search for: lethol

...It stands just as a 12" lcd screen with the cisco branding/logo and letters just as the 79xx series. I wonder if this is a new cisco model thats ready to roll out. It looks great, but then again, I doubt they will support SIP on it (at least on release) Anyone else know anything on this? Lethol

Hi, Reporting from Astricon, Mark uploaded the 1.0 release while giving his speech a few mintues ago.. Bring out the champagne :) Lethol

I have tried sjphone - worked well, although I think my 3 year old IPAQ had a bit of a hard time keeping up with the pace as there was quite a delay in the speech. Probably says more about my ancient IPAQ than SJPhone. Sam Lex Lethol <lethol@gmail.com> wrote on 23/09/2004 15:31:39: > I tried the xten one and didn;t like at all.. > > Havent tried to SJPhone, but my guess is that it has better support. > > Lethol > > On Thu, 23 Sep 2004 08:13:10 +0200 (CEST), Peter Svensson > <psvasterisk@psv....

...ot been able to find them. He had a working FWD registered thru Asterisk behing a NAT and a couple of samples of their new scripts, talking weblog (from Mr. Pulver) and other cool new features. If anyone got these please send em over or post.. maybe they should be added to the the wiki. Thanks! Lethol

...on) make: *** [.tmp/sccp_channel.o] Error 1 The README only instructs to change the path to asterisk source which is /usr/src/asterisk in my case. I have a recent asterisk installed running OK as far as I know, but I have not been able to compile this. Any feedback would be greatly appreciated. Lethol

Hi, I'm having a problem with one of our 7960. They all run latest 7.4 SIP firmware. The problem appears on the other end. The other end constantly hears a 'crackling' noise. I have tested using phone set, headset and speaker and the noise appears on all cases. I have other 7960 setup exactly same way (using same asterisk, firmware, etc) so it looks like a hardware issue.

...monitor > Trying to reawake monitor > Monitor restarted > CID record - skipped 602.460051ms trailing ring > CID record - recorded 1711.737009ms between rings Segmentation fault Any advice on how to correct this or the other problem would be appreciated. ;) Lethol

...topped record thread on vpb/1-1 == vpb/1-1: Ending play mode on vpb/1-1 > vpb/1-1: Setting state down == vpb/1-1: Hangup complete > Restarting monitor > Trying to reawake monitor > Monitor restarted > Monitor got null event Any advice/pointers/suggestion are greatly appreciated :) Lethol

I have installed a 6VPCI card from voicetronix's but i can't get astersik to use it! Now looking at the loaded modules the chan_vpb is not loaded- so I assume that is why it is not working. Now I modified my vpb.conf file and extensions.conf, have I missed something Has anyone a installation guide as I am very new to this!! I have had asterisk working with SIP extensions. by dowloading

Hi all, I used to have an OpenLine4 card, but decided against using it due to some problems with hangup detect. Does anyone on the list actively use Voicetronix's OpenSwitch12? What are your opinions on the card? Cheers, Flynn

Is there a soft phone for PocketPC or iPaq? If not, is someone working on it? I will be more than willing to contribute my mite if needed. Thanks, -- sudhir

Apparently asterisk's default way to a 3-way conference lets the user in the middle hangup and the other parties stay on the conversation. This is great some times but it creates quite a bit of problems when trunks dont have disconnect supervision or when trying to do accounting and billing on the user. Here is the 'normal' situation when a user tries the setup a 3 way conference once

Yesterday I upgraded my Zaptel to 1.2.26 or I think that was it, the latest 1.2 version at downloads.digium.com. I have a Digium 4 card populated with 4 red FXO cards using channels 1,2 and 4. Channel 3 is not used. It's been working fine for a few years. After upgrading to 1.2.26 calls stopped coming in on channel 1, Channel 2 still worked fine and I could get dialtone and make calls

Hi we've got Asterisk CVS-HEAD 18-Aug-04 (modified by Voicetronix as available on their site for use with the vpb driver) and an OpenLine4 (4xFXO). The same server also has two X100P. Calls on the Voicetronix card drop instantly when the called party picks up. The vpb driver reports that it detected a hangup (loop drop) yet there is no hangup when connecting the X100Ps or analog phones to

Hi All I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems

Hi, Ive been playing on a asterisk to orion gsm box E1 pri setup. I have achieved incoming calls to be passed to my asterisk box successfully but outgoing calls will just I have tried playing with various pridialplan and overlapdial settings and with no success. If anyone can make more sense from the log, I'd certainly appreciate it. I am sending a 10 digit number to be dialed. I guessed