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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

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

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

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

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'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 :

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

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]

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

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

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,

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

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

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

Dear Alex; Thanks for your kindly reply. Please explain for me what do u mean exactly in "a la" in the following sentence u wrote it below? " in SIP, this can be done via "re-INVITEs" a la the canreinvite= option for SIP peers in sip.conf" Another thing, do u mean that it is easier (better) if we need H.323 endpoint to talk with SIP endpoint then we use full