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Does anybody know if there is an app that will cause similar to occur on users PC? I have a scenario where users will have snom phones on their desks. Ideally when their phone receives a call I need to popup a web browser with a specific url. Any ideas appreciated. Neil on 5/7/05 10:52 PM, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> wrote:

> -- Started music on hold, class 'default', on > SIP/193.111.200.67-0815c790 > -- outgoing agentcall, to agent '1001', on 'Local/201@sip-0add,1' > -- Called Agent/1001 > -- Executing Dial("Local/201@sip-0add,2", "SIP/101|20|tr") in new > stack > -- Called 101 > -- Agent/1001 is ringing > --

On Mon, 13 Jul 2020 15:44:12 -0400, Matthew Fredrickson wrote: > > On Mon, Jul 13, 2020 at 2:34 PM Saint Michael <venefax at gmail.com> wrote: > >> > >> There is a big confusion here about Stir Shaken. It is NOT a provider issue. Un fact, all providers are whasing their hands and modifying their swihtches to pass-through the Signature. They cannot sign the call

In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law. We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are the only ones who don't need to receive the calls in our servers to sign them. We do this over a MySQL call, easily connectable to Asterisk via

I wonder if anyone can help me? Am trying to terminate to H323 Vegastream. I'm using OH323 with little success. I can dial out and answer but voip end just keepings ringing and ringing. Thanks for any help. Neil Config file: [general] listenAddress=ALL listenPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no jitterMax=100

Hi all, Am looking for everyones advise/recommendations. I have am setting up a network of both office and home based workers. The office workers will be on the same network as the Asterisk box so no NAT hassles there. However, the home workers are on their own DSL connections so I imagine that most will be behind NAT. I had hoped to use Sipura SPA-2000's but I presume I would need either

Please could someone help? I need a way of knowing the DNID for a call that is in our queues. I've tried setting varies variables with different _ & __ before them but none are inherited by the Local/ call that is placed from Queue. Any suggestions appreciated. Neil

I have a scenario where I have several agents in a call queue. I need to prevent certain agents receiving calls from certain callers. These calls are all calling a different DDI to they are easy to identify from that point of view. However, I need there calls to have the same priority as all other callers hence I want to put them all in the same queue. Is there anyway I can retrieve the DDI that

Please can anyone help? I'm trying to setup our asterisk system to allow any user to place a call regardless of whether they are registered or not. Ideally I want to place unregistered callers into a specific context but authentication is based then on the number they are dialing rather than having them registered to the server. Hope this makes some sense! Cheers, Neil

Please has anyone experienced a bug with queues in Asterisk? No matter what settings my queue always thinks it got agents available in it. Plus if I take all the members out, then calls don't join the queue even though I've specified join-empty. Any advice? Neil

Well the subject line probably says it all. I have a Quintum D3000 which I'm supposed to be getting connected up to our Asterisk system. No matter what I try, neither username or authuser config works. I've also tried md5auth and it still refuses to register. Any one have a config they could share with me? Any help would be much appreciated. Neil

Please has anyone on this list had experience with getting Quintum equipment to talk to Asterisk? Specifically a D3000 in my case. It is refusing to register and I'm out of ideas. Any help appreciated. Neil