October 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Clone freeswitch-contrib.git
by Ing Sergio Basurto J. 02 Oct '20

02 Oct '20

Hello, Anyone knows how can I clone contrib repo?, because when I ran: git clone https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git Cloning into 'freeswitch-contrib'... Username for 'https://freeswitch.org': Password for 'https://freeswitch.org': fatal: Authentication failed for 'https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git/' Ask for username and password, there is a default username and password? I just want the module mod_odbc_query, am I missing something? Any help will be appreciated!! Best Regards, --

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Force a reINVITE?
by Nathan Stratton 02 Oct '20

02 Oct '20

I am starting a call with OPUS only on the B leg, the call starts fine with inbound-late-negotiation and OPUS is sent back on the A leg. However, the call is then transferred to a user with OPUS and VP8 video with the new INVITE, but nothing ever is sent back to the original A leg to update VP8. Is it possible to send a reINVITE back so I can update the SDP with VP8 on the original A leg? ><> nathan stratton

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Microsoft Teams integration
by Stefan Kainz 01 Oct '20

01 Oct '20

Hello, we're trying to use Freeswitch as a Microsoft Teams direct routing SBC. So far everything works perfectly, except getting the call back from hold on an outbound call. The Problem is that Microsoft expects a Record-Route Header in every Request and every Response. I have been able to set/rewrite this Record-Route header with sip_h_ and sip_rh_ in most szenarios, but like i said - unhold on an outgoing call does not work. The hold is initiated with a reinvite with a=inactive in sdp, which does work. However, the unhold - also a reinvite with a=sendrecv in sdp does not work correctly. In the sip-trace i see that the OK ( in response to the INVITE for unhold ) gets retransmitted a couple of times, because the ACK from microsoft isn't received. This is due to the missing Record-Route Header in the OK Message. I have seen this behavior on inbound calls too, when i didnt set the Record-Route Header. Now my question: Is there any way to add the Record-Route Header to every Message on a sip-profile or gateway? Or if not, is it possible to handle a reinvite in dialplan so i could set the header manually? Maybe there is somebody who already has this working that could point me in the right direction. I know freeswitch isn't a SBC, but it would be nice to be able to use freeswitch as a Teams-SBC without Kamailio or something similar. Any help is very much appreciated 🙂 regards, Stefan

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