September 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Embedding Freeswitch
by Naveen Khanna 23 Nov '17

23 Nov '17

Hi, Can someone suggest a low cost standard single board computer, around $ 50, to run 25 concurrent sessions & 100 registrations of SIP clients with limited applications of Freeswitch. Regards, Naveen Khanna

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Options for initiating call from browser (not WebRTC)
by Oivvio Polite 14 Nov '17

14 Nov '17

I have a simple call recording solution set up with a Twilio-like PaaS. The user interacts with a web app where she enters a destination number. My web app then instructs the PaaS to 1. Call the users phone 2. Wait for the user to answer 3. Start a recording. 4. Connect the destination number. The resulting wav is available for the user to download in the web app. I now want to replace the PaaS with FreeSwitch. I've done a lot of experimenting with FS during the last year, and managed to get callrecording of PSTN and WebRTC calls to work. With those calls origination was from PSTN or WebRTC. Now I want to do origination with a HTTP request. When leafing through the FreeSwitch book from PACKT I see that I have multiple options for doing this. 1. fs_cli Have the browser talk to a separate web app (written in node or python or whatever) that the initiates the call with fs_cli. 2. mod_httapi Have the browser talk to a webapp that talks to FS via mod_httapi. In this scenario I can't find any info on how to originate calls in mod_httapi though. 3. mod_event_socket Have a the browser talk to a webapp that talks to FS via mod_event_socket. In all of these scenarios I figure that I'll have the webapp running on the same box as FS and that I'll do all authentication/authorization of the user in the webapp. But which option should I go with? regards Oivvio

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Temporary Equivalent of fs_path
by Colin Morelli 06 Nov '17

06 Nov '17

Hey all, Trying to figure out how to get the equivalent behavior of fs_path, but only for a single transaction. In other words, I want to start a SIP request hitting a particular proxy, and then simply let Record-Route headers determine where the call should be routed after that. I might be completely overthinking this, but everything I've tried so far (sip_route_uri, fs_path), have resulted in Freeswitch continuing to use the given route for all subsequent requests, rather than simply falling back to the session route. Am I missing something? Best, Colin

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FreeSWITCH 1.8
by Oleg Stolyar 06 Oct '17

06 Oct '17

Hi guys, sorry if I missed an announcement, but has version 1.8 been officially released?

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Cisco 8845
by Andrew Colin 05 Oct '17

05 Oct '17

Hi Guys Has anyone got a cisco 8845 IP phone working on freeswitch?

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Bug - LUA session:recordFile when using video phones
by Shaun Stokes 02 Oct '17

02 Oct '17

Hi All, FreeSWITCH version 1.6.19 We are using LUA “session:recordFile” to record greetings which works perfectly on non-video devices, however video phones (such as Polycom VVX500\VVX600, Yealink T49G etc) are not supported unless video is disabled on the device thus preventing it from attempting to negotiate video in the SDP message. We have H264 enabled in FS gobal_codec_prefs, disabling H264 also resolves the problem. From the point we begin the recording there is a 10 second delay before the recording starts on video devices, recording starts after the error “Unable to establish inbound video stream” occurs. 2017-09-28 13:01:39.885386 [NOTICE] switch_core_io.c:1202 Activating write resampler 2017-09-28 13:01:39.985376 [DEBUG] switch_cpp.cpp:895 getDigits dtmf_buf: 2017-09-28 13:01:49.985371 [ERR] switch_ivr_play_say.c:496 Unable to establish inbound video stream 2017-09-28 13:01:49.985371 [DEBUG] switch_ivr_play_say.c:560 Raw Codec Activated, ready to waste resources! 2017-09-28 13:01:49.985371 [DEBUG] switch_ivr_play_say.c:674 Raw Codec Activated We’ve tried to work around the issue by adding this to our dialplan before the call is answered but the same issue occurs: <action application="export" data=" absolute_codec_string=PCMA,PCMU"/> <action application="export" data="codec_string=PCMA,PCMU"/> Looks like this is a known issue: https://groups.google.com/forum/#!topic/2600hz-users/61XyiXya5Ik https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9632/FS-9632.h… The JIRA “FS-9632” was marked as resolved but doesn’t appear to apply to LUA when using “session:recordFile”. https://freeswitch.org/jira/browse/FS-9632 Perhaps we’ve missed something, does anyone have any ideas? Thanks, Shaun ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

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How to overwrite Q850 errors code in freeswitch?
by Raviram Chandran 01 Oct '17

01 Oct '17

Since we have a errors code in PSTN call (IP -PSTN call) which we wanton our telephony server to pass those errors code directly to mobile apps. When subscriber's balance is lower than some predefined value during call setup, Warning header with SIP 180/183 response for low balance alert goes to calling Party (mobile client) and the call continues. If heart-beat with calling dialog fails or maximum time limit for the call is reached, Warning header with BYE message goes to calling party (mobile client) and the call is dropped. Other than this, following failure scenarios may hit during a call 1. Authentication failed 2. Account temporarily deactivated 3. Zero or inadequate balance while making call 4. Account already in use by configured number of users 5. Rate not defined 6. Called destination blacklisted 7. Any other application failure For all such failure scenarios, IN will send 408 SIP responses to client. This response will contain a Warning header specifying cause of failure and appropriate text message (configurable at IN). Format of Warning Header is as follows Warning: Cause_Code APP_Name "Error Text", e.g., Presently error codes are configured at network provider end which we need to pass from our telephony server to mobile apps

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cli command to list profiles
by Bipin Patel 28 Sep '17

28 Sep '17

hi, is there any cli command to list names of all sip profiles as at times clients use different names other than internal and external -- Regards, Bipin ------------------------------------------------------------------------

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How can I use an external database to create new users?
by Jose David Jurado Alonso 27 Sep '17

27 Sep '17

Hi, I'm thinking to create a web application to create, edit and delete users in a simpler way than creating XML by hand but I don't find information about it. I've seen something about the module "mod_xml_curl" but I'm not sure how it interacts with the database nor if it's necessary some additional module more. I would like to use MySQL or postgreSQL, does anyone know how you can do it or some tutorial or reference application? Thanks, José D.

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Only call the last registered account when the same account is on multiple devices
by Jose David Jurado Alonso 27 Sep '17

27 Sep '17

Hi, I have the same SIP account (1008) registered from several mobile SIP clients and when I call from another number (1000) the call only arrive to one of the SIP clients and not to all. How can I make the call to all SIP clients? I need this: The call came to all clients and when one response (pick up the call), it was cut into the other clients.

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