similar to: Upgrading to 1.2.5?

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

I am testing 1.4 branch on OSX (10.4.8) and although it's running and passing calls ok, I am still not able to connect using asterisk -r. When I do open a CLI using asterisk -r, it appears to start up normally, but then is non responsive to commands (exit works though?). I am currently running SVN-branch-1.4-r46716. Any ideas on why this might be, or how to figure out how to fix it?

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 All, I'm thinking of creating a new asterisk server using the latest 1.4 stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its been a while!). My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 loaded) and to support them better, I remember compiling in a skinny(?) driver to replace the (from what I could tell) basic in built sccp support. After

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

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

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

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

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

I have been trying to get a DIALSTATUS output from a call started with originate. I searched a fair bit and have found several references to using local channels to do this. However, I could not find enough of the specifics to get it working myself. What I need to do is dial a zap channel and run various scripts if the channel is answered, busy, no-answer,etc. Here is the dial plan I am

Whoo hoo! I just received my WIP300 from voipsupply. I have to let it charge before I can play with it. A few quick comments: - I started a Wiki page at voip-info to post issues, firmware news, etc. I really like the wealth of info on the GXP-2000 page, so I wanted to start something similar for this phone. http://www.voip-info.org/wiki/index.php?page=Linksys%20WIP300 - My kit

Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>

http://grandstream.com/BETATEST/HT488_496_386/

XCP Users: I received an email from Alberto Gonzalez Rodriguez who created OpenXenCenter/OpenXenManager and is now working on OpenXenWebManager. Info to access the tool is: steps to test are: download from http://bit.ly/bres2U uncompress: tar xvfz xenwebmanager_rev24_full.tar.gz cd xenwebmanager python frontend.py and with a browser open http://localhost:8080 (or http://ip:8080) you need

Hi folks, I''m new to this... so hopefully someone can help me. I have a few objects... Property and Address and Landlord... where a Property has an address and a landlord has an address also. I''ve modelled this in the db with the properties table having an address_id and the landlord table having an address id. My rb looks like: class Property < ActiveRecord::Base

Hi there, I have tried compiling OpenSSH 4.3p2 using the following steps: Upgrade OpenSSL tar xvfz openssl-0.9.8a.tar.gz cd openssl-0.9.8a ./config make make install Upgrade zlib tar xvfz zlib-1.2.3.tar.gz ./configure make test make install Upgrade OpenSSH tar xvfz openssh-4.3p2.tar.gz cd openssh-4.3p2.tar.gz ./configure --with-tcp-wrappers --with-ssl-dir=/usr/local/ssl

Dear all, I am sorry in advance because probably my question was already discussed. I have installed an R version R-1.3.1 on Linux RedHat 6.2. As I want to install R-1.5.0, I have first followed a suggestion of Peter Dalgaard (mail in FAQ) in order to keep the version R-1.3.1 by renaming /usr/local/lib/R and /usr/local/bin/R (/usr/local/lib/R-1.3.1 and /usr/local/bin/R-1.3.1 )and then set

Hello, I've tried to make out of my samba 2.0.3 server a logon server. I' ve set domain logons = Yes domain master = No local master =Yes os level = 65 preferred master =Yes but: wins support = No I made a share as recommended [netlogon] and place there a logon.bat. The logon.bat contains net use z: \\servername\public for testing if the logon.bat is excecuted by login to the server.

We're setting up asterisk at the office (really doing some testing right now) and it is going to be hosted on a dual G5 XServe running OS X. We're an apple certified solutions provider, etc. so we want to build all our stuff on apple hardware and software. Anyway, the last sticking point is moh and meetme. Is there any solution to get moh and meetme working on OS X? Meetme

Sorry for re-posting this message - I am trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor. Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency. i.e a sentence which should finish in 4 secs finishes in much lesser time. Where can be the problem? any configuration issue? Thanks in advance. -------------- next