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...se 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 organisaties met het zakelijk gebruik van Internet. Kahuna Network Solutions levert beheerde oplossingen die de beveiliging, performance en beschikbaarheid van netwerk- en Internetinfrastructuur verbeteren. Kahuna Business Solutions levert oplossingen voor het verbeteren van on-line...

...3:E8:8F:63:75:8F:85:59:B5:17:BE:F8:47 truststore.jks on puppetdb has ''puppetdb ca'' with print 62:8F:76:CE:5C:9D:23:B0:1D:9D:7A:2F:39:5A:74:43:1D:BB:D9:1E $ openssl verify -CAfile /etc/puppet/ssl/ca/ca_crt.pem `puppet master --configprint hostcert` /etc/puppet/ssl/certs/puppetdb.kahuna.local.pem: OK (yes, I have the SSL certs in /etc/puppet) If someone could help, that would be great. I''m running in circles here. *Thanks!* kl [1] http://docs.puppetlabs.com/puppetdb/1.3/connect_puppet_master.html -- You received this message because you are subscribed to the Google...

...hone( 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 scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051128/55a5bb1d/attachment.htm

...Word I have is that Joe''s > done this on-purpose and that he is throwing in the towel. I just discovered the following, posted to news.admin.net-abuse.blocklisting: | Subject: Update on relays.osirusoft.com | Newsgroups: news.admin.net-abuse.blocklisting | From: Andy Finkenstadt <kahuna@panix.com> | Message-ID: <bigprp$ri3$1@reader2.panix.com> | Date: Tue, 26 Aug 2003 23:50:15 GMT [snip] | "Direct from Joe Jared''s mouth, he''s shutting down the service, that people | need to stop using it, that he''s been saying for weeks that people | need t...

Hi there, I have bought 70 sipura 841 phones for a customer of mine. When following the mass provisioning guide in the admin manual for the sipura, I see it download the spa841.cfg file from my tftp server Sometimes the phone also downloads is phone specific file via tftp, and it works okay then. But, after a reboot of the phone, it is very very likely that it won't startup

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

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

...ype: 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: <819464631e0f321166ba6007cfc155de44073908@kahuna.nl> Content-Type: text/...

...t; clients (very unlikely scenario) you'll be transcoding to/from GSM. G.729, > or whatever you're using. > >> >> Any other suggestions that you might have would really be appreciated. >> >> >> >> >> >> Joash Herbrink <Joash.Herbrink@Kahuna.nl> wrote: >> >> >> >> I have tested an asterisk server with over 5000 concurrent calls. >> >> The system setup was a P4 HT 3Ghz, 4 Gb RAM, and 1 gbps Ethernet >> connection on a cisco 3560 switch. >> >> >> >> This works, but puts...

I have installed several call centers in the netherlands with the eyebeam softphone (from the counterpath guys) It is not free, but very stable, and pretty easy to use. 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

Phil 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

Cisco phones act a as a switch. If you do not use the CDP protocol to "tell" the phone it needs to be in a special VLAN (802.1q) then it will just use the access port settings on the switch, and, also allow the PC connected to the 2nd Ethernet port to have access to the network. However, if you have an all cisco powered network, with all cisco phones, I could advise you to use the CDP

I downloaded and installed the none commercial g729 codec very often now I only disable HT on my systems I think * doesn't like this One of the guys @ digium advised me to turn it of, since they haven't written * to be multi treading any way The codec I download is the http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium4.so It should work fine. Wouldn't know what it