Though i am not an exper on digiumt, as i look at it, it seems like 2 RJ-48 network ports on your digium card.
(might sound arrogant, but why buy such a card if you dont know what it does?)
As i read from digiums page, it looks like it is a network card with some extra hardware on it to provide better sound quality over the Ethernet.
An ISDN line to asterisk would require a card with RJ-11 connectors. These you can find here
these analog cards should make your server able to recieve an ISDN connection on it.
An company who specializes in laying networks is best to use for this. if you want to do it yourself, normal UTP cables can be used, look on wikipedia for more info on those. though i recommend the company (they give ensurance)
Patch panel for 50 connections? i advise using a Mesh Topology network. more info here
Meaning you’d have multiple switches on which devices would be plugged, and those switches are connected to eachother. preferable, have the server connected to multiple switches too.
[quote]
Q5. Do i need any other cards for it? [/quote]
as i said on 2, you might need to place an Analog card into your server
The card you already have is a good one to get VoIP over since it has extra hardware on it that improves sound over the line.
If you need more info, post here ofc or try a digium reseller / digium helpdesk
Nope not at all arrogant its so true why buy anything if you dont know what it does/uses…!
actually i didnt brought that card its my company by some guesses and without proper calculation of things and without consulting me brought it and handed over to me take this and make asterisk work and then kick out nortel bcm 400, im clueless with pbx or telephony since then trying get help in here some nice guys have helped me here im so thankful to them my connection issues with TE212P is solved now playing around with Users.conf, sip.conf, extensions, dialplans etc.
Once im ready to make asterisk i will post everything on the net with screen shots so everyone can get help.
it might be a small challenge but adding LOADS of users to the sip.conf is kinda nasty. if you would need to maintain the system you are making you should try to get the user database to either sql or LDAP i dont know about sql, but the LDAP module in asterisk 1.6 works rather nice making and editing users in LDAP is much easier
[quote=“COpeter”]lolz, im amazed at the fast reply XD
Playing around with asterisk is fun
it might be a small challenge but adding LOADS of users to the sip.conf is kinda nasty. if you would need to maintain the system you are making you should try to get the user database to either sql or LDAP i dont know about sql, but the LDAP module in asterisk 1.6 works rather nice making and editing users in LDAP is much easier
zomg… i use quite alot…[/quote]
yeah its really fun but brings frustration too at times, right now, fast replied cause right now im just teasing Asterisk LOL.
can you explain to me how to just add two users (soft phone only)
and make them call each other INTERNAL ONLY no outbound in bound from outside world but purely LAN call?
[quote=“COpeter”]lolz, im amazed at the fast reply XD
Playing around with asterisk is fun
it might be a small challenge but adding LOADS of users to the sip.conf is kinda nasty. if you would need to maintain the system you are making you should try to get the user database to either sql or LDAP i dont know about sql, but the LDAP module in asterisk 1.6 works rather nice making and editing users in LDAP is much easier
zomg… i use quite alot…[/quote]
True playing around with Asterisk is FUN and Frustration at the same time for some ignorant like me…!
can you please give me a detailed explaination of how to setup two softphones register them with asterisk and call each other (Internal only)
no outbound or inbound from out side world but only local extension and can call each other thats it…im trying since yesterday but to no avail…please?
i got 2 standard users added too for testing purposes. the rest of my users should be just LDAP ones
in sip.conf (totally at the bottom i have)
[STDUSR01]
type=friend
callerid="STDUSR01" <0001>
username=0001
secret=abc123
context=standard
host=dynamic
dtmfmode=rfc2833
nat=no
allow=all
qualify=yes
insecure=very
canreinvite=yes
this makes the user (i’ve added a STDUSR02 too with the same things though different username ofc) oh, i’ve just pasted these on the total bottom of sip.conf
in extensions.conf
[standard]
exten => 0001,1,dial(SIP/0001,10,t)
exten => 0002,2,dial(SIP/0002,10,t)
This is to allow calls
further in the sip.conf you might need to edit some general settings
bindport=5060
bindaddr=0.0.0.0
realm=example.com (your domain name)
allow=all
the functions are already in there, so you might need to search for them for a bit
Maybe changing nat=no to yes. might help, also which softphone do you use? i use X-Lite and it connected properly.
Often it can also be settings in the softphone.
note that username with these is STDUSR01, as well is the auth-id. other then that i dont know :S if it tries to connect to asterisk it should give notion of errors in the asterisk CLI.
in the command line use
to get to the asterisk CLI. then let your softphone connect, if the phone settings are correct it should give notion in there as to what might be the problem ^^
[quote=“COpeter”]Maybe changing nat=no to yes. might help, also which softphone do you use? i use X-Lite and it connected properly.
Often it can also be settings in the softphone.
note that username with these is STDUSR01, as well is the auth-id. other then that i dont know :S if it tries to connect to asterisk it should give notion of errors in the asterisk CLI.
in the command line use
to get to the asterisk CLI. then let your softphone connect, if the phone settings are correct it should give notion in there as to what might be the problem ^^[/quote]
Im also using X-lite…!
can please let me know what to add for Service Provider?
i am using the 3.0.2 version with settings like these:
Display name: STDUSR01
Username: STDUSR01
password: abc123
auth. name: STDUSR01
domain: (in my case walbeekgroep.local cause i used that in my sip.conf as Realm=)
Register with proxy: 10.10.10.249 (my asterisk servers internal IP)
[quote=“COpeter”]i am using the 3.0.2 version with settings like these:
Display name: STDUSR01
Username: STDUSR01
password: abc123
auth. name: STDUSR01
domain: (in my case walbeekgroep.local cause i used that in my sip.conf as Realm=)
Register with proxy: 10.10.10.249 (my asterisk servers internal IP)[/quote]
So no real service provider needed we can add our own under “Custom Voip provider” ?
[quote=“COpeter”]i am using the 3.0.2 version with settings like these:
Display name: STDUSR01
Username: STDUSR01
password: abc123
auth. name: STDUSR01
domain: (in my case walbeekgroep.local cause i used that in my sip.conf as Realm=)
Register with proxy: 10.10.10.249 (my asterisk servers internal IP)[/quote]
So no real service provider needed we can add our own under “Custom Voip provider” ?
with the same things though different username ofc) oh, i’ve just pasted these on the total bottom of sip.conf