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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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rxfax now prints header data at top of image
by Jeff Pyle 30 Sep '20

30 Sep '20

Hello, I've upgraded a FreeSWITCH-based fax receiving application from 1.2 to 1.10. Now on 1.10 I see a fax header printed onto the top of the TIFF image that wasn't there in 1.2. Is there parameter or variable I can set to suppress it? Regards, Jeff

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Freeswitch and tel: uri for bridge ?
by Marcel Haldemann 29 Sep '20

29 Sep '20

Hey Guys, I must use a tel: Request URI for outbound Calls. Currently using FreeSwitch 1.10.3 I triy it this way: <action application="bridge" data="{sip_invite_req_uri=tel:+4199999999} sofia/carrier/+41999999999@${network_addr}"/> However i get the Error: URL Error! tel: uri's not supported at this time Does this mean Freeswitch does not support tel: URIs for bridge at all ? Is there any way to call a tel: URI ? Kind regards, Marcel

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Compact headers
by David Villasmil 28 Sep '20

28 Sep '20

Hello guys, Does freeswitch support mixing of compact and long headers? Thanks -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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CTI Web Application for Freeswitch
by Malik Ehtasham 28 Sep '20

28 Sep '20

Hi We would like to develop the Freeswitch CTI Web Application, from where a user can control all the CTI Features for example -Answer Call -hold/resume call -Consult/Transfer/Consult Transfer/Conference -Outbound Call I am looking for a library such as Cisco CUCM provides JTAPI <https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/jtapi_dev/10_5_2/C…> Do we have something similar in the Freeswitch, For now We are exploring Mod_commands <https://freeswitch.org/confluence/display/FREESWITCH/mod_commands> , can this be used to acheive such a functionality, Though we tried few of its commands to answer the 2 phones but it didn't work. if mod_commands is not the option than what other option or track we should follow to acheive this . Any help/hint would be really helpful !! Thank you -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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Distorted sound for one of conference members when AMR-WB codec used
by Rafał Telaczyński 28 Sep '20

28 Sep '20

Hi All, We are using FreeSWITCH to put 2 members on a conference and record it. The problem is that when AMR-WB codec is used, the second member to join the conference hears distorted sound(sometimes it lasts couple seconds, in some cases it lasts until the end of the call). There is no such sound heard by first member participant, nor is it generated by it(verified by decoding audio from RTP packets sent from first member participant towards FreeSWITCH). Moreover, distorted sound to second member can be heard after decoding RTP packets sent from FreeSWITCH towards second conference participant - proof that it is not the issue with second participant phone or so. Intrestingly, there is no such issue when AMR codec is used(tested by removing AMR-WB from global_codec_prefs) - the sound is clear for both participant throughout the whole duration of the conference. Considering that everything is OK with AMR codec used, it seems like the issue is not with conference itself, only using AMR-WB during conference. We tested with some of the FreeSWITCH parameters related to RTP(e.g. auto-rtp-bugs, rtp-rewrite-timestamps, send_silence_when_idle, etc.) but it did not help. Is it known issue? Did anyone faced something similar before? What else can be done to make it work? Please, let me know if you have any questions regarding the topic/need more info from my side. Thanks, Rafal

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Call Center
by Nathan Stratton 28 Sep '20

28 Sep '20

Any way to log off an agent if we lose SIP registration? ><> nathan stratton

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FS Not restarting RTP after unhold
by David Villasmil 25 Sep '20

25 Sep '20

Hello all, I have a scenario where FS is not restaring RTPs after an UNHOLD reInvite. The scenario is like this: FS--------------------------------------Endpoint <<<<<<<<<<<<< established call >>>>>>>>>>>>>>> |<---------- INVITE (SDP sendonly) ----------| |------------------ Trying -------------->| |----------- 200 OK (SDP recvonly)---------->| |<------------------ ACK -----------------| <<<<<<<<<<<<<<<<< ALL GOOD >>>>>>>>>>>>>>>>>>> |<------------ INVITE (no SDP) --------------| |------------------ Trying -------------->| |----------- 200 OK (SDP sendrcv) ---------->| |<----------- ACK (SDP sendrcv) -----------| But freeSWITCH never starts sending RTP after the UNHOLD... Don't know why. Anyone encountered this? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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