search for: hacklocalhost

...(yes/no) + CallerID String + Name + Number + Enable mailbox indication? + Mailbox number(s) to be associated with this channel + Context Would you like to setup registration for FWD, IAXtel, SIPPhone or iptel? + -- +------------------------------------------+ |Leif Madsen - http://www.hacklocalhost.com| +------------------------------------------+ | @| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969 sipph| 1-747-386-1618 | +------------------------------------------+ -- +------------------...

...ver, then it crashes. If I use the mp3player() app then it crashes. Using mpg123 -t filename.mp3 seems to be fine. If you have any idea's on things I can test for, it'd be greatly appreciated. Thanks in advance, +-------------------------------------------+ |Leif Madsen - http://www.hacklocalhost.com | +-------------------------------------------+ | @| leif at hacklocalhost dot com | | FWD| 18924 | | IAX| 1700-363-0761 | | SMS| sms at hacklocalhost dot com | | ICQ| 3445119 | |iptel| 8972-1969...

...The inside_net is the ip address of > the asterisk server, and the inside_mask is the subnet mask. At least > that is how I have mine setup in my sip.conf, and it works. > > inside_mask for the internal mask would make more sense to me as well :) > > -- > Leif Madsen <leif@hacklocalhost.com> > http://www.hacklocalhost.com In my configuration I have internal SIP clients registering from 192.168.0.0/28 and my * address is at 192.168.0.100. Using the host address of the * box as the inside_net variable the audio from 192.168.0.0/28 was sent to the outside_addr variable giving...

.../path/to/patch Nothing should fail. cd /usr/src/asterisk/ make cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/ Restart your Asterisk and try it. If you want to call a NAT'd Asterisk box, my Free World Dialup number is 18924. Currently online. -- Leif Madsen <leif@hacklocalhost.com> http://www.hacklocalhost.com

...f my SIP calls will still work. Box right now is a RH9 computer using iptables as the FW. I wouldn't mind placing my * box behind it, but I'm wondering if anyone has actually gotten NAT working with *? Thanks, -- +------------------------------------------+ |Leif Madsen - http://www.hacklocalhost.com| +------------------------------------------+ | @| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969 sipph| 1-747-386-1618 | +------------------------------------------+

...If anyone has any suggestions it would be greatly appreciated. I saw some software called Laboratory Document Control System, but I don't have $999 for a single user license :) (or any money at all actually) Thanks, -- +------------------------------------------+ |Leif Madsen - http://www.hacklocalhost.com| +------------------------------------------+ | @| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969 sipph| 1-747-386-1618 | +------------------------------------------+

Grandstream and Global IP Sound have inked a deal in which Global IP Sound will provide its royalty free iLBC codec to Grandstream. GS will integrate this codec into the BT and HT product lines

Hello all I would like change the all announcemennt(Voicemailmain,Voicemail etc.). But I don't know how to change the these each prompt. Do we have any guide book for this? Please teach me about changing the voicemail or other prompt. Thanks

...lt I would also like suggestions from any bash script coders on anything I may not be doing very effectively / wrong (for instance, I don't feel that the way I'm checking to see if the sources already exist is the best way to do it, but it works for me) You can find the code at http://www.hacklocalhost.com/asterisk/_asterisk-update Hopefully someone finds this useful. -- Leif Madsen <leif@hacklocalhost.com> http://www.hacklocalhost.com

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

In attempting to make my asterisk server the latest and the greatest tonight I attempted to upgrade my CVS. In the asterisk directory I ran the make clean, executed the CVS update -d, and after all the files completed ran make upgrade. My problem is that when I pull up the CLI the cvs version that is showing is the same date as my initial install. Does this mean that the upgrade did not go

Hello-- I've been asked an interesting question, and I'm too ignorant to answer it authoritatively (yet). Can anyone help me? Question: If I'm going to implement a somewhat small (10-80) phone system, and I have a choice of using VOIP phoneset (like SNOM or Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly what features will I kiss goodbye if I use the cheap

Hi all, I would like people to email me at 'leif at hacklocalhost dot com' some example configuration files for VoIP providers which * can register with. I am going to expand upon the FWD php "wizard" I created for these other providers, but I need some examples as I don't actually use anything but IAXtel and FWD. So far sipphone and iaxte...

Hi, I have a strange problem trying to update Asterik on one of my boxes. I have done the following: - delete all the old source files - download the new file using: cvs checkout zaptel zapata libpri asterisk - compiling the new source file using: make clean ; make install Everything is compiled fine, except.... when I stop and start Asterisk server, it is still the old CVS version:

Has anyone tried installing * on a box with two eth interfaces which is acting as a NAT box? I have only one IP at this point and I would like to get * working without all of the NAT issues. My idea is to run * on my gateway (which is also running the firewall and masquerade services). All of my UAs (Grandstream + Xten X-LITE + gnophone) will be inside the NAT screen, and will connect to the *

I'm trying to get the total Linux/* installation size as small as possible. I'm wondering if anyone has looked at the installed packages list from the Redhat installation [rpm -qa] and has parsed out all packages not needed for * to run. I follow the custom install guide from Andy Powell but the installation yields 948+ Meg with 340 installed packages. I'm sure most of those packages

Go to www.google.com type in your search query add this to the end of your search query: site:lists.digium.com e.g. http://www.google.com.au/search?hl=en&ie=utf-8&oe=utf-8&q=Australia+site:lists.digium.com The mailing list used to be on www.marko.net, I'm not sure if the whole archive was moved across, you might want to search with site:www.marko.net OR site:lists.digium.com

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as

Hi, I've been on the list for slightly under a month now and noticed; a) a fairly high amount of traffic, b) a lot of questions which come up more than once, c) the archives at lists.digium.com are not searchable. I have started development to import the mailinglist archives into a MySQL database and creating a full text search possibility on this. My questions; 1) Is this already done

Hi all, I had a chance to use some call conferences that had some very neat functionalities: - When you call you are first asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have