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...ach but pci-list-assignable-devices shows nothing and pciback module show that the device is seized from dom0. I''m really lost... thanks in advance. -- _____________________________________________ Francisco Javier Funes Nieto [esencia@gmail.com] CANONIGOS Servicios Informáticos para PYMES. Cl. Cruz 2, 1º Oficina 7 Tlf: 958.536759 / 661134556 Fax: 958.521354 GRANADA - 18002 _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hi, As it has been brought to my attention (by several parties), my current MB does not support VT-d. Since I don''t feel like purchasing a brand new system, I have searched for replacement MB-s that would fit into my existing uATX case, support my C2D CPU, and support at least 4x2GB ram. I have narrowed it down to the following four models: SUPERMICRO MBD-C2SBM-Q-O (Q35 chipset)

...you describe. I can call from a SIP softphone, through asterisk , through the cisco and out to our meridian system or the PSTN. In fact, it works very well. Are you sure that you have the dial-peers on the router configured correctly? -----Original Message----- From: Cerrajetto [mailto:cerrajetto@pyme.net] Sent: 31 July 2003 09:09 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RTP codec 13 received - Cisco incompatibility? Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls d...

Hello, I've seen that Nuance SIP audio provider supports additional information (parameters and extra headers) in SIP URIs, using the format: sip:user:password@host:port;uri-param1;uri-param2?header1&header2 For example, sip:1234@myserver.com;extra_header=Uui?Uui=Hello Does Asterisk support this format? Is there a way to retrieve the value of these additional headers, and then decide

Hello, We are running Asterisk-1.0.12 in a CentOS 4-4.2 system, kernel 2.6.9-42.0.3.ELsmp. We have some custom AGI, and when we launch Asterisk the system works fine. But **after some time**, each AGI execution generates a zombie <defunct> process. We believe that it's not a problem in the AGI code, because Asterisk+AGI is working fine in the first "n" minutes/hours. This

Hi All, We''ve limitation of 500 GB per server as per ServerLoft. we are needing to store word/pdf files. We are planning to use following setup. /dev/sda1 - boot - ext2 /dev/sda2 - swap /dev/sda3 - dom0 - ext3 /dev/sda4 - LVM - one LV for each domU, We''re planning 3 DomUs. /dev/sda5 - FreeNAS. Is that possible? Can it be in one of the LVs under 4th partition? Currently, one

Xen Community: As you are aware, the Xen Cloud Platform has been available for download since last November and is becoming a very successful release for Xen.org. In fact, we are seeing substantial network traffic for this solution as we had 26 TB of data from XCP downloads in the past 4 weeks. Of course, this amount of traffic is forcing a re-examination of our co-location agreements as we are

Xen Community: As you are aware, the Xen Cloud Platform has been available for download since last November and is becoming a very successful release for Xen.org. In fact, we are seeing substantial network traffic for this solution as we had 26 TB of data from XCP downloads in the past 4 weeks. Of course, this amount of traffic is forcing a re-examination of our co-location agreements as we are

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

La referencia de las cuchillas, no son todas la misma Date: Thu, 10 Dec 2015 22:04:24 +0100 Subject: Re: [R-es] Tiempo de vida From: cof en qualityexcellence.es To: j.para.fernandez en hotmail.com CC: r-help-es en r-project.org Gracias. ¿Y qué es "Referencia"?... Gracias, Carlos Ortega www.qualityexcellence.es El 10 de diciembre de 2015, 21:48, Jesús Para Fernández

...; Xen-users mailing list >> Xen-users@lists.xensource.com >> http://lists.xensource.com/xen-users >> > > > > -- > _____________________________________________ > > Francisco Javier Funes Nieto [esencia@gmail.com] > CANONIGOS > Servicios Informáticos para PYMES. > Cl. Cruz 2, 1º Oficina 7 > Tlf: 958.536759 / 661134556 > Fax: 958.521354 > GRANADA - 18002 > > > ---------- Forwarded message ---------- > From: Benjamin Knoth <knoth@mpdl.mpg.de> > To: xen-users@lists.xensource.com > Date: Tue, 10 Aug 2010 11:29:09 +0200 &gt...

Hello! Is there a way to communicate from Asterisk to a TTS server? I've seen festival.conf, but it seems that it works only with Festival server. Thank you.

Hello, Do you know a **real** sip fax server to _send_ faxes with Asterisk?. By example: an employee sends a TIFF facsimile by email to the SIP fax server; SIP fax server uses SIP to communicate with Asterisk; Asterisk communicates with a Cisco SIP2PSTN Gateway via SIP; Cisco sends the fax to the PSTN. I've read some posts than mention Hylafax, but it needs for real modems/analogic lines:

Hello: I would like to use Asterisk as a redirect/proxy sip server to route SIP calls on a sip header/parameter basis. I've tried some things successfully: - SIP registration from clients. - On-the-fly compression for wan VoIP transfers: SIP G.711 --> GSM IAX --> (wan) --> GSM IAX --> SIP G.711 - Sending custom parameters in URI: exten => 1,1,Setvar,VXML_URL=var1=value1

Hello, I'm trying to registrate a Nuance Server in Asterisk (using SIP) with no success. It seems that Nuance does not send any secret/password (there is no way to define it!), this is the list of parameters that Nuance provides for registration: audio.sip.UserAgentURI=sip:user@domain audio.sip.UserAgentPort=<port> audio.sip.ProxyServerURI=sip:<IP>:<port>

Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls directly via Cisco to the PSTN, the call works successfully. If the client softphone calls via Asterisk to other SIP internal extension, it work fine too. The problem is when a client calls an Asterisk extension, and Asterisk transfers