similar to: Conference bridge.

different stand alone linux server which act as my routers. Here is a picture showing the output from the CISCO switch going to the two linux servers: http://www.grmtech.com/blog/wp-content/uploads/2009/02/cisco2950-24ports-farleft-two-output-300x89.jpg My questions are: 1. The black wire coming into the Mc Manstel box is that a fibre optic cable ? 2. What is the Mc Manstel box doing ? 3. What

Hi, I still have the same problem trying to configure ITSP failover in extensions.conf for a connected PRI. Any comments thoughts or direction would be greatly appreciated. I sympathize with wanting inbound DID failover. If we have a client with multiple DIDs we will spread them across two or three ITSPs so that all inbound connectivity will not be lost if one of them has an issue. I

Bourvine, > > So, why won't we save the big bucks we pay them, hire two professionals > (who cost less) and support an open source code by ourselves? This way > we depend on ourselves only. > > > > Thanks, __Yehavi: I remember hearing University of Pennsylvania have been using Asterisk for sometime. I am not certain where I came across that

On Tue, 27 Nov 2007, Alex Balashov wrote: > > Our provider gives us four PRIs as a trunk group hunt group. Meaning, the > provider's switch will cycle through B channels in span 1, 2, 3, ... until > it finds one that is available. > > I have moved spans 2-4 onto another machine. But we have one remaining > box with a PRI full of calls and I don't know what to do

help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it

In the Perl FastAGI API, how does one set the port the service runs on? [root at donkey queue_login_arbiter]# perl arbiter_agid.pl 2007/12/06-17:16:27 Evariste::QueueMemberArbiter (type Asterisk::FastAGI) starting! pid(31737) Port Not Defined. Defaulting to '20203' Binding to TCP port 20203 on host * Group Not Defined. Defaulting to EGID '0 10 6 4 3 2 1 0' User Not Defined.

Greetings, As a developer and consultant who spends considerable time on projects involving the fusion of Asterisk and products derived from the SER ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have found that there is a great volume of interest in this topic on the mailing lists associated with all communities involved, but a comparative lack of focus that results in

I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work. -----Original Message----- From: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: 11/29/2008 1:13 PM Subject: asterisk-users Digest, Vol 52, Issue 81

I made a little write-up that attempts to synthesise a lot of the information out there about how to get HylaFAX working with Asterisk by way of IAXmodem for inbound faxing: http://blog.evaristesys.com/?p=24 Of course, there are bound to be some things I've left out or are grossly in need of correction. So, before I link it off the voip-wiki I am extremely eager to solicit the input of

Hi all, I am having a very strange problem. I am terminating a PRI (5ESS switch type, national plan, 23B+1D (24)) into a Sangoma A104 and am not able to produce any audio heard on the PSTN end of the call. Not sure what's wrong - the card worked before under a Trixbox setup. I'm running kernel 2.6.19 (tried 2.6.24.3 but had to downgrade as wanpipe stuff would not compile), zaptel

So, is there anyone out there that provides rather generic but comprehensive CNAM-style directory services via SIP, to end-users? So I can put names to my calling numbers? Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

I have two phones on a network behind NAT. Enabling canreinvite=yes on the Asterisk server allows them to talk to each other very effectively through the local network. Unfortunately, calling any outside destinations yields one-way media issues where the far end can hear me but I can't hear them, probably due to lack of an ALG on the NAT router that understands the SDP negotiation of the

extension from another phone, it should place you in a position to listen in on a bridged call (a call whose media runs 'through' Asterisk). -- Alex On Thu, 21 Jun 2007, Carlos Garcia Mujica wrote: > How can I use the Asterisk command ChanSpy If I need to spy on a call? > > I already added the function to the extensions.conf, and I get the beeps, > but then what do I do??? I

Dear Alex; Thanks for your kindly help and answer. The question here is: how asterisk will be able to receive calls at two network cards where each network card has a different IP address. Maybe we need to know if asterisk is doing a hear on the ports only without caring for IP or it is doing a hear only on the IP:port? Any advise? Bilal, There is no technical difference, from Asterisk's

Greetings, I have a problem getting Asterisk registered as a UAC against the MetaSwitch call agent, because the customer insists on running it on a NAT'd box. Thus, the Contact: field in the REGISTER request betrays the private IP address of the Asterisk box, but the source IP of the message is a public one. Most registrars don't have a problem with this, including Asterisk. However,

Greetings, I have a very basic equal-weight ring-all queue set up in queues.conf: [sales-queue] ;music = default strategy = ringall periodic-announce-frequency = 20 announce-holdtime = no timeout = 15 maxlen = 0 member => SIP/1xxxxxxxxxx at junction_networks,1 member => SIP/1xxxxxxxxxx at junction_networks,1 member => SIP/dude,1 member => SIP/homie,1 member => SIP/fellow,1 But

Has anyone figured out a way to instantaneously swing over PRIs bearing calls in progress to another media gateway without dropping them? Obviously, this would require a DACS of some sort. But I am thinking that it is possible to swing T1s over in a DACS without actually causing the endpoint to reframe as long as the other endpoint is kept in sync. So, it'd be nice, for example, to bring

I'm sure this has been asked a million times before, but is there an easy wa to have Asterisk register more than one (distinct) contact binding concurrently? The goal is to have two phones register with the same credentials from different locations and consistently and reliably ring on inbound calls, irrespective of their registration intervals and so on. -- Alex Balashov Evariste Systems

Is there a way to dynamically alter the sip.conf properties of a SIP peer in runtime without doing a SIP reload? I am specifically thinking of enabling reinvites for users dynamically based on whether they are registered from a public address. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

As I said, please keep discussion on list. asterisk at opensourcesolution.in wrote: > hi all, > > first of all i appologise for sending u pvt email. i have installed > asterisk on Centos 5.3, plz open the attachment in which i had drawn a > tolpology. i had installed one asterisk machine and two windows machine. > now i want to install softphone in both windows machine.