similar to: Re: RedHat Distribution

Hallo.. if you have one more for me...thanky you.... Stephan -----Urspr?ngliche Nachricht----- Von: WipeOut . [mailto:wipeout@linuxmail.org] Gesendet: Donnerstag, 10. April 2003 19:28 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Asterisk-Redhat 9 install guide. Hi, Not sure if anyone will be intersted but I have put together an install guide for Asterisk on RedHat 9.. Its

Hi, I'm new in * and I would like to know what version of the Linux kernel or RedHat Distribution do you recomend. Best regards, Francisco Mesquita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030902/74942dc2/attachment.htm

Hi, Not sure if anyone will be intersted but I have put together an install guide for Asterisk on RedHat 9.. Its nothing special but it may be of use to the newbies.. Like me.. If you would like a copy let me know and I will send it to you.. later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

I thought I would give RedHat 9 a try with Asterisk..I thought it would be a good idea to use the latest version.. Zaptel, Zapata and Libpri all appear to have compiled sucessfully.. But.. (Why is there always a but??) It seems Asterisk is having issues with 'termcap' or 'tgetent' whatever that is.. Here is the output from 'make install'.. --------Start-------- if [ -d

This is troubling. Shouldn't your hubs/routers autosense the 10MBPS? ---------- Original Message ---------------------------------- From: WipeOut <wipe_out@lycos.co.uk> Reply-To: asterisk-users@lists.digium.com Date: Wed, 15 Oct 2003 07:53:13 +0100 >Steven J. Sobol wrote: > >>On Wed, 15 Oct 2003, Jon Pounder wrote: >> >> >>Nothing works. Call transfer

My thoughts on this after reading Steven's very politically worded reply is that IMO your best bet would be to go with RedHat, I am not going to go into details about the if's, when's, why's, and but's.. I am running Asterisk quite well on RedHat 9 and if you like I have created an install guide for setting up an Asterisk box on RedHat 9 which I can send to you if you are

Ok.. I have done some more digging and the problem seems to be caused by chan_capi not detecting that the call has been answered.. I downgraded chan_capi from 0.2.3b to 0.2.2 and the system is working fine.. Kapjod, do you know of this problem? Later.. > I seem to be having problems with my 2 asterisk systems.. > > This works.. Call originating from the PSTN and then routed to the

Look like SNOM have rolled back the firmware version of the 200's from 1.16w to 1.16q.. Anyone know why? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hi.. How do you accept user input while waiting or playing moh? My Dialplan is as follows.. ring,ring,.. Hello thanks for calling blah blah... Please enter the extention number blah blah... WaitMusicOnHold(10) If no input pass call to operator.. The problem is that the user has to input the extension while they are being told what to do.. any input during Wait or WaitMusicOnHold is ignored...

I see in the changelog that ENUM support has been added.. anyone know what this is? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Callgroups/pickupgroups are allocated per channel, not in the dialplan. sip.conf and zapata.conf are the two files you're interested in. -wade ---- Original Message ---- From: wipeout@linuxmail.org To: asterisk-users@lists.digium.com, Subject: RE: [Asterisk-Users] Using callgroups (was: Taking a call for someone elses extensionfrom my extension) Date: Sun, 20 Apr 2003 16:39:15 +0000

Subject: Re: [Asterisk-Users] IAX Bandwidth Question Hi, I have changed the codec to lpc10 (I see what they mean by Mr. Roboto!!) and the ping times generally dont go over 300ms. It seems very odd that GSM saturates the link. I would love to give g.723.1 a try but have no idea where to get the proprietary codec... can anyone help? Thanks Jay > ----- Original Message ----- >

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Hi, I installed s fresh system yesterday and it seems that zttool did not install!! ztcfg is there.. Anyone else had this problem or is it just me? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Can the system command be used to retrieve a variable from a mysql database using the mysql command line client?? or would it be simpler to write some sort of AGI type application?? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Thanks, annoying but only course of action I guess ... (c; > -----Original Message----- > From: WipeOut [mailto:wipe_out@lycos.co.uk] > Sent: 29 September 2003 10:36 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] RE: Asterisk list a SPAMer > (uol.com.br), I think not ... > > > Just add a filter to your mail client to delete all mail from >

Hi, I have just been playing with the latest X-Lite.. It works fine with Asterisk.. As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only one that didn't work.. not sure why.. But the bigger problem is that when I call another extension that is using a Snom200 the call connects but there is no audio in either direction.. I have tried G.711a/u and GSM and while X-Lite shows that

Hi, Is Asterisk (or can it be set up as) a SIP proxy? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi, What is the correct chan_capi dial syntax?? This is what I think it is.. exten => _90.,1,Dial(CAPI/[msn here]:${EXTEN:1}) This seems to work for local numbers.. but I have an access number for cheap long distance calls.. wich gets dialed and then the number I want to call is sent as DTMF after a few waits (w).. On my X100P I used the following.. exten => _9001.,1,Dial(Zap/1/[access