Closed GoogleCodeExporter closed 9 years ago
that can be problem with voip provider, btw. I faced same problem with one voip
provider.
Original comment by zdevel
on 27 Jun 2010 at 7:08
I assume you are talking about ringback.
Some SIP trace could be valuable.
It's probably due to a not well managed sequence on my side. This specific
sequence, as zdevel said, can be due to the voip provider but should be handled
properly by CSipSimple.
If you can send me (directly by mail - it's better for your information
privacy) a logcat of what is happening. It could help me to see what's going
wrong and when the ringback should be stopped.
Original comment by r3gis...@gmail.com
on 28 Jun 2010 at 9:18
I'll try to get a sip trace to you later... but right now I get this condition
with csipsimple version: 0.00-12 simply calling another extension. I just
updated Asterisk to 1.6.2.9 from 1.6.2.0-beta4 and still get the same effect.
Original comment by voip%nom...@gtempaccount.com
on 29 Jun 2010 at 5:29
I have the same problem with sip2sip.info. Given that sip2sip accounts are
free, hopefully the problem can be easily diagnosed. I had to add account
using expert as I suspect the proxy matters.
Original comment by kro...@gmail.com
on 13 Jul 2010 at 6:41
Ok i've just opened a sip2sip account.
In fact, call management appears extended with sip2sip :
When you receive a call, there is in reality 2 call simultaneously handle by
sip.
As for now I've made code only for one call... Everything become really buggy
when the second call say it is disconnected (since call is complete elsewhere)
and the first call say it is confirmed.
So to solve this issue, I have to start the work on multiple call management.
Original comment by r3gis...@gmail.com
on 13 Jul 2010 at 8:42
The new version :
http://code.google.com/p/csipsimple/downloads/detail?name=CSipSimple_0.00-12-05.
apk
should be better with servers that announce two calls (sip2sip, asterisk with
specific configuration...).
Uninstall previously installed version before installing this one.
Original comment by r3gis...@gmail.com
on 22 Jul 2010 at 8:53
[deleted comment]
I'm still having the same issue. My configuration is simple... 2 sip extensions
on asterisk version 1.6.2.9.
Original comment by voip%nom...@gtempaccount.com
on 23 Jul 2010 at 3:09
Ok, I reproduced the issue on my openser. Fixed in my branch, will be delivered
with next build.
The cause was the fact sip server made multiple "Ringing" response (besides
it's not each time, but once it's made, then each next call will have the
issue).
Original comment by r3gis...@gmail.com
on 26 Jul 2010 at 12:34
Thanks for fixing this. This app in now working beautifully.
Original comment by voip%nom...@gtempaccount.com
on 10 Aug 2010 at 10:44
Original comment by r3gis...@gmail.com
on 29 Aug 2010 at 11:04
Original issue reported on code.google.com by
voip%nom...@gtempaccount.com
on 26 Jun 2010 at 11:51