- FreeSWITCH-users - FreeSWITCH Mailing Lists

VM not deleted with vm-keep-local-after-email=false
by Sean Devoy 30 Oct '19

30 Oct '19

I have this config in the directory: <param name="vm-mailto" value="sdevoy(a)bizfocused.com"/> <param name="vm-keep-local-after-email" value="false"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true" /> It is working great now except the VM is not deleted regardless of vm-keep-local-after-email. Any ideas? Sean

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mod_voicemail send email --- lost
by Sean Devoy 30 Oct '19

30 Oct '19

Hi, I recently upgraded to v 1.10 on Debian 10. My log shows emails being mailed just fine. However, I cannot find them on the recipient end. I have searched all folders and spam. Any help would be greatly appreciated. I tested with this first: sendmail -vt < ./mailtest.txt mailtest.txt is: To: sdevoy(a)bizfocused.com Subject: FreeSwitch sendmail test two From: freeswitch(a)bizfocused.com And here goes the e-mail body, test test test.. That works fine. But FreeSwitch emails do not. Switch.config.xml has: <param name="mailer-app" value="/usr/sbin/sendmail"/> <param name="mailer-app-args" value="-t"/> <param name="dump-cores" value="yes"/> The extension params for this extension are: <param name="vm-mailto" value="sdevoy(a)bizfocused.com"/> <param name="vm-mailfrom" value="voicemail(a)bizfocused.com"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true" /> The FS log says: 2019-10-29 20:41:28.276313 [DEBUG] mod_voicemail.c:2832 Deliver VM to 220@voicemail_BFIS 2019-10-29 20:41:28.276313 [DEBUG] mod_voicemail.c:1925 Update MWI: Processing for 220@voicemail_BFIS in inbox 2019-10-29 20:41:28.276313 [DEBUG] mod_voicemail.c:1950 Update MWI: Messages Waiting yes 2019-10-29 20:41:28.276313 [DEBUG] mod_voicemail.c:1951 Update MWI: Update Reason NEW 2019-10-29 20:41:28.276313 [DEBUG] mod_voicemail.c:1952 Update MWI: Message Account 220@voicemail_BFIS 2019-10-29 20:41:28.276313 [DEBUG] mod_voicemail.c:1953 Update MWI: Voice Message 1/0 2019-10-29 20:41:28.876348 [DEBUG] switch_utils.c:1298 Emailed file [/tmp/mail.157238168887fe] to [sdevoy(a)bizfocused.com] 2019-10-29 20:41:28.876348 [DEBUG] mod_voicemail.c:3042 Sending message to sdevoy(a)bizfocused.com 0735ca19-76d1-460e-9571-ef499f962cd1 2019-10-29 20:41:28.876348 [DEBUG] switch_core_session.c:2905 sofia/internal/14108084699(a)64.136.173.31 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 0735ca19-76d1-460e-9571-ef499f962cd1 2019-10-29 20:41:28.876348 [DEBUG] switch_core_state_machine.c:651 (sofia/internal/14108084699(a)64.136.173.31) State EXECUTE going to sleep 0735ca19-76d1-460e-9571-ef499f962cd1 2019-10-29 20:41:28.876348 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/14108084699(a)64.136.173.31) Running State Change CS_HANGUP (Cur 2 Tot 327) 0735ca19-76d1-460e-9571-ef499f962cd1 2019-10-29 20:41:28.876348 [DEBUG] switch_core_state_machine.c:848 (sofia/internal/14108084699(a)64.136.173.31) Callstate Change ACTIVE -> HANGUP 0735ca19-76d1-460e-9571-ef499f962cd1 2019-10-29 20:41:28.876348 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/14108084699(a)64.136.173.31) State HANGUP Mail.log for this time period: Oct 29 20:46:46 fs-bfis2 sendmail[15253]: x9TKkkOP015253: from=root, size=153, class=0, nrcpts=1, msgid=<201910292046.x9TKkkOP015253(a)fs-bfis2.bizfocused.com>, relay=root@localhost Oct 29 20:46:46 fs-bfis2 sm-mta[15261]: x9TKkkDb015261: from=<root(a)fs-bfis2.bizfocused.com>, size=389, class=0, nrcpts=1, msgid=<201910292046.x9TKkkOP015253(a)fs-bfis2.bizfocused.com>, proto=ESMTP, daemon=MTA-v4, relay=localhost [127.0.0.1] Oct 29 20:46:46 fs-bfis2 sm-mta[15261]: STARTTLS=client, relay=bizfocused-com.mail.protection.outlook.com., version=TLSv1.2, verify=FAIL, cipher=ECDHE-RSA-AES256-SHA384, bits=256/256 Oct 29 20:46:48 fs-bfis2 sm-mta[15261]: x9TKkkDb015261: to=<sdevoy(a)bizfocused.com>, ctladdr=<root(a)fs-bfis2.bizfocused.com> (0/0), delay=00:00:02, xdelay=00:00:02, mailer=esmtp, pri=30389, relay=bizfocused-com.mai...ction.outlook.com. [104.47.37.36], dsn=2.0.0, stat=Sent (<201910292046.x9TKkkOP015253(a)fs-bfis2.bizfocused.com> [InternalId=15448997367314, Hostname=BN6PR01MB2611.prod.exchangelabs.com] 8530 bytes in 0.197, 42.133 KB/sec Queued mail for delivery) Oct 29 20:46:48 fs-bfis2 sendmail[15253]: x9TKkkOP015253: to=sdevoy(a)bizfocused.com, ctladdr=root (0/0), delay=00:00:02, xdelay=00:00:02, mailer=relay, pri=30153, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (x9TKkkDb015261 Message accepted for delivery) Oct 29 20:47:31 fs-bfis2 sendmail[15317]: x9TKlVic015317: from=root, size=153, class=0, nrcpts=1, msgid=<201910292047.x9TKlVic015317(a)fs-bfis2.bizfocused.com>, relay=root@localhost Oct 29 20:47:31 fs-bfis2 sm-mta[15320]: x9TKlVxv015320: from=<root(a)fs-bfis2.bizfocused.com>, size=389, class=0, nrcpts=1, msgid=<201910292047.x9TKlVic015317(a)fs-bfis2.bizfocused.com>, proto=ESMTP, daemon=MTA-v4, relay=localhost [127.0.0.1] Oct 29 20:47:31 fs-bfis2 sm-mta[15320]: STARTTLS=client, relay=bizfocused-com.mail.protection.outlook.com., version=TLSv1.2, verify=FAIL, cipher=ECDHE-RSA-AES256-SHA384, bits=256/256 Oct 29 20:47:32 fs-bfis2 sm-mta[15320]: x9TKlVxv015320: to=<sdevoy(a)bizfocused.com>, ctladdr=<root(a)fs-bfis2.bizfocused.com> (0/0), delay=00:00:01, xdelay=00:00:01, mailer=esmtp, pri=30389, relay=bizfocused-com.mai...ction.outlook.com. [104.47.38.36], dsn=2.0.0, stat=Sent (<201910292047.x9TKlVic015317(a)fs-bfis2.bizfocused.com> [InternalId=14349485741504, Hostname=MWHPR01MB2414.prod.exchangelabs.com] 8583 bytes in 0.193, 43.379 KB/sec Queued mail for delivery) Oct 29 20:47:32 fs-bfis2 sendmail[15317]: x9TKlVic015317: to=sdevoy(a)bizfocused.com, ctladdr=root (0/0), delay=00:00:01, xdelay=00:00:01, mailer=relay, pri=30153, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (x9TKlVxv015320 Message accepted for delivery) Oct 29 20:48:08 fs-bfis2 sendmail[15323]: gethostbyaddr(165.227.222.107) failed: 1 Oct 29 20:48:08 fs-bfis2 sendmail[15323]: gethostbyaddr(10.17.0.8) failed: 1 Mail.warn: Oct 29 20:48:08 fs-bfis2 sendmail[15323]: gethostbyaddr(165.227.222.107) failed: 1 Oct 29 20:48:08 fs-bfis2 sendmail[15323]: gethostbyaddr(10.17.0.8) failed: 1 Any ideas? Thanks, Sean

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Re: [Freeswitch-users] replace SIP from user and host part
by David Villasmil 30 Oct '19

30 Oct '19

If you need to set a sip-from to be the same always, you can set that in the gateway configuration: <param name="from-user" value="cluecon"/> On Tue, 29 Oct 2019 at 15:56, Tamer Higazi via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi <th982a(a)googlemail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Sun, 27 Oct 2019 23:36:54 +0100 > Subject: replace SIP from user and host part > 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 > > > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Tue, 29 Oct 2019 08:56:21 -0700 (PDT) > Subject: [Freeswitch-users] replace SIP from user and host part > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

Use the From instead of P-Asserted-Identity
by igor.potjevlesch＠gmail.com 30 Oct '19

30 Oct '19

Hello, I have a call scenario with From + P-Asserted-Identity. I would like to give precedence on From. I configured the caller-id-in-from to True for this gateway and force on leg B sip_cid_type to none in the dialplan. I have tried some others combinations, but each time the PAI is inserted into the From header on the leg B. Any idea of the right way to do this? Regards, Igor.

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CHANNEL_HANGUP_COMPLETE events received not keeping pace with number of calls
by Neil Patel 30 Oct '19

30 Oct '19

Hi All, I have an event listener script (Python) that listens for “CHANNEL_HANGUP_COMPLETE” events. Logs show that the script, which runs continuously in an infinite loop within supervisor, receives 1000 events every 15 mins. At the same time we are making about 15K calls every 15 mins. So this script is 15x slower than the volume of calls and falls behind and often gets the “Queue size: [100000/100000] Max queue size reached” error. My question is: how do I make the events reach the listener faster to keep up with the outbound call traffic? Thanks.

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About FreeSWITCH HA
by 王聡 30 Oct '19

30 Oct '19

Hey all, I’m trying to build a FS cluster with HA, but I got some trouble on recovering calls. HA system is based on HA instruction(https://freeswitch.org/confluence/display/FREESWITCH/High+Avail… <https://freeswitch.org/confluence/display/FREESWITCH/High+Availability>), and my freeswitch dialplan is based on conference app. When I tried to recover a call on secondary server, a new coming call arised to callee and the old call turned into mute. The same setting works well based on bridge, which correctly failover to secondary server. The dialplan showed like: <extension name="apptest"> <condition field="destination_number" expression="apptest1002"> <action application="answer"/> <action application="set" data="conference_auto_outcall_prefix={execute_on_answer=lua welcome.lua 1001-1002 hasid apptest}"/> <action application="conference_set_auto_outcall" data=“user/9991002"/> <action application="conference" data="1001-1002@default”/> </condition> </extension> I wonder if this problem occurred by conference or mistaken configure. Any idea would be appreciated. Regards, C.Wang

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Voicemail to email with user XML from Lua
by Paul Smith 30 Oct '19

30 Oct '19

Hi, I'm new to freeswitch but have been playing with configs and reading confluence and the Packt book for a few weeks. Is there a difference between setting up a user as static XML and getting the user XML snippet from a lua script? I can't seem to get voicemail to email to work if the user XML snippet is generated by lua, but it works fine if I have the static xml in a file. The following fs_cli command to post the email works fine. So the mail server, templates etc are OK. vm_fsdb_msg_email default 123 1039 f8df8b69-21d7-4339-836e-777dbec87db3 customer1@mytestdomain I suspect that loading the XML at startup as opposed to generating snippet from lua is significant? This static user directory/123/1039.xml works: <include> <user id="1039"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1000"/> <param name="vm-mailto" value="testing(a)mydomain-ni.com" /> <param name="vm-mailfrom" value="paul(a)mydomain-ni.com" /> <param name="vm-email-all-messages" value="true" /> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1039"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> Email is sent and log shows: *2019-10-24 09:38:21.955085 [DEBUG] mod_voicemail.c:1952 Update MWI: Message Account 1039@123 2019-10-24 09:38:21.955085 [DEBUG] mod_voicemail.c:1953 Update MWI: Voice Message 28/1 2019-10-24 09:38:22.875122 [DEBUG] switch_utils.c:1300 Emailed data to [testing(a)mydomain-ni.com] 2019-10-24 09:38:22.875122 [DEBUG] mod_voicemail.c:3042 Sending message to testing(a)mydomain-ni.com e5e12f00-e39b-4eda-8e27-a22adc14ddb3 2019-10-24 09:38:22.875122 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] * But generating a very similar XML from LUA... the voicemail is saved, and MWI activated. But no email is sent. 2019-10-24 10:03:18.271420 [DEBUG] mod_voicemail.c:1925 Update MWI: Processing for 1039@123 in inbox 2019-10-24 10:03:18.271420 [NOTICE] switch_cpp.cpp:1443 Debug directory xml snippet: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory"> <domain name="123"> <user id="1039"> <params> <param name="password" value="sjkdfsjhdf4895sdkjh8934tsdvsjdkgsf"/> <param name="vm-password" value="0000"/> <param name="vm-mailto" value="checkto(a)mydomain-ni.com"/> <param name="vm-mailfrom" value="voicemail(a)vociemail.co.uk"/> <param name="vm-email-all-messages value="true"/>   </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="user_context" value="default"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1039"/> <variable name="outbound_caller_id_name" value="Frank"/> <variable name="outbound_caller_id_number" value="10390"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </domain> </section> </document> 2019-10-24 10:03:18.271420 [DEBUG] mod_voicemail.c:1950 Update MWI: Messages Waiting yes 2019-10-24 10:03:18.271420 [DEBUG] mod_voicemail.c:1951 Update MWI: Update Reason NEW 2019-10-24 10:03:18.271420 [DEBUG] mod_voicemail.c:1952 Update MWI: Message Account 1039@123 *2019-10-24 10:03:18.271420 [DEBUG] mod_voicemail.c:1953 Update MWI: Voice Message 29/1* *2019-10-24 10:03:18.271420 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en]* 2019-10-24 10:03:18.291420 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-goodbye.wav] (en:en) 2019-10-24 10:03:18.291420 [DEBUG] switch_ivr_play_say.c:1492 Codec Activated L16@8000hz 1 channels 20ms

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BLF missing NOTIFY Error?
by Sean Devoy 29 Oct '19

29 Oct '19

Hi, I have been working with the engineers at Fanvil (China) on a problem. Their English is just slightly better than my Mandarin. Their diagnosis is that this is a FreeSwitch error (of course). Here is the scenario: Ext 220 (Fanvil) has a BLF button/lamp for ext 225 (any sip phone). Things work as expected for incoming call to 225 (blink), 225 on call (solid), 225 hang up (off). The single item that fails is when 225 calls 220. When the call end 220's BLA for 225 stays on solid. Note that the other direction works fine (220 to 225). Fanvil says there should be a notify after the hangup when 225 calls 220 (monitoring 225). I have verified with wireshark that the is not one. I don't know if there should be. Would someone more knowledgeable than I care to weigh in? Do I need to learn how to do an actual JIRA? WireShark files attached. Let me know if there is another way to share them. "worked.dmp" is 220=>225 "error" is 225=>220 Thanks, Sean

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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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