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I am out of the office until 28/05/2015. Estar? ausente de la oficina, para cualquier apoyo que necesites favor de contactar a Jose Manuel Flores Dublan Email: jfloresd at iusacell.com.mx Cel: 5530300533 Note: This is an automated response to your message "CentOS Digest, Vol 124, Issue 25" sent on 05/25/2015 7:00:02 AM. This is the only notification you will receive while this

I am out of the office until 26/12/2015. Estar? ausente de la oficina, para cualquier apoyo que necesites favor de contactar a Edson Cota Email: edson.cota at nextel.com.mx Note: This is an automated response to your message "CentOS Digest, Vol 131, Issue 21" sent on 12/21/2015 6:00:02 AM. This is the only notification you will receive while this person is away. ?Este mensaje y

From: Jorge D Cisneros <jorge.cisneros at hp.com> Problem: The actual code only check for entries with kernel or linux, so this example will not work, this patch is to add support to gfxboot to detect the entry COM32 and use any module, in this case the module is chain.c32 Version: Syslinux 4.04 Example: UI gfx...

When i make a call using oh323 channels, how i can send a ringing sounds to indicate to the users that the call is in progress thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030623/95bfcc74/attachment.htm

Hi Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C Site A Site B Site C ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186 Thanks -------------- next part -------------- An HTML attachment was scrubbed...

Hi All, My PXE configurations works fine for a bios PXE (the server in legacy mode) but hangs in an EUFI mode. Look like it can transfer the bootx64.efi but not the next one ldlinux.e64 Any ideas? Thanks Software> syslinux ver 6.3 atftp 7.1 Log server side >> Booting Embedded LOM 1 Port 1 : HP Ethernet 1Gb 4-port 331i Adapter - NIC (PXE IPv4) >> Booting PXE over

On Thu, Mar 5, 2015 at 2:03 PM, Jorge Cisneros via Syslinux <syslinux at zytor.com> wrote: > Hi All, > > My PXE configurations works fine for a bios PXE (the server in legacy > mode) but hangs in an EUFI mode. Look like it can transfer the bootx64.efi > but not the next one ldlinux.e64 > > > Any ideas? My...

No, we can use that, because I want to use the com32 module, in this case I want to use the chain.c32 to boot to Winpe. Cisneros, Jorge (George) wrote: > Problem: The actual code only check for entries with kernel or linux, so this example will not work, this patch is to > add support to gfxboot to detect the entry COM32 and use any module, in this case the module is chain.c32 > Version: Syslinux 4.04 > > Ex...

Hi I have one question, somebody can tell me if the card TE110P work in mexico, and maybe can tell me the config. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050923/9dedf2bc/attachment.htm

Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263

toy buy my first cisco 186 but when i read this page http://www.djernes.org/~shawn/ata186.htm i cant find in my dev page some parameters just like " UseSIP " what i need to do to show this parameters Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030626/42e66005/attachment.htm

Hello, Again one more question about hardware. What could you suggest me to buy. I need hardware to connect let's say 4 analog lines. (FXO). This hardware should "talk" to Asterisk of course.. Thanks very much for some advices :) Bartek -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have a couple of droppables in my page. Each of them looks like a list and is a <div> containing other <div> for the lines, each line formed by <span>s for different fields. Each line is declared draggable (right after each line, since ithey''re created inside a loop). I have both of them inside a <div> since I need to refresh both at the same time and

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Hi I have a gw linux in this machine i have one quicknet card, how i can reserver a prt of my bandwidth to voice data, for example when i download a big file the voice don't loss quality thanks

Hi This bug is really crazy, please help me In the follow scenary ATA-186 -> SIP -> Asterisk -> SIP -> ATA 186 : No DTMF gets through * in outbound mode, Sip conf [204] type=friend username=204 secret=somesecretpassword host=dynamic canreinvite=no ; The follow line don't work dtmfmode=rfc2833 nat=1

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Hi I buy this card just few month ago, but i have a problem when i try to use in the asterisk pbx the software detect when i plug or unplug the phone cord, but when i plug the phone cord just 2 seconds later asterisk detect a ring event but nobody is calling what happend??? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a wired problem, i can recive call but i can't make any call. ATT say that is my problema because the call is operator mode (???) The log is MFC/R2 Chan 1: Call control(1) MFC/R2 Chan 1: Make call MFC/R2 Chan 1: Making a new call with CRN 32772 MFC/R2 Chan 1: 0001 -> [1/ 1/Idle /Idle ] Chan 1: -- Dialing on channel 0 Chan 1: -- Dialing on