GoogleCodeExporter
commented
9 years ago About point 1, it's usually linked to the sip provider. In fact CSipSimple
allows you to configure some field that are defined in SIP protocol. Depending
on the sip provider and the remote side field can have different meaning. On
most betamax clones I tested the way I set the "caller-id" which in fact for
regular SIP protocol is the SIP username produces that betamax sends this on
regular landline calls as number.
On a regular sip client, it has all informations so it can display anything it
wants.
If you want to play more with that here's what you can test :
Long press your account row in the list of accounts, select "Choose wizard" and
choose "Expert".
Then simple click the row, and you'll see now a lot of settings to configure
the account. Here you are at very lower level of SIP protocol and can configure
everything that is normalized in SIP protocol. It made no assumption on the SIP
provider.
Here you'll see several things,
the SIP ID : (aka AoR). It identify the sip user address. it's something like
"Display Name" <sip:username@domain>
the < and > char are important. For betamax clones the "Display Name" is not
taken into account while making a call, but the "username" is important. While
on most other sip provider the username is the same than authentication name,
here their servers allows you to use a different username than the
authorization name.
On the PC receiver, as it get the complete "Display Name" <sip:username@domain>
it can decide to display anything it wants. It really depend on the sip client
you have on the other side. In CSipSimple I display the "display name", and try
to resolve and search inside contacts using both display name and username.
You'll see also a "username/login" field in this mode. This one is the
authentication name. It's what's is used to authenticate on betamax server.
Normally no need to change it as correctly filled by the betamax wizard
previous setup.
About point 2, it's probably because the sip server doesn't acknowledge the
CANCEL. In this case, as the SIP dialog is not yet considered as finalized
(because it's not acknoledged from other side that call was cancelled), the
application stay in the call screen unless the SIP dialog times out. This
timeout has a normalized value. So what you observe reflect the state of the
call. Since cancel not acknowledged from other side, it remains like that. This
can happens when there is some problem on server. It could also happen because
of something with configuration. Maybe I can find a clue if you collect and
send me logs. Turn on log collection, reproduce the problem, and just after
stop and send logs :)
About point 3 :
For the particular point of "603/Decline" it's not a string from the
application but from SIP protocol itself. So safer to keep it untranslated.
However, you're right, many things are not fully translated in Spanish and I
need help for translations :).
You can join translation effort here :
https://www.transifex.net/projects/p/csipsimple/ . There's no team yet for
spanish, so just request and I'll give you rights on the group.
Original comment by r3gis...@gmail.com on 3 Jul 2012 at 8:52
- Changed state: Need-Details
Original issue reported on code.google.com by
yed...@gmail.comon 3 Jul 2012 at 1:08