Google is your best friend when looking for this type of assistance my friend.
try callcentric vonage packet8 for reliable retail DIDs.
Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Sat, 1 Oct 2011 00:51:59 +0530
Subject: [asterisk-users] USA Did required Hello members,
I am looking for USA incoming DID which can be registered on softphone/IP Phone/ Pap2 devices. The DID will only be required to receive inbound calls and no outbound calls.
We have setup Asterisk HA, basically what we have is:
Virtual IP (for asterisk 1 and 2) 192.168.2.6
Poly IP300 192.168.2.13: The polycom is configured to use whatever machine is
running on VIP 192.168.2.6 HA is configured that if ast1 goes down, ast2 takes over the virtual
ip. When only
one machine is up, the server handles both the incoming and outgoing. When both
machines are up, incoming is handled by ast1 and outgoing is handled
by ast2? Again,
On Eweek it has been reported that: "Engine Yard has delivered JRuby on its cloud platform to enable Java developers programming in Ruby to innovate faster and scale their apps. Engine Yard has announced the general availability of JRuby on the Engine Yard Cloud. With JRuby support, the Engine Yard Platform-as-a-Service (PAAS) brings together the combination of Java performance and Ruby agility. JRuby, a Java implementation of the Ruby programming language, is a popular open source package that enables Ruby applications to run on the…
Friends, SIPit is an event organized by the SIP Forum and partners. It has been running for 15 years twice a year, making sure that SIP clients and servers interoperate. By testing, we also find issues with the myriad of RFCs in this area and correct them. Testing interoperability is important. The first time I brought Asterisk to SIPit in Stockholm many years ago I was terrified. The SIP stack back then was, well, peculiar. It worked with some SIP phones for basic calls, but not much more. During the tests I learned a lot, got a lot of help…
This is just a speculative shot in the dark, but remember that the domain of the From URI is important, and that the authentication "realm" (domain) is part of the authentication credentials. So, what you have in your 'fromdomain' and 'host' settings on the peer does matter.
I have a pstn line can have the local, STD and ISD
capabilities. My local number is 91471-2527XXX and the region is India. I
would like to use the number for all possible calls ( local, STD and ISD
call facilities to Land line and mobile phones) through an FXO card
configured in asterisk freepbx. Can anybody help me to create an outbound route and inbound route required
in freepbx for the above requirement ?
Hi, We have got a new PRI card at one of our Office locations and now I need to install the the device on a remote server. Is there any way to know if the device is loaded already. When I give " cat /proc/zaptel/* " it returns the following. # cat /proc/zaptel/* Span 1: WCT1/0 "Wildcard TE122 Card 0" (MASTER) B8ZS/ESF RED IRQ misses: 2 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In…
I could have sworn this working at one time...
But it doesn't look like any of the functions provided by features.so is
working for me. (one-touch monitoring, attended/blind transfer, etc) I've (re)loaded features.so, as well as bridge_builtin_features.so. The config file looks sane. What else should I try? TIA,