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Goodday, I'm an opensource fanatic and I have already installed asterisk in my mandriva linux. Actually, I'm also planning to install the asterisk management portal for GUI of asterisk. If anyone could help me guide in installing this. Thanks a mill for the help.. -Rommel-

(udp/5060, udp/2727, among others). One way to tell for sure would be to run 'lsof -i' which would show you the process associated with the port. As far as the call not reaching asterisk or being a firewall issue, one way to tell might be to start a tcpdump just prior to making the incoming call. Something like this: tcpdump -n 'port 5060' That would show the connection

supported, beside Idle, On call and Ringing ? Can we expect this list to match DEVICE_STATE's one (UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD) > Might be worth seeing if other phones do the same. > > S > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by

honored by DSCP (first 5 bits)- even old equip should be DSCP "compatible"...or I need to do more reading :) -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, October 01, 2009 3:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] QOS/DSCP for IAX? Michelle

, as it seems to be running Asterisk-11. =A0I&#39;ve previously installed A= sterisk-11+FreePBX in a VM, and this appears to be very similar. =A0Is ther= e any upside to using AsteriskNOW vs. Asterisk+FreePBX? Other than the obvi= ous fact that everything is nicely placed on an iso for ease of installatio= n?<br> <br> As for the actual upgrade, is it possible to step through each

modern technology, shorter durations could work. Most phones of all types don't make a standardized tone burst but produce tones only while the button is pressed. Fast "punching" will produce short tones. On the other hand, a redialed number will be very well formatted. Reliability of TT data transfer for audio applications (over the phone voice mail, credit card, IVR, etc) would

mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? > > However, the MWI does not indicate voice mails for 610 and I keep seeing > this error message: > > ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox > 610 in context a10 > > However, mailbox 610 is clearly defined in voicemail.conf: >

?having problems with installing asterisk@home i downloaded the asteriskathome-1.5.iso file from asteriskathome.sourceforge.net link & burned it on a cd but it is not booting what seems to be the problem hoping for a quick reply -------------- next part -------------- An HTML attachment was scrubbed... URL:

Robb, I wrote up a small tutorial on setting up the standard tftp server for linux check it out on my site. http://asterisk.titaniumsoft.net/ Mitchel -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Robert Boardman Sent: Thursday, May 13, 2004 2:44 PM To: asterisk-users@lists.digium.com Subject:

MD -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michiel van Baak Sent: September 12, 2008 2:08 PM To: Asterisk Users List Subject: Re: [asterisk-users] Setup speed dials on Cisco 7921 On 13:15, Fri 12 Sep 08, OCG Technical Support wrote: > I've added lines like this: > > > >

Or even: <a class="moz-txt-link-freetext" href="http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946">http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&amp;mid=4946</a> (same thing from the UK site:) <a class="moz-txt-link-freetext"

Hey guys, Im about to sign up for VoicePulse Connect. Of course, I plan on using my asterisk server to "register =>" with the service. I would rather sign up with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is behind another Linux box serving as my nat/firewall. Does anybody think I will have a problem ? Should I stick to IAX and VoicePulse Connect or can I use

Appreciate the kindly help and advise. Regards Bilal --------------------- > > Bilal, > > I suggest you turn on logging on your tftp server to see > what files are actually being requested, and if the the tftp > server is dishing them out... Try adding a few v's to your > tftp setup: > > File: /etc/xinetd.d/tftp > Line to change: server_args = -s /tftpboot -v

I have checked and nothing is running on 1720, I even tried other ports Thanks ------=_Part_156950_6447813.1223890893752 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline <div dir="ltr">Hi<br>I am trying to get H323 to run on Asterisk, basically I had Asterisk running so I followed this tutorial <br><a

1. http://bugs.digium.com/view.php?id=12362 2. http://bugs.digium.com/view.php?id=12925 3. http://bugs.digium.com/view.php?id=12921 Also how do you go about changing details for device in DB and not using "sip realtime prune PEER" + 'sip reload'? Without that your changes to devices are not active. Good luck! Regards, Mindaugas Kezys http://www.kolmisoft.com >

<br> Or even:<br> <br> &nbsp;<a href=3D"http://www.blackbox.com/Catalog/Detail.aspx?cid=3D425,1423= ,1424&amp;mid=3D4946" target=3D"_blank">http://www.blackbox.com/Catalog/Det= ail.aspx?cid=3D425,1423,1424&amp;mid=3D4946</a><br> <br> (same thing from the UK site:)<br> <br> <br> <a

Hi Im also new but you should know very well all the interfaces you are going to connect the sistem, the number of users you'll have (hardware requeriments), know a lot about the soft/hardphones you'll use and download the asterisk handbook or the big one (i don't remember the name) Good luck Jose Manuel Cortes David X Semestre Ingenieria Electronica PONTIFICIA UNIVERSIDAD

YUP, this is the way that asterisk works. It is going to quelch all DTMF that goes out via a SIP gateway via asterisk. I spent a long time working this through and it has to do with the way that asterisk deals with DTMF and the DSP.c module that sits inband to the RTP/audio stream. There is a flag called DSP_DIGITMODE_NOQUELCH that is broken that might allow the inband DTMF after answer to work

<asterisk-users at lists.digium.com> Subject : Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall Date : Sat, 2 Feb 2008 18:25:16 -0700 > And the firewall is in between the phones and both servers or are you registering the phones on a local server and trunking to the other server through the firewall? > > In terms of nat and Cisco 7960s I've never had a

use depending on what the subnet mask is. The output provided shows two possible networks: 172.31.253.0/24 and 172.31.254.0/24. Or is this all part of the same address space with a different mask? If it is all the same space, then is the asterisk server network stack properly configured with a proper subnet mask? The bb can reach the asterisk server because it registers. Hope this helps On