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...s 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 asterisk, in order that when a fax try to send a fax, the call is automatically routed through the PSTN and not through the VoIP. Thanks. GIBERT Fr?d?ric Mobile: +33 6 72 08 35 16 Fax : +33 1 30 71 39 33 Mail : frederic.gibert@viginetworks.fr -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040721/f689808d/attachment.htm

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.

Hello, Does someone know how to setup snmp monitoring on asterisk. I?ve plan to deploy 50 asterisk, so I need some monitoring tools. I try with nagios as I read in the wiki, there is some project on it, but I can?t reach the end. Can someone help me? Thanks. GIBERT Fr?d?ric Ste VigiNetworks Mobile: +33 6 72 08 35 16 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040708/85e32a53/attachment.htm

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

...ly, 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 : ----- Pour: <asterisk-users@lists.digium.com> De: "GIBERT Fr?d?ric" <frederic.gibert@viginetworks.fr> Envoy? par: asterisk-users-admin@lists.digium.com Date: 23-06-2004 10:46 Objet: [Asterisk-Users] Call generator Hello, Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plate...

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

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

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):

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