similar to: AW: Asterisk-Redhat 9 install guide.

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 ----- >

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

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away

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

hi all, I am using centos3.5 for my firewall home. I would like to do some tests with bridge firewalling. When i try to create the bridge, brctl givtes me this error: br_add_bridge: Package not installed. i find some answers about this error, but the most important is the one that says driver is not loaded into kernel. In my /boot/config-2.4.21-32.0.1.EL file appears compiled as a

How is a callgroup used in the dial plan?? I can't see an example in the extensions.conf.. > See configs/sip.conf.samples search for pickupgroup > > RTFSC (Sample Configs) > > > Jeremy McNamara > > > > > WipeOut . wrote: > > >Is there a way to pickup a call whn using a SIP phone? > > > > > > > >> From

Anyone having problems getting transfer to work here is the answer... It appears the manual is incorrect.. The manual says: 1)Press "Transfer" button. 2)Dial the target extension. 3)Hangup the phone. This will disconnect the call.. Here is how it can be done.. Matteo gave this solution.. (thanks) NOTE:'Use # as Dial Key:' must be set to YES To trasnfer: 1)Press

Unfortunately this does not work. Example: Yes, when i give it a few Days, the client will retrieve the actual crl faster. But the auth still works. I have tried it. I revoked an cert. Installed a new win10 client and joined the domain. After login with the revoked p12 cert on a yubikey, i can see he queries the CDP and still allows the login. With certutil and a cert in DER format, i tried

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

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

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

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, Is there and trick to getting a conference room up and running.. I have 'conf => 7500' in the meetme.conf file and 'exten => 7500,1,MeetMe(7500)' in the extensions.conf file (in the same context as my phone extensions).. When I dial extension 7500 I get the voice saying "That is not a valid conference number, Please try again.." <beep> <beep>

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

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

Hi, Has anyone worked out a way to use Asterisk in a Hot Desk environment?? I have not been able to think of a way for the user to have control over which IP phone will ring when that users extension is dialed without the user needing to reconfigure the phone.. Something like this would be cool.. User dials *8555 (or similar) and is prompted to enter their extension and then password, after

Hi, I have just got my SNOM 200 to start doing some real testing with *.. I am trying to use the GSM codec but the quality is really bad, Is that normal? does anyone actually use GSM?? Also are there any 'gotcha's' that I need to look out for so I don't spend hours trying to get somthing working that really doesn't work anyway.. Thanks.. later.. --

----- 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 >