Problems of Siptosis with new Skype 4.3 for Linux

角色

最近用QANP TS-435 Pro行VM，尝试用siptosis做gateway，但是接通后却没有Skype能打入来，但是没有声音，最后发现是Skype 4.3改了audio方式。

Solution:

Install the pulseaudio package (agt-get install pulseaudio) such that it can be seen by Skype as shown in the following diagram:

下載 (34.01 KB)

2015-2-1 15:38

In order to have better voice quality, it is advised to switch off the option of "allow Skype to automatically adjust my mixer levels".

角色

source：http://community.freepbx.org/t/s ... dden-rtp-lack/24477

[SOLVED] FreePbx SipToSis and Skype on same machine : sudden RTP lack


I'm using succesfully a sip trunk between freepbx and SipToSis to manage Skype calls
All is installed on same machine and works fine for months until the previous Skype needed to be replaced due to changes in authentication.
Done this, all calls are placed correctly but I have no audio both directions.
I have modified nothing other than new Skype installation (just a .bz2 decompressed on new folder and made run) ; usual FreePbx modules upgrade.
It seems to be an RTP issue:
chan_sip.c:28883 check_rtp_timeout: Disconnecting call 'SIP/Skype-0000001d' for lack of RTP activity in 31 seconds
(then Asterisk drops the call)
Since RTP packets are managed between Asterisk side of SipToSis and FreePbx trunk itself (where nothing has been modified) it's quite tricky to understand what is happened.
I think it isn't a NAT issue or something like, as both peers reside inside the same machine, with the same IP address on different ports.
SipToSis trace reports 0 RTP packets received.
RTP debug on FreePbx traces a lot of packets between its IP address and the related IP phone address , nothing more.

Does anybody experienced something like this ?
Where can I start to dig deeper for solving the issue ??

Thank you

===================
Just done a fresh SipToSis/Skype installation into an already existing updated FreePBX using the same documented step by step procedure I always used for this.
The behaviour is the same , no audio both ways : perhaps something changed with latest upgrades ?
Any hint ??
Thanks.
=========================
Sorry...My fault on diagnosis : after some hours of debug on wrong place, I discovered the broken pipe between Skype and SipToSis :
Asterisk----SipToSis---xxx----Skype

" As of version 4.3, the ALSA sound system is no longer supported without PulseAudio." Skype says.....so no PulseAudio no voice toward SipToSis thus no packet to FreePbx then RTP timeout.
Just added PulseAudio to CentOS and all is loud and clear as usual.
Hope this help ...

角色

Possible way may resolve the audio issue:

http://www.mhspot.com/sts/siptosis_config_issues.html

http://mrblog.org/2010/10/12/han ... -sip-skype-gateway/

ckleea

Have fixed the problem.

Skype and siptosis are still very good tools for our VOIP asterisk