- FreeSWITCH-users - FreeSWITCH Mailing Lists

Tone segment f1 and f2
by Raimundo Pérez Nieves 23 Jan '18

23 Jan '18

Hi guys What is the meaning of this (specially f1 and f2): mod_spandsp_dsp.c:705 Tone segment: f1 = -1, f2 = -1, duration = 11232 mod_spandsp_dsp.c:705 Tone segment: f1 = 1, f2 = 3, duration = 48 mod_spandsp_dsp.c:705 Tone segment: f1 = -1, f2 = -1, duration = 944 mod_spandsp_dsp.c:705 Tone segment: f1 = 2, f2 = 3, duration = 32 mod_spandsp_dsp.c:705 Tone segment: f1 = 3, f2 = -1, duration = 32 mod_spandsp_dsp.c:705 Tone segment: f1 = -1, f2 = -1, duration = 944 mod_spandsp_dsp.c:705 Tone segment: f1 = 1, f2 = 2, duration = 64

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start_dtmf_generate not working with opus
by Varghese Paul 23 Jan '18

23 Jan '18

Hi all, I am facing a issue with start_dtmf_generate when i use opus codec and SIP INFO dtmf. Caller ( opus/dtmf ) ------------------------ Freeswitch ----------------------------- Calleee ( ulaw/inband) Caller is using jssip webrtc and dtmf is in INFO. For inband dtmf generation i am using start_dtmf_generate in caller channel. The setup is working when i use ulaw or ilbc or alaw codec. When i use webrtc with OPUS codec i am seeing the inband dtmf generation is not working. Webrtc with ulaw codec and dtmf INFO is working fine. Version : FreeSWITCH Version 1.6.17~64bit ( 64bit) I am attaching the opus.conf.xml for the reference. Any idea, why start_dtmf_generate is not working with opus codec ?. Do i need to set any setting for opus ?. Regards Varghese Paul

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SIP 482 "Request Merged" on parallel forking
by Carlos Ruiz Díaz 23 Jan '18

23 Jan '18

Hey guys, It seems like FS is unable to process a parallel forking originating in Kamailio (or others proxies of its type). Kamailio does the routing logic, and creates various branches all with different request URIs but keeping the same CSeq and Call-ID and changing the branch parameters, but FS interprets these requests as the same and merges them running the forking impossible. - I've seen replies that the CSeq should be increased, but this is not correct according to the RFC. - Change Call-IDs, which should not be the case for parallel forking. - And the only reported solution, change the T4 SIP timer to less than 4 seconds and put a sleep() on Kamailio for each branch. I personally don't consider this a solution as it plays with the timers. Do you have any recommendations or ideas to solve this? Thanks, -- Carlos

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Need use static variable every time execute DialPlan
by Raimundo Pérez Nieves 23 Jan '18

23 Jan '18

Hi, Iam having this situation: The leg A calls freeswitch and I execute a script in Lua and make an originate to leg B. I execute an outbound socket and listen a socket which my $uuidLegB = $info->getHeader("unique-id"); is legB uuid and inside this outbound scrip I detect tone and transfer into DialPlan. The problem is that I want to bridge leg A with B, but Iam trying to use uuid_bridge but I lose uuid Leg A, how can I get the uuid Leg A inside script outbound or how I can set an static variable inside DialPlan which can I put this uuid of Leg A. Because I test creating a variable inline or some tricks and every time DialPlan executes forget everything Thanks

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message test
by jungle Boogie 22 Jan '18

22 Jan '18

Hi All, This is a test to see if my account can post to the users list. I received an email stating I was unsubscribed. -- ------- inum: 883510009027723 sip: jungleboogie(a)sip2sip.info

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verto - different browsers support
by Mundkowsky, Robert 22 Jan '18

22 Jan '18

Do you support different web browsers for Verto? I am guessing looking at the code that newer versions of Chrome, Firefox, and Edge likely work. And Safari might work. Robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________

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Tone Frecuencies
by Raimundo Pérez Nieves 22 Jan '18

22 Jan '18

Hi guys, There is a way when we use detect tone to avoid the busy tone detection. Iam able to detect busy tone, but I want to succeed in contrary case. case. <action application="tone_detect" data="ring_busy 400,425,600x120,620+480 r +1000 transfer '11 XML default' 2"/>

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ICTCore , The communications development platform / framework based on freeswitch
by Tahir Almas 19 Jan '18

19 Jan '18

ICT Innovations pleased to announce release of ICTCore Version 0.8.0 , the unified communications CTI platform for web developers to develop their custom ICT based applications https://github.com/ictinnovations/ictcore/releases/tag/v0.8.0 Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT *

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FS & RTCP Packet Versioning?
by Sp Pho 19 Jan '18

19 Jan '18

Hello, I'm receiving console warnings that the switch is receiving unsupported RTCP packet version 3. However, a quick trace is showing that the packets are indeed in v2. Any ideas on why this is happening would be greatly appreciated. Below, also find console output and parts of the trace. Many thanks in advance. Cheers SP *FS Console:* 2018-01-19 10:42:12.220489 [INFO] switch_rtp.c:6737 Auto Changing audio RTCP port from A.B.C.D:29315 to 10.0.3.33:29897 2018-01-19 10:42:12.220489 [DEBUG] switch_rtp.c:2567 Setting RTCP remote addr to 10.0.3.33:29897 2 2018-01-19 10:42:12.220489 [WARNING] switch_rtp.c:6588 Received an unsupported RTCP packet version 3 *Wireshark:* 20895 49.775408 W.X.Y.Z 10.0.3.33 RTCP 154 Sender Report Source description Frame 20895: 154 bytes on wire (1232 bits), 154 bytes captured (1232 bits) Internet Protocol Version 4, Src: W.X.Y.Z, Dst: 10.0.3.33 User Datagram Protocol, Src Port: 29315, Dst Port: 29897 Real-time Transport Control Protocol (Sender Report) [Stream setup by SDP (frame 17312)] 10.. .... = Version: RFC 1889 Version (2) ..... Real-time Transport Control Protocol (Source description) [Stream setup by SDP (frame 17312)] ..... [RTCP frame length check: OK - 112 bytes]

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List problems, test for bounces
by Giovanni Maruzzelli 18 Jan '18

18 Jan '18

Hello, I received a mail from the list server I was suspended because of bounces, and I must reactivate the subscription. I done that. This is a test. Btw, this is a gmail account, how can it bounces? Other lists works well... -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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