search for: psknet

Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too expensive. Can anyone recommend a decent channelbank that won't break the bank? TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

...ected via 2xT1 (per-packet load balancing) between a Cisco 7206 and a 3640. When the total bandwidth pushes much past 50%, I start getting some crazy distrotion (jitter?), making it impossible for one or both parties to understand the other. TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638

...cf>] [<f89c7852>] [<c01d0349>] [<c01d0568>] [<c01d2fa8>] Code: 8b 01 85 45 f0 75 1c 8b 02 89 d3 89 c2 0f 18 00 39 f3 75 e9 <0>Kernel panic: Aiee, killing interrupt handler! BTW, this is kernel 2.4.25-gentoo-r2 -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638

Hi, I am trying to receive a fax with the spandsp library. The sending fax says "success" but there is no tiff file generated. I use "exten => 7000,1,rxfax(/tmp/testfax.tif)" in my extensions.conf. The connection is via SIP/G.711 as I have read on the list that this can sometimes work (I know Fax over IP is troublesome without T.38). I think the transmission should not

Quick question: My PHP script is now able to connect to the manager port and successfully authenticate using MD5. I would strongly prefer not to do plaintext authentication at all. Would anyone object to plaintext authentication being left out? -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve

...extension via an FXO port. Of course, I'd also end up with a few VOIP extensions as well. Does this sound like a feasable plan? Aside from eliminating the partner system (which isn't really an option), is there anything I'm missing? -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

...yet. If I understood Chris correctly, he's going to work on finding a developer to write the drivers and get them committed to FreeBSD. I'm not up on Net/Open, so don't ask me about those, but I do want them included in this bounty. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

Hi all ! Have anyone a resource / link for documentation to configure Asterisk to act as a SIP 2 PSTN gateway (ISDN PRI) ? Thx. Regads, Andreas. -- "If you want to pray. Go to the sea." ---------------------------------------------------------------- Andreas Czerniak <cognac@amcs.net> PGPkey http://pgp5.ai.mit.edu:11371/pks/lookup?op=get&search=0xEDB224EC

I have asterisk successfully dialing out using a X100P over a normal analogue PSTN line. But when I try to dial out over an analogue line that goes via a hardware PBX the call asterisk does not dial. Is there a configuration change I should make ? I am thinking of something like not wating for dial tone.

...0 seconds, and if still no answer, ring all 4 groups until the call is answered. What do I need to do to get this behavior? If the answer involves $$, tell me about it, I'm not afraid to spend some cash to help streamline my business. Thanks, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638

...r is definitely using some disk space: $ du -sx ~user3 796 /var/mailboxes/user3 Now, user2 did the same thing until I ran quotacheck, then it worked fine. Any ideas? Please respond directly, as I'm not subscribed to this list. Thanks, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 - 866.477.5638

The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing

The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox

...ng the CNAM. BUT if you are seeing the CNAM on the TNT that may not be the issues, if this problem is present you usually will not see the name on the TNT either, just the number. > > Thanks, > > -- > Troy Settle > Pulaski Networks > 866.477.5638 > http://www.psknet.com > > > > Damon Estep wrote: > > If you are looking for real high density VOIP termination I would look > > at > > > >>something like a Lucent APX 8000, configure correctly it can pass > > > > 2500+ > > > >>g.729 calls to the P...

If you are looking for real high density VOIP termination I would look at > something like a Lucent APX 8000, configure correctly it can pass 2500+ > g.729 calls to the PSTN course we paid lots of $ for ours. > > Chris > Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is

When setting include => daytime|9:00-21:00|mo-fri|*|* How does this determine what is different between 9 AM and 9 PM And after hours ??? I want different hours on Saturday and Sunday And a different welcome message after hours Any help appreciated Regards Mick

Are they any issues still with hyperthreading processors, I've read and been told by a few people to make sure its disabled in bios if I want to use * on a hyperthreading machine. Kind Regards, Chris Bond

Friends, Romans, Countrymen, lend me your ears! Or, in this case, your eyes. It has become a problem I've noticed over and over again, that we tend to think the mailing list is some sort of forum, used to post whatever your hearts desire. This list, not unlike any other mailing list, has a purpose. A focus if you will. Hopefully this 'document' will point out some key things we

*This message was transferred with a trial version of CommuniGate(tm) Pro* Hey all, I've just done a quick (but functional) web front end for searching the CDRs in a MySQL database. Anyone interested in trying it out? I'm wondering what to add to it next. So far you can seach using source, destination, CLI, channel and date ranges. It also displays ALL fields in the database table.

The soundfiles I submitted earlier today have been cleaned up, and added to the Digium CVS server in a more formal manner. Also, some of the really bad formatting in my .txt description file has been rectified. All of the sounds on my website are now on the Digium site, and I will be submitting future changes via patches to Digium for additional sounds. Ideas welcome for more text; I may