similar to: CNAM.

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

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

http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/ FUD? Interesting? Boring? New news? Old news? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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

I've got the ztdummy kernel module loaded and seem to have all the desired prerequisites in place, but Asterisk never seems to compile with MeetMe() application support enabled, nor does there appear to be a module I am failing to load that would contain this application. Is there something really obvious I am missing? Thanks, -- Alex Balashov Evariste Systems Web :

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

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

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

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

please unsubscribe Moshe Wahrhaftig IT Manager Talk'n'Save Israel: 02-655-0313 Cell: 052-2771738 USA: 516-204-4444 -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Guillermo Rodriguez Sent: Monday, September 03, 2007 10:51 To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users]

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

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

Hi all, This may be a really stupid question, but, what preset global dialplan variables can I use to determine the calling leg when using Dial()? Say I have phones (SIP peers) originating calls out of the same context, and I need to set the ANI differently depending on who is calling out in order to make it consistent with their inbound DIDs? Asterisk appears to provide a wealth of variables

Hi folks, I know this isn't an Asterisk question, but I'm really desperate and wondering if someone could help me. I apologise for the off-topic post. Cisco phones connected to CallManager can forward calls. But when they do, CallManager conserves the originating caller's ANI in the new leg that is built. I cannot find a way to get it to rewrite the ANI to be that of the phone.

Inspired by AG Projects' Adrian Georgescu's post of Eric S. Raymond's classic "How to Ask Questions the Smart Way" to the OpenSIPS-users mailing list[1], I'm going to repost it here: http://www.catb.org/~esr/faqs/smart-questions.html As Adrian said, "This a good read for those who show up on mailing lists without any guidance about how to ask the right

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,

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 with them; the box is failing, but dropping them by simply yanking the PRI is

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

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

Any recommendations for an affordable SIP conference bridge unit? I mean one that isn't crappy; something where the duplex and cancellation functions that are traditionally built into such devices actually work. I am referring to something that looks like this . . . http://www.hardware.com/products/cnet/I212272.jpg But not necessarily that. -- Alex Balashov Evariste Systems Web :