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I have installed the sipTAPI from http://sourceforge.net/projects/siptapi/ when I use user names like joash.herbrink in Asterisk, it is not working when I change the sip username to my internal extension, like 1006, it works fine. Anybody any idea as to why this is? met vriendelijke groet, Joash Herbrink Technical Consultant "Control the flow" De Kahuna groep helpt organ...

...at it won't startup properly. Several reboots and factory default reset later, it sometimes works. Has anybody successfully done a mass provisioning of sipura 841 phones, and got it working, even after reboots. If so, can you please ! tell me how you did it? Kind regards, Joash Herbrink -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060223/ec5b7978/attachment.htm

...<asterisk-users@lists.digium.com> > Message-ID: <376DE673C9E840767FBB0EA6@dhcp-2-206.wgops.com> > Content-Type: text/plain; charset=us-ascii; format=flowed > > > > --On February 3, 2006 3:56:21 AM +0900 Vic <svictor@yahoo.co.jp> wrote: > >> >> Hi, Joash, >> >> thank you for your email. I was very relieved to hear that someone was >> already doing this. >> >> Can you please tell me more about your test? Why did you test it in a >> first place? >> >> For me, we need to come up with a system that needs...

I know the subject of faxing has been covered in some detail, but I was wondering if anyone has a hardware configuration similar to ours that has faxes working successfully and would be willing to share any settings/insight. We are unable to fax reliably with a Sipura 2100 connected to Asterisk. We do not route calls over the Internet and our network has very low latency. The Asterisk

...e. It works great with asterisk (specially the presence option, so agents can see whether somebody is actually ready to take a call). In combination with sennheiser headset CC series, I have had no complaints. We also use a tapi to make automated dialing possible, which also works fine. Enjoy, joash -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrey Loginov Sent: Thursday, January 05, 2006 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP/IAX softphone...

Hi Everyone, Does anyone know of any ATAs that can do proxy failover without using SRV. I don't want to rely on dns if at all possible. Basically, I have Asterisk boxes in two different data centers and I need ATAs to be able to uses the server at DC2 if DC1 goes down. The servers are already in a HA setup at each datacenter. I am looking for added protection if one of the datacenters

...l I have very good experience with the vegasteam ATA's devices.(you might also want to look @ sipura ATA's, since vegastream is doing an oem on there boxes) They support modem until v.90 speeds and faxes for g3. They are expensive, and again, work great and configure very easy joash ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of phil.dawson@marnock.com Sent: Friday, January 27, 2006 12:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ATA's ??? Hi...

Hey guys. If my Asterisk box connects to the PSTN using SIP and IP over ethernet and doesn't require any authentication, what sort of a trunk would need to be created? Thanks, -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality

Can someone give me an idea of the processor power I will need for 1 x TDM240 with 2xquad FXO's and 8 sip phones/ATA's on a quite 100Mbit LAN. The machine we have available of hand is a P4 1GHz with 768MB RAM. Tx Marnus van Niekerk -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093

...is detected (or at least no CDP capable device) the switch will automatically make it an access port, allowing only access to the native vlan, so, the switch port can be used very dynamically. Of course you need to define the vlan first, before you can create configs like this. Hope this helps, joash interface FastEthernet3/1 switchport access vlan 200 switchport trunk encapsulation dot1q switchport trunk native vlan 100 switchport mode trunk switchport voice vlan 101 qos trust dscp qos trust extend spanning-tree portfast trunk -----Original Message----- From: asterisk-users-bounce...

...g that the number is unavailable. But, what I need is to have the original PSTN status transferred to the SIP phone( xten eyebeam in this case) so I can see whether the end point was just busy, or that the number dialed was just plain wrong. Any help would be very very much appreciated. Joash Maanlander 14a/b m: +31 6 53 80 28 20 3824 MP Amersfoort e: joash.herbrink@kahuna.nl t: +31 33 4500370 ext 1006 URL: www.kahuna.nl <file:///\\www.kahuna.nl\> f: +31 33 4500371 -------------- next part -------------- An HTML attachment was scrubbe...

...the system. But test servers run normal * 1.2.1 with this codec, and, no problems even when beating up the system with a lot of calls. Use centos 4.1 very basic installation for operating system. * at home might be different though, but "normal" * works fine for me, also @ home. joash ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of sdcharly@gmail.com Sent: Sunday, January 22, 2006 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Insta...

...360.12962@sasami.anime.net> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Thu, 2 Mar 2006, John Jensen wrote: > You might want to get hold of the SPA3102 if you can. ... where? -Dan ------------------------------ Message: 9 Date: Thu, 2 Mar 2006 19:44:52 +0100 From: "Joash Herbrink" <Joash.Herbrink@Kahuna.nl> Subject: RE: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x) To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <819464631e0f321166ba6007cfc155de4...

Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost

Hey, all. I quoted a customer about $100 for some cheap SIP phones. I was planning on using the BT-102's, but he called said they look like "Princess phones," and I have to admit that he has a point. Some of the other inexpensive phones look decent, but (for example) the SPA-841's wiki entry says the remote end gets a lot of static. Since it'll be being used from a noisy

Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: