early media video stream. It just sends an early media audio stream. Is there propably a change in the signaling needed? (On another P2P SIP Server the early media video works.) 2018-04-10 12:29 GMT+02:00 Benjamin Marty <benjamin.marty at gmail.com>: > Hi Florian > > I already have the external_media_address set in the

Heya! Thanks to Marty Haught, script.aculo.us (as of changeset 2281) now sports a new Control for horizontal and vertical sliders. See the functional test file (test/functional/slider_test.html) for information on how to use it. Basically, you do: <div id="track1" style="width:200px;background-color:#aaa;height:5px;"> <div id="handle1"

Hey Guys, I know this isn''t specific to scriptaculous libraries but it''s something that I''m exploring as I use the ajax functions in prototype. My application server is not rails but java and I was thinking of returning a javascript array in the ajax response. When I get back my response is it, of course, text and I''m not sure how best to convert it into