similar to: SNMP Monitoring

Thanks for this informations. Do you know where I can find the icd-snmp package for a redhat 9 distri? I can't find it. Thanks. Message: 6 Date: Fri, 09 Jul 2004 15:45:57 +0200 From: Andrea Fino <af@faino.org> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] SNMP Monitoring Reply-To: asterisk-users@lists.digium.com Holger Schurig wrote: >>You could try

Hi to all, today I have released the version 0.3 of ast-ax-snmpd. It is some code that adds the snmp subagent functionality to asterisk, using the ucd snmp framework and following the agentX standard. It has tested under debian woody, with specific version of ucd_snmp, so your mileage may vary. I am evaluating to transfer this effort under a more recent net-snmp umbrella, this could permit

Hello, Can someone explain me how to do caller based routing. Here is my example. I have an asterisk between a PBX and the PSTN. The second company get the same, and so, I can interconnect them by VoIP. Classic architecture. My problem is when I want to place fax. The calls between the 2 sites are in gsm codec. So the fax doesn?t work! Is there any possibilities to do caller based routing in

I have just started developing asterisk, and am trying to start simple. I have a X100P device and an S100U device. I am trying to use the examples provided, where I add a few lines to the /etc/zaptel.conf, /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may connect an analog line to the X100P and an analog phone to the S100U. When I dial the analog line, it should ring

Hi, sipp (http://sipp.sourceforge.net/) seems to be a good app. Take a look at http://www.voip-info.org/wiki-SIPP on the wiki to have more info about it... Basically, there is scenario which are describe there and I personnally generated about 3,000,000 calls before having to restart asterisk and i placed about 90 concurrent calls. Good luck! -----asterisk-users-admin@lists.digium.com a ?crit :

Hi, I was wondering if there are any Asterisk MIBS (specifically regarding call information) ? I noticed a post citing www.faino.org, but this site doesn't seem to exist anymore, and The Book v2 doesn't have any references to MIBS. Any pointers greatly appreciated. Keith Burns "The dogs may bark but the caravan rolls on.... " -------------- next part -------------- An

Hello, Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. Thanks by advance.

Hi, Is there a module within the Asterisk standard distribution that provides SNMP features? Is there any third party software for that purpose? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/918b5ebf/attachment.htm

Can anybody, please, explain me how many parameter are estimated using randomLCA? For examples, model "dentistry.lca2random" estimate 1 scale (or variance, b_j) parameter and 2 position parameters (a_cj)? Doesn't it? Do I need at least 4 diagnostic tests for such a model? What happens if I specify options blocksize and byclass? How many diagnostic tests (or rater) I need?

Hello, I had installed several asterisk, but I every time had a problem with callerID. On each phones I don't reveive the first digit. For example: Caller 0672083516 called an IP Phone 0123456789. The IP Phone see 672083516 as callerID. I think there is a patch for it, but I don't know where to find it out. Thanks. -------------- next part -------------- An HTML attachment was

Full_Name: Doug Hershberger Version: 1.2.2 OS: Red Hat-Linux 7.0 Submission from: (NULL) (216.99.65.36) In the R FAQ I find the following entry explaining that R no longer has problems with memory. http://cran.r-project.org/doc/FAQ/R-FAQ.html#Why%20does%20R%20run%20out%20of%20of%20memory%3f However in my installation: R Version 1.2.2 (2001-02-26) Installed from the red hat RPM on your

Dear participants to the list, this is my problem: I want to obtain an expression that represents the second derivative of one function. With "deriv3" (package "stats") it is possible to evaluate the second derivative, but I do not know how I can get the (analytical) expression of this derivative. For example: Suppose that I have a function of this form:

I use Samba as a PDC (non-AD-style) on the server n°1. I have successfully created a BDC on the server n°2. I followed the official guide SAMBA (https://www.samba.org/samba/docs/man/Samba-HOWTO-Collection/samba-bdc.html) On servers n°2 I have also transferred the users database from the server n° 1 (https://www.cyberciti.biz/faq/howto-move-migrate-user-accounts-old-to-new-server/) When I

CRAN (and crantastic) updates this week New packages ------------ * disclapmix (0.1) Maintainer: Mikkel Meyer Andersen Author(s): Mikkel Meyer Andersen and Poul Svante Eriksen License: GPL-2 http://crantastic.org/packages/disclapmix disclapmix makes inference in a mixture of Discrete Laplace distributions using the EM algorithm. * EstSimPDMP (1.1) Maintainer: Unknown Author(s):

I have upgrade the pdc server from debian sarge to etch. After this operation we can't login to the domain (client windows 2000 & XP) smbclient works fine and from a windows local account I can browser the samba directories too (puttin username and password before..) I use tdbsam of course and pdbedit -L let me see all the accounts, it seems to work fine I tried to remove a pc from

Hello, Is it possible to have 2 E100P cards on one asterisk? I'm able to do that now, but I'm not sure about my config. Here is my zaptel.conf [root@ConnecteurPRI02 asterisk]# more ../zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=fr defaultzone=fr span=1,2,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 loadzone=fr defaultzone=fr Here is my zapata.conf

Hi, Is there a way to use swissvoice IP10 in MGCP mode behind NAT. I already use this fonctionnality with a PBX other then asterisk, and it work very well. In fact, I need to use this settings in the IP10 telnet session set xgcp rgw_name MAC_ADDR set tcid 0 notify_entity ca@[@IP PBX]:2427 In fact, the phone register with the MAC@ and not with the IP@, and the PBX reply the phone in the UDP or

Hi, I had a question regarding asterisk installation with 2 E100P card. I would like to do such installation but I had some troubles: PABX -> E100P -> asterisk -> E100P -> PSTN In fact I would like to add some IP Phones to my PABX wich is full. But with this installation, I can't place call. I can receive call without problems. I also can place call from IP Phone, but I