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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 organisaties m...

...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

...ulaw or alaw on your SIP > 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. >> >> >> &gt...

...962@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: <819464631e0f321166ba6007cfc155de44073908@k...

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

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