similar to: Cisco QoS Howto

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

First time around, I just unloaded/reloaded the modules. The box locked up tight. On reboot, I get this: general protection fault: 0000 CPU: 0 EIP: 0010:[<c01defb3>] Not tainted EFLAGS: 00010097 eax: f61d4260 ebx: f61d4260 ecx: ffffffff edx: f61d425f esi: f61d4264 edi: f61d4260 ebp: f4de7f14 esp: f4de7ef4 ds: 0018 es: 0018 ss: 0018 Process sh (pid: 387,

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

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

I'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the

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.

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

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

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

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

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

I am sure this has been asked before, but why not use Google newsgroup or at least some forum BBS software instead of this cumbersome mailing list process? -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

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

*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.

I would appreciate some help with this. I read the GPL license and basically it says you can do whatever you want with the software (sell, modify) as long as you include the source code, the License and make any changes you make available in the same manner to all others. My questions is this: If I develop an external application (say a Call Center application or a GUI management application)

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

All, I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway. Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know which cards, if any, exist for a 7206VXR to act in a similar capacity, either as a T1/PRI, DS3, or POTS FXO/FXS? What other Cisco routers can act as SIP gateways today? Thanks, Dave

Ok, this is probably an incomplete thought, but over the last few weeks of reading this list, I think I'm ready to start designing my system, and would like to solicit input. Currently, I have 2 POPs, one each in 2 different states. Each has a Lucent TNT w/~400 trunks. Each TNT has the capability to provision a PRI that can be piped into an * system. So, my interface to the PSTN is

http://bugs.digium.com/bug_view_page.php?bug_id=0000847 I'm placing a $250 bounty on getting Zaptel drivers working under BSD. While the 'hood' is open on this, we'd like to have drivers completed for FreeBSD, NetBSD, and OpenBSD on the x86 platform (other platforms optional). This needs to be done in a timely fashion. I'd like it completed by 6/30/2004, but once the project