[jitsi-users] simple Jitsi setup?


#1

I am admittedly pretty new to SIP. But I am pretty tech savvy with wide IT
background. (networking, *nix/windows servers, java/sql, scripting,
workstations, etc)

Anyway I have an asterisk server (with a dialer running on top of it), and
we have cisco 303 phones, and we normally just point them to the IP address
of the Asterisk server and enter username and password, and that's it.
They work. Jitsi, I am able to find place to enter my own proxy and
user/pass... In cisco phones it tells if it's "registered" or not - not
sure where to see that in Jitsi, I just see my status "online".

I try to dial a number (for our server 8-1-###-###-#### to manual dial) and
it says

`Failed to create the call. 81########## could not be resolved to an
internet address`

more info link says....

ยทยทยท

---------------------------------------------
java.lang.IllegalArgumentException: 813363030236 could not be resolved to
an internet address.
at
net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl.getIntendedDestination(ProtocolProviderServiceSipImpl.java:2592)
at
net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl.getOurSipAddress(ProtocolProviderServiceSipImpl.java:1854)
at
net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl.getOurSipAddress(ProtocolProviderServiceSipImpl.java:1831)
at
net.java.sip.communicator.impl.protocol.sip.SipMessageFactory.createInviteRequest(SipMessageFactory.java:773)
at
net.java.sip.communicator.impl.protocol.sip.SipMessageFactory.createInviteRequest(SipMessageFactory.java:878)
at
net.java.sip.communicator.impl.protocol.sip.CallSipImpl.invite(CallSipImpl.java:481)
at
net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImpl.createOutgoingCall(OperationSetBasicTelephonySipImpl.java:199)
at
net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImpl.createCall(OperationSetBasicTelephonySipImpl.java:141)
at
net.java.sip.communicator.service.protocol.media.AbstractOperationSetBasicTelephony.createCall(AbstractOperationSetBasicTelephony.java:122)
at
net.java.sip.communicator.impl.gui.main.call.CallManager.internalCall(CallManager.java:2715)
at
net.java.sip.communicator.impl.gui.main.call.CallManager.access$900(CallManager.java:60)
at
net.java.sip.communicator.impl.gui.main.call.CallManager$CreateCallThread.run(CallManager.java:2596)
---------------------------------------------

to "login" to the dialer software, you dial *77, when I do that I get

`"failed to create the call. sip:*77: missing hostname.`

more info link...

---------------------------------------------
java.lang.IllegalArgumentException: 81########### could not be resolved to
an internet address.
at
net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl.getIntendedDestination(ProtocolProviderServiceSipImpl.java:2592)
at
net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl.getOurSipAddress(ProtocolProviderServiceSipImpl.java:1854)
at
net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl.getOurSipAddress(ProtocolProviderServiceSipImpl.java:1831)
at
net.java.sip.communicator.impl.protocol.sip.SipMessageFactory.createInviteRequest(SipMessageFactory.java:773)
at
net.java.sip.communicator.impl.protocol.sip.SipMessageFactory.createInviteRequest(SipMessageFactory.java:878)
at
net.java.sip.communicator.impl.protocol.sip.CallSipImpl.invite(CallSipImpl.java:481)
at
net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImpl.createOutgoingCall(OperationSetBasicTelephonySipImpl.java:199)
at
net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImpl.createCall(OperationSetBasicTelephonySipImpl.java:141)
at
net.java.sip.communicator.service.protocol.media.AbstractOperationSetBasicTelephony.createCall(AbstractOperationSetBasicTelephony.java:122)
at
net.java.sip.communicator.impl.gui.main.call.CallManager.internalCall(CallManager.java:2715)
at
net.java.sip.communicator.impl.gui.main.call.CallManager.access$900(CallManager.java:60)
at
net.java.sip.communicator.impl.gui.main.call.CallManager$CreateCallThread.run(CallManager.java:2596)
---------------------------------------------

Any direction would be greatly appreciated. I have full root access to the
servers, firewalls, workstations, etc. I'm new to SIP, but not
networking/firewalls/servers. A point in the right direction is likely all
that I need.

Thanks!

p.s. in the cisco phones we do define a dial plan. (i.e. the types of
character strings that qualify as something to dial and how long to wait
until we dial or wait for more digits if we have a match, etc, etc.... I
don't see anything like that in jitsu, but maybe I am missing it, or maybe
there is a default plan that I just need to abide by.

Any direction would be greatly appreciated. I have full root access to the
servers, firewalls, workstations, etc. I'm new to SIP, but not
networking/firewalls/servers. A point in the right direction is likely all
that I need.

*Ben Beetle*
*Opinion Access Corp.*
*bbeetle@OpinionAccess.com*
*347-472-0533*

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#2

I am admittedly pretty new to SIP. But I am pretty tech savvy with wide IT
background. (networking, *nix/windows servers, java/sql, scripting,
workstations, etc)

Anyway I have an asterisk server (with a dialer running on top of it), and we
have cisco 303 phones, and we normally just point them to the IP address of
the Asterisk server and enter username and password, and that's it. They
work. Jitsi, I am able to find place to enter my own proxy and user/pass...
In cisco phones it tells if it's "registered" or not - not sure where to see
that in Jitsi, I just see my status "online".

I try to dial a number (for our server 8-1-###-###-#### to manual dial) and
it says

`Failed to create the call. 81########## could not be resolved to an
internet address`

You most likely accidentally created a registrarless SIP account, i.e. just a name/number without a server address. Since you're using Asterisk, you'd need to create a regular SIP account, e.g. 1001@example.com or 1001@192.168.1.1

more info link says....

---------------------------------------------
[...]
---------------------------------------------

to "login" to the dialer software, you dial *77, when I do that I get

`"failed to create the call. sip:*77: missing hostname.`

more info link...

---------------------------------------------

[...]
---------------------------------------------

Any direction would be greatly appreciated. I have full root access to the
servers, firewalls, workstations, etc. I'm new to SIP, but not
networking/firewalls/servers. A point in the right direction is likely all
that I need.

Thanks!

p.s. in the cisco phones we do define a dial plan. (i.e. the types of
character strings that qualify as something to dial and how long to wait
until we dial or wait for more digits if we have a match, etc, etc.... I
don't see anything like that in jitsu, but maybe I am missing it, or maybe
there is a default plan that I just need to abide by.

There are no dialplan/pause options in Jitsi.

Any direction would be greatly appreciated. I have full root access to the
servers, firewalls, workstations, etc. I'm new to SIP, but not
networking/firewalls/servers. A point in the right direction is likely all
that I need.

Ben Beetle

Ingo