October 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Write wav files at 48000hz and I need 8000hz
by Jose Figueroa 28 Oct '19

28 Oct '19

Hello, I am working with FS 1.8 and everything goes well enough so far. We record all the inbound and outbound calls. The inbound calls are recorded as wav 8000hz sample rate. But the calls outbound are recorded as 48000hz sample rate as wav files, the question is, is it a config to set the default or force the recording to be saved as 8000hz always? I use verto and sip clients, the telephone-event and the sample from codecs is 8000hz so I dont understand how to achieve this. Any help would be appreciate it. Thanks JNF

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Fwd: [CFP] FOSDEM 2020, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 28 Oct '19

28 Oct '19

FOSDEM - Real Time Communications devroom CfP ============================================= Overview -------- [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2020 takes place 1-2 February 2020 in Brussels, Belgium. This document contains information about: - Real-Time Communications developer room (devroom) and lounge - speaking opportunities - volunteering in the devroom and lounge - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities) - the Planet aggregation sites for RTC blogs **NEW:** Save yourself entering Free-RTC events and CFP deadlines into your calendar and task list, follow our iCalendar feed: https://freertc.org/events.ics Call for participation - Real Time Communications (RTC) ------------------------------------------------------- The Real-Time devroom and Real-Time lounge are about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. **We are looking for speakers for the devroom and volunteers and participants for the tables in the Real-Time lounge.** The devroom is only on Sunday, 2nd of February 2020. The lounge will be present for both days. To discuss the devroom and lounge, please join the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). ### Speaking opportunities Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications devroom: deadline 23:59 UTC on 15th of December. Please use the [Pentabarf](https://penta.fosdem.org/submission/FOSDEM20/) system to submit a talk proposal for the devroom. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". Other devrooms and lightning talks: some speakers may find their topic is in the scope of more than one devroom. It is encouraged to apply to more than one devroom and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. Here you can find the [full list of devrooms](https://www.fosdem.org/2020/schedule/tracks/) and here you can apply for a [lightning talk](https://fosdem.org/submit). ### First-time speaking? FOSDEM devrooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the devroom administrators personally if you would like to ask any questions about it. ### Submission guidelines The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - The purpose of your talk - Any other talk applications (devrooms, lightning talks, main track) - Availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and devroom themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the devroom administrators based on the number of received proposals. As the two previous devrooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ----------------- To make the devroom and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - Organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 1 February - Participation in the Real-Time lounge - Circulating this Call for Participation to other mailing lists Social events and dinners ------------------------- The traditional FOSDEM beer night occurs on Friday, 31st of January. On Saturday night, there are usually dinners associated with each of the devrooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss) for further details about the Saturday night dinner options and how you can register for a seat. Related events around FOSDEM ---------------------------- As per usual, the [XMPP Summit](https://wiki.xmpp.org/web/Conferences/Summit_24) is happening ahead of FOSDEM. This time it will take place on the 30th and 31st of January also in Brussels. All details are available on the [summit website](https://wiki.xmpp.org/web/Conferences/Summit_24). Spread the word and discuss --------------------------- If you know of any mailing lists where this CfP would be relevant, please forward this document. If this devroom excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: - All projects https://planet.freertc.org planet(a)freertc.org - XMPP https://planet.jabber.org ralphm(a)ik.nu - SIP https://planet.sip5060.net planet(a)sip5060.net - (Español) https://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ------- For any private queries, contact us directly using the address **fosdem-rtc-admin(a)freertc.org** and for any other queries please ask on the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). The devroom administration team: - Saúl Ibarra Corretgé <s(a)saghul.net> - Ralph Meijer <ralphm(a)ik.nu> - Daniel-Constantin Mierla <miconda(a)gmail.com> - Daniel Pocock <daniel(a)pocock.pro> - Guus der Kinderen <guus.der.kinderen(a)gmail.com>

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replace SIP from user and host part
by Tamer Higazi 27 Oct '19

27 Oct '19

Hi people, My ISP provider is very special and forces me to reset the user and host part of my external profile. At the current stage, FS sets the endpoint user ID as sip-from user, and the external ip-address as the sip-from host and the ISP gateway returns me a 404 not found. When I turn off the PBX and register to the gateway directly with a softphone, the softphone can place a call. sip-from user is the userID of the gateway login and host the proxy address of the gateway. How can I set a new sip-from-user and host prior bridging a call to the gateway? I found 2 channel variables. variable_sip_from_user - sip_from_user variable_sip_from_host - sip_from_host Is that the solution manipulating these variables? Am I capable manipulate those ? Would that be the solution ? I am a little bit new to FS. For a dialplan example I welcome thankfully any advise. wireshark dump: JITSI Softphone Call: Session Initiation Protocol (100) Status-Line: SIP/2.0 100 trying -- your call is important to us Status-Code: 100 [Resent Packet: False] [Request Frame: 2481] [Response Time (ms): 13] Message Header Call-ID: bb85334b160ce6cb26ab480fd2a13236@0:0:0:0:0:0:0:0 [Generated Call-ID: bb85334b160ce6cb26ab480fd2a13236@0:0:0:0:0:0:0:0] CSeq: 2 INVITE Sequence Number: 2 Method: INVITE From: "117xxxxxx" <sip:117xxxxxx@sip12.e-fon.ch>;tag=51a3e1d6 SIP Display info: "117xxxxxx" SIP from address: sip:117xxxxxx@sip12.e-fon.ch SIP from address User Part: 117xxxxxx SIP from address Host Part: sip12.e-fon.ch SIP from tag: 51a3e1d6 To: <sip:0764823425@sip12.e-fon.ch> SIP to address: sip:076xxxxxxx@sip12.e-fon.ch SIP to address User Part: 076xxxxxxx SIP to address Host Part: sip12.e-fon.ch Via: SIP/2.0/UDP 217.162.xxx.xxx:5060;branch=z9hG4bK-383638-321b4bfd0c24f1f48906ff6382c3b18b;rport=5060 Transport: UDP Sent-by Address: 217.162.xxx.xxx Sent-by port: 5060 Branch: z9hG4bK-383638-321b4bfd0c24f1f48906ff6382c3b18b RPort: 5060 Content-Length: 0 (successfull placing the call) FS Call: Session Initiation Protocol (404) Status-Line: SIP/2.0 404 Not found Status-Code: 404 [Resent Packet: False] [Request Frame: 12659] [Response Time (ms): 13] Message Header Via: SIP/2.0/UDP 217.162.xxx.xxx:5080;rport=5080;branch=z9hG4bKmNHFUDQav3tUj;received=217.162.xxx.xxx Transport: UDP Sent-by Address: 217.162.xxx.xxx Sent-by port: 5080 RPort: 5080 Branch: z9hG4bKmNHFUDQav3tUj Received: 217.162.xxx.xxx From: "1000" <sip:1000@217.162.xxx.xxx>;tag=gNe9eHQm44HBj SIP Display info: "1000" SIP from address: sip:1000@217.162.xxx.xxx SIP from address User Part: 1000 SIP from address Host Part: 217.162.xxx.xxx SIP from tag: gNe9eHQm44HBj To: <sip:076xxxxxxx@sip12.e-fon.ch>;tag=56dedfd3319cd1b9fedc1197b499329c.26ea SIP to address: sip:076xxxxxxx@sip12.e-fon.ch SIP to address User Part: 076xxxxxxx SIP to address Host Part: sip12.e-fon.ch SIP to tag: 56dedfd3319cd1b9fedc1197b499329c.26ea Call-ID: 84f02fe4-71ef-1238-b881-00012e808c2d [Generated Call-ID: 84f02fe4-71ef-1238-b881-00012e808c2d] CSeq: 11458424 INVITE Sequence Number: 11458424 Method: INVITE Content-Length: 0

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Re: [Freeswitch-users] Voicemail on AWS
by Per Modin 26 Oct '19

26 Oct '19

On 2019-09-23 20:10 (CEST), H Yavari via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > Date: Sun, 22 Sep 2019 09:32:56 +0000 (UTC) > From: H Yavari <hyavari(a)rocketmail.com> > To: "freeswitch-users(a)lists.freeswitch.org" > <freeswitch-users(a)lists.freeswitch.org> > Subject: Voicemail on AWS > X-Mailer: WebService/1.1.14303 YMailNorrin Mozilla/5.0 (Windows NT 10.0; > Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/77.0.3865.75 > Safari/537.36 > > I am looking for some hints how we can make an voicemail server for > our company users to have their own voicemail box, then we save all > voice files on AWS S3. I see mod_http_cache, but afraid of it's only > solution or there is better one. Hi, we used to run on S3, where we mounted the recordings with s3fs-fuse[1] which we ran successfully on 1.6.20. HOWEVER, when we updated to 1.8.5 we ran into problems with the wave files, the media stream was not saved and we ended up with empty media containers, with only some meta data of around 250 bytes ended up being saved. This issue affected about half of all recordings, and we found no correlation in regards to codecs used etc. If this was an issue with configuration, s3fs-fuse or Freeswitch we didn't investigate. We solved this by moving the storage from S3 to EFS, and are currently running it in production without issues in that regard. [1]: https://github.com/s3fs-fuse/s3fs-fuse -- Per Modin Software Engineer Working Group Two m: +46724452037 w: https://wgtwo.com e: per(a)wgtwo.com

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ICE/DTLS handshake
by Mirko Brankovic 25 Oct '19

25 Oct '19

HI, Has anyone experienced DTLS handshake takes 5s to get to SETUP state: > 2017-09-21 11:40:40.021178 [INFO] switch_rtp.c:3515 Changing audio DTLS > state from OFF to HANDSHAKE > 2017-09-21 11:40:45.319457 [INFO] switch_rtp.c:3172 Changing audio DTLS > state from HANDSHAKE to SETUP In network dump i see that answering side is not sending STUN for this 5s and then suddenly answers last 5 STUNs from A side. Has anyone encountered this kind of problem ? I have a pcap if necessary... Thanks, -- Regards, Mirko

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uuid_record to .raw crash
by Callum Guy 25 Oct '19

25 Oct '19

Easily reproducible issue, simply try and record to a file with the .raw extension. This causes FreeSWITCH to immediately crash: uuid_record 01386fbc-b552-487e-a5af-4bd1661a97f0 start /tmp/file.raw Using: FreeSWITCH Version 1.10.1-release-12-f9990221e6~64bit (-release-12-f9990221e6 64bit) OS: Debian 10 Buster (4.19.0-6-amd64) Apologies, no experience capturing a core dump and have failed to do so from the notes on https://freeswitch.org/confluence/display/FREESWITCH/Debugging, wanted to report the issue. -- *0333 332 0000 | www.x-on.co.uk <http://www.x-on.co.uk> | ** <https://www.linkedin.com/company/x-on> <https://www.facebook.com/XonTel> <https://twitter.com/xonuk> * X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments.

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Re: [Freeswitch-users] webrtc DTLS1.2 support in Freeswitch
by Mike Jerris 25 Oct '19

25 Oct '19

Newest release defaults to 1.2 as long as you use an openssl that supports it (note that this is not the case on some older distros but should be on at least buster). That was not the case in previous releases… You can check release notes to confirm version but I think that change was in 1.10.0. The commit note and release notes will have more details on this. > On Oct 22, 2019, at 7:55 AM, kaiduan xie via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > From: kaiduan xie <kaiduanx(a)yahoo.ca> > Subject: webrtc DTLS1.2 support in Freeswitch > Date: October 22, 2019 at 7:54:37 AM MDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Hi, > > Does Freeswitch support DTLS 1.2 for webrtc? If yes, which version supports DTLS 1.2? Google will drop DTLS1.0 support in M81 per > > https://bugs.chromium.org/p/webrtc/issues/detail?id=10261 <https://bugs.chromium.org/p/webrtc/issues/detail?id=10261> > > Thanks, > > /Kaiduan > >

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Date/time of call when listening to voicemail
by Peter Steinbach 25 Oct '19

25 Oct '19

Hello, when listening to voicemail, it only tells me the date of the recording but not date/time. Example * Message number 1 * 18:10 And I would like to hear * Message number 1 * Wed 23/10 at 18:10 How can I solve this? When I look into my german VM phases, I see for 'voicemail_say_date' <action function="say" data="$1" method="pronounced" type="current_date_time"/> So I expect it should tell me date and time. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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WS SETUP FAILED
by Mundkowsky, Robert 24 Oct '19

24 Oct '19

Hi, I am getting "WS SETUP FAILED" when trying to use Verto. Is there anyway to get more detailed information about what is wrong? Note the logs are already set to debug level. 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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where repo for FS 1.8 + Debian 9 ?
by Yuriy Nasida 24 Oct '19

24 Oct '19

Hi there, I know that currently we should use Freeswitch 1.10 + Debian 10 but I really need FS 1.8 + Debian 9. Looks like that repo is removed. Please advice if I am not correct and stil able to use it. Thanks

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