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notes about amazon AWS?
by Денис 02 Feb '16

02 Feb '16

Hello, who can share AMI for freeswitch and maybe some notes about setup (best practices)? But need centos 6 x86-64, it's _requirement_ some other our software. PV or HWM mode? https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 - too few info about centos.

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T.38 US providers
by Diego Rivera 01 Feb '16

01 Feb '16

Hello all, We use Freeswitch as our main fax server with a T38 trunk from Babytel, but now we're looking for alternatives on other reliable T38 providers in the US... mostly to see if we can get a comparable success rate at a better price. We typically fax in and out about 250k pages a week... What other good T.38 trunk providers do you guys know/recommend? Thanks, Diego

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Simple LuaScript to Record Session Failing
by Leonardo Lima Ribeiro 01 Feb '16

01 Feb '16

Hello all, I’m trying to record an IVR using my gateway to do the outbound call in my luascript: local new_session = freeswitch.Session("[origination_caller_id_name=‘987654321',origination_caller_id_number=‘987654321']sofia/gateway/MY_GATEWAY/3157244022", session); new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav") new_session:sleep("10000") new_session:hangup() So in the above script I just call to the Bank Of America as an example and try to record the first 10 seconds of the call in the recordings path. The problem is that I have an empty recording file.. Why? The funny thing is: if I add this command after the record_session command: new_session:streamFile("voicemail/vm-goodbye.wav”); And then this is the entire new script: local new_session = freeswitch.Session("[origination_caller_id_name=‘987654321',origination_caller_id_number=‘987654321']sofia/gateway/MY_GATEWAY/3157244022", session); new_session:execute("record_session","/usr/local/freeswitch/recordings/myrecording.wav”) new_session:streamFile("voicemail/vm-goodbye.wav”); new_session:sleep("10000") new_session:hangup() I can hear the Good Bye sound from my script and then hear the Bank of America IVR. I just don’t understand why the record works if I play a sound in our side and the record does not work if I don’t play any sound. Do you know what’s happening? How can I solve this? Thank you!

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Sending Data in SIP BYE message
by sandeep goje 01 Feb '16

01 Feb '16

Hi, I have a issue while setting the sip_bye_h_ headers. Here is the scenario A-->B A is a sip end point and B is a PSTN call placed through the dialplan. In this scenario, when B hangs up the call, I get proper Bye message sent to sip_bye_h headers set in the Bye message.But when A hangs up, the sip_bye_h headers are not sent in the 200 OK message. Is there a way to send the sip_bye_h headers in both BYE and 200 OK messages. I am trying to send bridge_channel or channel_name from B to A. Regards, Sandeep The most profound statements are often said in silence. -Lynn Johnston

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[Video Calling Issue][Freeswitch 1.7]
by Bilal Abbasi 01 Feb '16

01 Feb '16

Hi Users, I am writing this question after reading old questions and forums, so my problem is i am unable to dial video call using freeswitch 1.7. I have enabled video codecs in var.xml (VP8,H263,H264) and i am using bria sipfone for UA1 and UA2. I am able to dial the normal call but unable to dial video. do i need any further variables to activate video calling? One of my friend told me to shift to FS 1.6 as that provides transcoding. so can anybody help me out. FS Version: 1.7.0 OS: jessie 8 Softphone: Bria Regards Abbasi

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DTMF capture and early media
by Борисов, Дмитрий / Dmitriy Borisov 30 Jan '16

30 Jan '16

Hi, All! https://tools.ietf.org/html/rfc3960 has next part: Early media refers to media (e.g., audio and video) that is exchanged before a particular session is accepted by the called user. Within a dialog, early media occurs from the moment the initial INVITE is sent until the User Agent Server (UAS) generates a final response. It may be unidirectional or bidirectional, and can be generated by the caller, the callee, or both. Typical examples of early media generated by the callee are ringing tone and announcements (e.g., queuing status). Early media generated by the caller typically consists of voice commands or dual tone multi-frequency (DTMF) tones to drive interactive voice response (IVR) systems. But I know that in FreeSWITCH in preanswer state I can`t capture DTMF. Why? This part of RFC is conflicting with some other standard? It can be very intresting feature. -- С уважением, Борисов Дмитрий Отдел облачных технологий 1С-Рарус Тел. +7 (495) 231-20-02 доб.: 15-94 -- with best regards, Dmitriy Borisov 1C-Rarus Cloud Services tel.: +7 (495) 231-20-02 add.: 15-94

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Audio Codecs over Mobile Connection
by Oleg Stolyar 30 Jan '16

30 Jan '16

Hi guys, does anyone have experience using different codecs to reduce choppiness and metallic/robotic sound when connecting to FreeSWITCH from a mobile device? I know both OPUS and G.722 have packet loss concealment but I wonder if someone actually has practical experience with different codecs over iffy mobile connections. OPUS vs. G.722 vs G.711

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Routing and multicast question
by Chris Mandra 30 Jan '16

30 Jan '16

Hi guys - I have a question: I want an endpoint sending an audio RTP stream to freeswitch, which freeswitch is transcoding and then sending the transcoded RTP stream to another device on my network from which I then want to send back INTO freeswitch before sending to the 2nd endpoint. I’d like the stream from freeswitch to the other device on my network to be a multicast RTP stream. What’s the best way to do this? thanks! chris

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sips uri originate
by Sergey Safarov 30 Jan '16

30 Jan '16

Please tell me how to make a call with sips uri I has tried execute originate like "originate sofia/internal/ sips:2345@example.com;transport=tls &park()" but INVITE has "sip" uri whatever. I has looked sofia_glue_do_invite function on src/mod/endpoints/mod_sofia/sofia_glue.c and try several cases but no luck. Please suggest me correct way to use sips in INVITE, From, To, Contact headers. P.S. I want avoid use manually create sip_invite_to_uri, sip_invite_from_uri and similar variables.

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Resetting FS call counters
by Kurtis Heimerl 29 Jan '16

29 Jan '16

Hello All, I'd like to gather statistics about call failures. FS already does this for gateways: freeswitch@internal> sofia status external Profile::Gateway-Name Data State Ping Time IB Calls(F/T) OB Calls(F/T) ================================================================================================= NAME::NAME sip:FreeSWITCH@IP_ADDRESS NOREG 0.00 0/0 290/453 ================================================================================================= 1 gateway: *Inbound(Failed/Total): 0/0,Outbound(Failed/Total):290/453* However, this statistic is already aggregated and I'd like it in time series. Is there a way to reset it to zero after a lookup in order to figure out how many calls have happened since the last query?

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