March 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Multi tenant / user 1 in domain A can not reach user 2 in domain B
by Thorsten Göllner 15 Mar '18

15 Mar '18

Hi, I setup freeswitch with fusionpbx in a multi tenant evironment. Works fine. But currently I can not call "cross domain". So user 1 in domain A can not reach user 2 in domain B. When calling my log show the following: 2018-03-14 12:37:39.580028 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1@domain-A [19bbf41c-277c-11e8-81fa-4dba4b7d93a7] 2018-03-14 12:37:39.580028 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1@domain-A) Running State Change CS_NEW (Cur 1 Tot 84) 2018-03-14 12:37:39.580028 [DEBUG] sofia.c:9873 sofia/internal/1@domain-A receiving invite from 1.2.3.4:54948 version: 1.6.19 64bit 2018-03-14 12:37:39.580028 [DEBUG] sofia.c:10044 IP 1.2.3.4 Rejected by acl "domains". Falling back to Digest auth. 2018-03-14 12:37:39.580028 [WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile 'internal' for [2@domain-B] from ip 1.2.3.4 2018-03-14 12:37:39.580028 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/1@domain-A) State NEW 2018-03-14 12:37:39.580028 [DEBUG] sofia.c:2334 detaching session 19bbf41c-277c-11e8-81fa-4dba4b7d93a7 2018-03-14 12:37:39.619980 [DEBUG] sofia.c:2442 Re-attaching to session 19bbf41c-277c-11e8-81fa-4dba4b7d93a7 2018-03-14 12:37:39.619980 [DEBUG] sofia.c:9873 sofia/internal/1@domain-A receiving invite from 1.2.3.4:54948 version: 1.6.19 64bit 2018-03-14 12:37:39.619980 [DEBUG] sofia.c:10044 IP 1.2.3.4 Rejected by acl "domains". Falling back to Digest auth. 2018-03-14 12:37:39.639961 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7fdb641160c0 Connected. 2018-03-14 12:37:39.639961 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7fdb641160c0 released. 2018-03-14 12:37:39.639961 [WARNING] sofia_reg.c:2906 Can't find user [12(a)telequest.qcall.cloud] from 1.2.3.4 You must define a domain called 'telequest.qcall.cloud' in your directory and add a user with the id="12" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2018-03-14 12:37:39.639961 [WARNING] sofia_reg.c:1737 SIP auth failure (INVITE) on sofia profile 'internal' for [2@domain-B] from ip 1.2.3.4 Is there a special setting or configuration I missed? Is it a ACL issue? Thanks in advance, Thorsten

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Verto - audio only between FS and client
by Rick Jarvis 14 Mar '18

14 Mar '18

What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why: My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server. So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS. The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code. I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now! Hoping for some ideas or guidance. Thanks R

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Multi-Tenant in Freeswitch
by MOWHABUTH MOHAMMAD SHAFDAAR ALLY 14 Mar '18

14 Mar '18

Dear all, I have been using freeswitch in the default context for the past few months. Everything works fine in the default context. Now i want to use multi-tenancy and i have followed all the steps at https://wiki.freeswitch.org/wiki/Multiple_Companies . However, i am unable to register my softphone, with user 1000 in the company-a.org directory, to the server. I am trying on CSipSimple(softphone) the credentials as follows: Account Name: 1000 User:1000 Server: company-a.org Password: 1234 Bad Gateway is the response. Can anyone please help me with this? Mowhabuth Shafdaar BSc (Hons.) Applied Computing Faculty of Informations, Communications and Digital Technologies University of Mauritius <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> -- <http://www.uom.ac.mu/prospectivestudents>

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exessive bounces issue
by Tihomir Culjaga 14 Mar '18

14 Mar '18

Hi all, this is the only list that keeps complaining about "excessive bounces" I'm subscribed to .. currently I'm subscribed to 7 lists with no bounce issues whatsoever. Im using gmail and i cannot see if any bounces are really happening... so i mailed reeswitch-users-owner(a)lists.freeswitch.org <freeswitch-users-owner(a)lists.freeswitch.org> to come to the root of the problem but seems the mail went to /dev/null So, does anyone experiences a similar problem with freeswitch-users list ? Regards, Tihomir.

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Setting a c= line underneath audio m= SDP
by Rob Capo 14 Mar '18

14 Mar '18

Hi, I'm using an MRCPv2 server for TTS in a particular call environment. I have written the MRCPv2 client in a Go application that is talking to FreeSWITCH via ESL. I've set this application to directly send the audio to remote_media_ip:remote_media_port. However, some VoIP clients seem to filter the UDP data because the c= line specifies the address of the FS server instead of the MRCP server. SDP supports adding a c= line in a media section if it's different than the c= line at the top of the SDP. I'm wondering if FreeSWITCH supports doing this via setting some variable or executing some command. Thanks, Rob

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start_dtmf false positives
by Roman Dissauer 14 Mar '18

14 Mar '18

Hi Guys, we have troubles reliably recognizing DTMF tones by using dialplan application start_dtmf. From our carriers we get some calls with inband DTMF and some with RFC2833. To our customers we need to always do RFC2833. Therefore we check SDP information and enable start_dtmf when RFC2833 is not available on the incoming call. When start_dtmf is executed we have a lot of false positives mostly at the start of audio but also during the call. Is there a way to prevent false DTMF detection? Does anyone have same issues and a solution for it? Kind regards Roman Dissauer RDI SOLUTIONS e.U. Rosasgasse 13/24 1120 Wien, Österreich T: +43 1 3530349 - 10 F: +43 1 3530349 - 99 roman.dissauer(a)rdi.at www.rdi.at

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How to bridge a call to an extension without the display updating?
by Michael Avers 13 Mar '18

13 Mar '18

Hello, I'm trying to create a random dialplan extension (say 5050) that when dialed into by a registered extension would bridge the call to another registered user (say 3000), but not change the display or P-Asserted-Identity. Right now on the bridge it sends to the calling extension *P-Asserted-Identity: "5050" <sip:3000@sip.domain>.* I tried setting all the possible values for sip_cid_type and setting ignore_display_updates to false, but it still sends the P-Asserted update as shown above. The bridge itself I tried both with user/3000(a)sip.domain and ${sofia_contact(566055(a)rsi.solidphone.net)} - same result. This is a problem because A. the display on the phone changes to "3000" and more importantly, B. if the calling user later hits redial it will dial 3000 instead of 5050. Any way to achieve what I'm trying to do? Thanks Mike

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MUX mode, don't want to see my own video
by Dom Rumsey 11 Mar '18

11 Mar '18

Hi Guys, I have compiled FS using the master branch (on a Debian 8 Jessie) and I'm using MUX mode for conferences with layout set to 1x1, so whoever speaks is seen full screen. Can anyone tell me how I can get it so I don't see my own video feed when I talk? In a multiway conversation, it would be useful for me to see the video feed of the last person that spoke while everyone else sees my video feed. I wasn't sure if this was easily achievable with the correct settings. Thanks in advance for any help. Dom

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javascript get session extension number
by Francesco Facco de Lagarda 10 Mar '18

10 Mar '18

Goodmorning to all. Can anyone please tell me how I can get the extension number that originated the session in javascript?

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Javascript
by Francesco Facco de Lagarda 10 Mar '18

10 Mar '18

Good morning all.. Could someone please tell me how to initiate a call recording in javascript? Simplyfied code I have: if (session.ready()) { var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060") if (sessOut.ready()) { bridge(session, sessOut); } } Any help appreciated.. have a happy and productive day and Happy Women's day for any ladies there! Francesco

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