Hi, I am new with Asterisk, since my company wants to provide whole communication (instant messinging as well as voice communication) on company servers I want to ask if this is possible with asterisk.
What I need is one server that can provide us with instant messinging and voice calls amongst the users, the question is what do I need of Asterisk and what clients do I use for it, can someone explain me whole procedure how is this doable.
Thank you for you answers.
Digium’s Switchvox SMB product makes use of Asterisk and ejabberd in order to provide both voice calling and text messaging between users. For more information, see:
Hi,
I’m trying to learn reading your post. Really thanks you very much for all help. But I have a question, Do I need to have a VoIP provider in order to call from one laptop with Asterisk to another laptop with also Asterisk (both of them with Softphones)? I can call from one laptop to itself but I cannot comunicait each other. Could you help me, please?
Thanks you,
Regards,
Isa
Howdy,
No, you’d simply create a trunk between the two laptops over SIP or IAX.
Yesterday, I did it. But it did not change. I don’t know why but I try to do it and when I write SIP show registry it does not register anything. I can call only to the extensions I have created in the same server.
Do you know why?
I try to install another VM in the same laptop and comunicate each other, and it does not work… Iwrite you here the configuration of my two diferent laptops both with AsteriskNOw running in Centos in VMware.
Laptop 1: asterisk 192.168.140.128 extensions (1000-1099), Softphone: X-Lite, Zoiper
Laptop 2: asterisk2 192.168.200.134 extensions (1800-!899), the same Softphone, Zoiper
Configuration (I copy it, I cannot copy-paste it because it is in a VMware):
sip.conf
[general]
context=default
allowoverlap=no
bindport=5060
binaddr=0.0.0.0
srvlookup=yes
language=en
videosupport=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=no
register=>asterisk2:asterisk2@192.168.200.134/asterisk
[asterisk]
type=friend
host=dynamic
secret=1001
context=users
disallow=all
allow=ulaw
allow=alaw
allow=gsm
insecure=invite
[1001]
type=friend
host=dynamic
secret=1001
context=users
[1000]
type=friend
host=dynamic
secret=1001
context=users
[1009]
type=friend
host=dynamic
secret=1001
context=users
extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
[users]
exten=>1009,1,Answer()
exten=>1009,2,Playback(hello-world)
exten=>1009,4,Hangup()
exten=>1001,1,DIAL(SIP/1001,10)
exten=>_18XX,1,NoOp()
exten=>_18XX,n,HangUp()
And in the other laptop the same but with 18…
it tells me:
[Jun 6 18:20:34]NOTICE[3511]: : :No registration for peer ‘1001’ (from 192.168.140.1)
[Jun 6 18:20:34]NOTICE[3517]: : :No registration for peer ‘1001’ (from 192.168.140.1)
==Using SIPRTP CoS mark 5
–Executing [1801@users:1]NoOp(“SIP/1001-00000002”,"")in new stack
–Executing [1801@users:2]Dial(“SIP/1001-00000002”,“SIP/asterisk2/1801,30”)in new stack
[Jun 6 18:20:34]WARNING[11002]: : : No such host:asterisk2
[Jun 6 18:20:34]WARNING[11002]: : : Unable to create channel of type `SIP’ (cause 20- Unknown)
==Everyone is busy/congested at this time (1:0/0/1)
–Executing [1801@users:3] Hangup(“SIP/1001-00000002”,"") in new stack
==Spawn extension (users,1801,3) exited non-zero on ‘SIP/1001-00000002’
[Jun 6 18:20:34]NOTICE[3512]: : :No registration for peer ‘1001’ (from 192.168.140.1)
[Jun 6 18:20:34]NOTICE[3515]: : :No registration for peer ‘1001’ (from 192.168.140.1)
And the phone tells me: Temporarily Unavailable
I do not know what happens?Could you help me, please?
Thanks you
Isa