similar to: Which channels are required for FAX, GTALK and Jaber

Hello; As I am using vicidial and its asterisk version which is 1.8, I need to know the required channels to be existed so the asterisk will support fax, SMS, gtalk, Jaber? In other words, how I can know that it is enabled in this asterisk (actually it is 1.8.21-vici)? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've just got Uverse installed. I had dsl, but ATT insisted I couldn't keep my old dsl, but had to switch to Uverse internet - vdsl. My setup: linux box as router : 10.10.11.252 asterisk box: 10.10.11.180 10.10.11.252 is multihomed and connected to the Uverse Residential Gateway. I've set it up as DMZplus, and it shows the public ip address as eth1. I can ssh into the

Hi, I have some questions about tinc. " *Automatic full mesh routing*Regardless of how you set up the tinc daemons to connect to each other, VPN traffic is always (if possible) sent directly to the destination, without going through intermediate hops. " 1- To have client to client direct connection and don't have traffic through the server should I have port forward at the NAT? Or

Hi, Last week I finished installing a small network in a private school : one server (an old IBM X225), seventeen desktops (Fujitsu Siemens PIV 2.4 GHZ, 512 MB RAM, 40 GB HD), all running CentOS 5.5. One extra machine is acting as a router, in that it is installed between the DSL modem and the network, with two Ethernet cards, and it's taking care of DHCP, DNS, NTP and also acts like a

Hi Guys, We are getting a strange issue on certain polycom phones, sometimes when a call comes in it just "flashes" to show there is a call but does not play any sound. This problem is very intermittent and happens to maybe 2 out of 10 calls. Has any else experienced this before? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an asterisk 1.8.x box that intermittently returns a 401. Calls come through the same peer all the time, from the same carrier. However intermittently the asterisk box returns a 401. Below is the output of a failed call (1st) and a successful call (2nd). I can't see any difference until we get to these lines. Bad call: --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT)

Hi Charlie, Thanks for your reply. I've modified config.mak according to README. I just want to output the xapian dll for php, so i removed other config such python, csharp, etc. I download source code from php site & set the path in config.mak accordingly. It still reports this error. I'm confused which config i missed. Thanks. Jaber 2010/6/3 <xapian-devel-request at

Hello; How I can compare between Asterisk 1.8 and 11 with reference to the following points: 1) SMS. 2) gtalk and other social media. 3) GUI. 4) Any main difference? Regards Bilal

On Mon, Mar 9, 2015 at 9:40 AM, David Wessell <david at ringfree.biz> wrote: > Welcome to our hell. > > We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally > got Polycom to issue a hotfix firmware version. I'll be happy to share it > with you offlist, just email me. > > Officially Polycom will fix the issue in 5.3 in a few months.. > >

Hi there I have the following setup Home - Main Tinc server with public IP running on PfSense work - tinc client running behind a CISCO ASA firewall with public IP running on Windows 10 offsite - tinc client running on tomato router behind a double NAT Home & offsite connect & i can see all PCs & devices & connect to them easily, on either side work to Home or offsite connects

Dears; The problem was related to something else. The Digium card has two PRI ports, actually to get it UP, I have to configure the two ports and both of those two ports to take the timing from span 1. Why this, I do not know ! Although I am using only one E1 connected to span 1, so why I have to configure the other span !! After configuring the second span, so now one D channel for span 1 is

Dear; Thanks a lot for guiding me. Is it possible that the installation libpri-1.4.11.5 newer than the libpri-1.4.11.5-patch? Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for the libpri-1.4.11.5): libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch It gave me that patched detected as shown below (example of one file, and I got same for other files): patching file

On Mon, Jan 05, 2015 at 08:10:48AM -0700, Lance Fredrickson wrote: > Allowing tinc to run without RSA keys is a very big > bonus for us embedded device users. We have a limited amount of nvram space > for storage (tomato firmware). RSA keys took up quite a bit, so being able > to run using only ED25519 makes tinc very appealing for this platform, and > saves tons of space for more

Greetings- As many of your are Polycom "experienced", I was hoping some kind soul could provide direction on a specific issue. On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an instance where, using intercom/paging functionality of FreePBX, I need to override an end user's 'Do Not Disturb' selection on the handset. By default, DND simply

Hello All; Wanpipe is working only with sangoma cards so it does not work with digium cards? Also, who is better: to have echo canceler built in with the hardware or using olsec? Regards Bilal

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW?

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

I have a curious problem with an old WRT54GL router, which I use as a WiFi access point on my LAN: Internet->ADSL modem->CentOS-7 computer->WRT54GL router The router has always had a slight problem of losing connection every so often - it used to be every couple of days, but recently it has become much more often. My cure was always to disconnect the power from the router for 10