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No RTP send after Sleep
by Jurijs Ivolga 28 Aug '18

28 Aug '18

Hi, I observer weird behavior and I'm not sure if this is a bug or not. Before playing announcement I'm running sleep command and if there is no RTP packets received from other side, Freeswitch never starts to play announcement and sending RTP to endpoint. Is it expected behavior or should I fill a bug report? I'm using latest 1.6 Freeswitch Thank you! With kind regards, Jurijs

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Re: [Freeswitch-users] how to enable transcoding
by Michael Jerris 27 Aug '18

27 Aug '18

We will always try to avoid transcoding the best we can, and will only transcode when required. There is no way to enable it, as its enabled by default. Why do you think its not working? Make sure you have the codec modules you require loaded, and that the codecs you want to support are in the allowed codecs. > On Aug 27, 2018, at 5:37 PM, robert mundkowsky via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > From: robert mundkowsky <rfmundkowsky(a)yahoo.com> > Subject: how to enable transcoding > Date: August 27, 2018 at 5:24:08 PM EDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Can I get instructions on how to enable video transcoding on FreeSWITCH 1.6? > I would recommend using 1.8 > I did not see this covered in the books. > > I tried the following: > > sip_profiles/internal.xml: > > <param name="disable-transcoding" value="false”/> This setting is default, you dont need to set it > <param name="inbound-late-negotiation" value="true"/> > <param name="inherit_codec" value="true"/> > > vars.xml: > > <action application="set" data="media_mix_inbound_outbound_codecs=true"/> > <action application="set" data="proxy_media=false”/> This setting is default, you dont need to set it > And I enabled some different video codecs. > > Also tried: "Mixing Media/Codecs on different legs (transcoding)" from Codec Negotiation - FreeSWITCH - Confluence <https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation> but these instructions seem odd, for example I think you would want "late-negotiation".

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how to enable transcoding
by robert mundkowsky 27 Aug '18

27 Aug '18

Can I get instructions on how to enable video transcoding on FreeSWITCH 1.6? I did not see this covered in the books. I tried the following: sip_profiles/internal.xml: <param name="disable-transcoding" value="false"/> <param name="inbound-late-negotiation" value="true"/><param name="inherit_codec" value="true"/> vars.xml: <action application="set" data="media_mix_inbound_outbound_codecs=true"/><action application="set" data="proxy_media=false"/> And I enabled some different video codecs. Also tried: "Mixing Media/Codecs on different legs (transcoding)" from Codec Negotiation - FreeSWITCH - Confluence but these instructions seem odd, for example I think you would want "late-negotiation".

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P-asserted ID in E164 format
by heather.graham＠yahoo.com 27 Aug '18

27 Aug '18

Hello, How can I get FS to send the P-Asserted-Identity back in a 183 Session Progress in E.164 format? Is there a setting in sip_profile? For example, we get an INVITE that contains a npdi;rn=. INVITE sip:5556667777;npdi;rn=8889990000@5.5.5.5;user=phone SIP/2.0 From: "FromUser"<sip:5553334444@2.2.2.2;user=phone>;tag=714539940-1535391478434- To: <sip:5556667777@1.1.1.1:5060;user=phone> Call-ID: 181910844198(a)6.6.6.6 CSeq: 1044076882 INVITE Contact: <sip:5553334444;tgrp=8000;trunk-contextBS@2.2.2.2:5060;transport=udp> P-Asserted-Identity: "FromUser"<sip:5553334444@2.2.2.2;user=phone> FS responds with ... SIP/2.0 183 Session Progress From: "FromUser"<sip:5553334444@2.2.2.2;user=phone>;tag=714539940-1535391478434- To: <sip: 5556667777@1.1.1.1:5060;user=phone>;tag=p4Q6eH4D5KcaD Call-ID: 181910844198(a)6.6.6.6 CSeq: 1044076882 INVITE Contact: <sip:5556667777@5.5.5.5:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.19~64bit P-Asserted-Identity: "5556667777;npdi;rn=8889990000" <sip:5556667777;npdi;rn=8889990000@1.1.1.1 Is there a way to get FS to respond with rn=+1<number>? P-Asserted-Identity: "5556667777;npdi;rn=+18889990000" <sip:5556667777;npdi;rn=+18889990000@1.1.1.1 Thanks,Heather

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Early Media and "Late Negotiation"
by Sebastian Kemper 27 Aug '18

27 Aug '18

Hello list, I have a bit of a transcoding issue when calling out. This is a tiny (literally) home PBX running FreeSWITCH 1.8.1. I have <param name="inbound-late-negotiation" value="true"/> in the SIP profiles and <action application="set" data="inherit_codec=true"/> in the dialplan. This works well when there is no early media, meaning the late negotiation results in the same codec on both legs. But when there is early media from the provider, then this usually results in transcoding. Example: 1. Phone offers this to FS: G722 PCMA G729 2. FS offers the following to provider: G722 PCMA 3. Provider sends 183 with PCMA 4. FS sends 183 with G722 to local phone (transcoding begins) 5. Provider sends 200 OK with PCMA 6. FS sends 200 OK with G722 to phone (transcoding continues) This is obviously a bit inefficient, because PCMA would've be fine on both legs :) I found some variables like ignore_early_media and bridge_early_media, but after a bit of reading up on them I don't think they're what I need. On the WWW somebody said inbound-codec-negotiation set to greedy should take care of this, but I tried it and didn't see a difference. On the Confluence Early Media page there is actually a paragraph "Early Media and Late Negotiation", but as of now it's just a stub. Anybody knows some dialplan magic to get what I'm after? Thanks! Seb

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SIP Internal y External Dialplan behavior
by Social Boh 27 Aug '18

27 Aug '18

Hello everyone, I'm doing some tests registering a Endpoint to Internal and External SIP profile. If I using internal profile the call go to Default Dialplan. If I comment out this line: <variable name="user_context" value="default"/> on the configuration file of Endpoint (ej: 1000.xml) the call go to Public Dialplan. If I register the Endpoint to External SIP Profiles (using port 5080) the call always to Public Dialplan although the line about dialplan (1000.xml) is not commented out Thank you Regards -- --- I'm SoCIaL, MayBe

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Incorrect ptime
by Andrew Colin 25 Aug '18

25 Aug '18

Hi guys One of our idiot providers sends a ptime of either 20, 30 or 60 on each call.This causes quality issues as customers are all using 20 I have tried autofix-timing but that makes it worse. Any way i can set it to force the provider to use only 20? Kind RegardsAndrew Colin Sent via android mail

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record_session pauses recording during hold by first endpoint
by Matthew Nichols 24 Aug '18

24 Aug '18

In our dialplan we have a section like this: <action application="export" data="recording_follow_transfer=true"/> <action application="record_session" data="$${recording_dir}/${uuid}.wav"/> The problem we have is if the original caller (over SIP) goes on hold, the recording stops (the file size remains the same) until the original caller goes off hold. Is there a setting that would prevent this behavior? We would like the recording to continue (possibly recording the hold music, though more importantly continuing to record the audio of the other endpoint(s)). record_waste_resources is set to true. We've also tried to export send_silence_when_idle=-1 though that doesn't work. Thanks, Matthew Nichols, WestTel International

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respond to received SIP INFOs with a customized 200-OK
by Elliot 24 Aug '18

24 Aug '18

We're looking for a way to modify content-body of 200 OK message(CSeq INFO), insert XML content into its body. Purpose: FreeSWITCH to handle INFO interactions based on RFC5707: Media Server Markup Language (MSML) https://tools.ietf.org/html/rfc5707 > INFO message sent by UAC carries XML content. > UAS(FreeSWITCH) tries to parse the XML content on INFO message. 200 OK message should contain XML body in order to notify UAC with the structured "response codes" and "failed reason descriptions" by the INFO message. Current behavior: FreeSWITCH reply back to all INFO messages with 200 OK automatically, without carrying any content body on it(content-length=0) nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:529 nua_signal() nua(0x7fcf0c009d70): sent signal r_respond nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua_stack.c:573 nua_stack_signal() nua(0x7fcf0c009d70): recv signal r_respond 200 OK nua_params.c:485 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0x7fcf0c00a600, ...) called tport.c:3257 tport_tsend() tport_tsend(0x7fcf0c004700) tpn = UDP/x.x.x.x:5060 tport.c:4046 tport_resolve() tport_resolve addrinfo = x.x.x.x:5060 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7fcf0c004700): not found by name UDP/x.x.x.x:5060 tport.c:3594 tport_vsend() tport_vsend(0x7fcf0c004700): 544 bytes of 544 to udp/x.x.x.x:5060 tport.c:3492 tport_send_msg() tport_vsend returned 544 nta.c:6791 incoming_reply() nta: sent 200 OK for INFO (2) 06:12:20.678673 IP (tos 0x0, ttl 64, id 22618, offset 0, flags [none], proto UDP (17), length 572) 192.168.89.162.sip > 192.168.89.45.sip: SIP, length: 544 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.89.45:5060;branch=z9hG4bK-d8754z-8b0c4e23b5df6b68-1---d8754z-;rport=5060 From: <sip:192.168.89.45:5060>;tag=098fbe06e806d500 To: <sip:192.168.89.162:5060>;tag=tgX42HcKaeSma Call-ID: ZTM5MzMwYTI3NzA1ZDkxOGQ2MGZlYjcyYzNlZGUzMjY.e806d500(a)165.254.89.45 CSeq: 2 INFO User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 Expected behavior: when receiving INFO, be able to customize its content body on 200-OK response message. either by setting channel variables or optionally treat this 200OK as FreeSWITCH event A sample 200 OK message: SIP/2.0 200 OK Call‐ID: 36C7E305@10.10.0.16:15062 CSeq: 324630703 INFO From: <sip:user@10.10.0.16:15062> ;tag=10.10.0.1615062+1+1090000+fa0329b5 To: <sip:msml@10.14.9.13:5060>;tag=10.14.9.135060+1+1060000+81d0b080 Via: SIP/2.0/UDP 10.10.0.16:15062 ;branch=z9hG4bK+fdebd5691fe0202fc7d685d1e1138955+10.10.0.1615062+1 Server: DC‐SIP/2.0 Supported: precondition Supported: timer Supported: 100rel Accept: application/sdp, text/*, application/msml+xml, application/moml+xml Content‐Length: 220 Content‐Type: application/msml+xml Contact: <sip:msml@10.14.9.13:5060> <?xml version="1.0" encoding="US‐ASCII"?> <msml version="1.1"> <result response="200"> <auditresult targetid="conf:conf04010105a"> <confconfig deletewhen="nocontrol" term="true"/> </auditresult> </result> </msml> -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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Voicemail MWI multiple mailboxes notify one user
by Mircea Botoca-Huh 23 Aug '18

23 Aug '18

Hello, I would like to know if there is any possibility to notify one user when messages are left in multiple mailboxes? I want to have just one registered user which should monitor multiple voicemail mailboxes. Thank you for your time. Best regards, Mircea

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