#64 Line continues to ring after call is connected

[Anonymous]

Originally created by: voip%nom...@gtempaccount.com
Originally owned by: r3gis...@gmail.com

What steps will reproduce the problem?
1. Dial out any valid number
2. The Called Party Picks up
3. The Caller continues to hear ringing.  The ringing continues even after the call has ended.  The only way to stop the ringing is to restart app.

What is the expected output? What do you see instead?
I expected the the ringing to stop when the called party picked up.

What version of the product are you using? On what operating system?
I'm using version: 0.00-12 with MOTOROLA DROID Rooted (CyanogenMod?-5.0.6.2-Dr0id).

Please provide any additional information below.

[Anonymous]

Originally posted by: zdevel

that can be problem with voip provider, btw. I faced same problem with one voip provider.

[Anonymous]

Originally posted by: r3gis...@gmail.com

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.

Owner: r3gis.3R
Status: Accepted

[Anonymous]

Originally posted by: voip%nom...@gtempaccount.com

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.

[Anonymous]

Originally posted by: kro...@gmail.com

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.

[Anonymous]

Originally posted by: r3gis...@gmail.com

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.

[Anonymous]

Originally posted by: r3gis...@gmail.com

The new version :
https://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.

[Anonymous]

Originally posted by: voip%nom...@gtempaccount.com

I'm still having the same issue. My configuration is simple... 2 sip extensions on asterisk version  1.6.2.9.

[Anonymous]

Originally posted by: r3gis...@gmail.com

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).

[Anonymous]

Originally posted by: voip%nom...@gtempaccount.com

Thanks for fixing this. This app in now working beautifully.

[Anonymous]

Originally posted by: r3gis...@gmail.com

(No comment was entered for this change.)

Status: NextRelease