August 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] JIRA FS-9657 - Inbound HEP SIP capture for TLS transport
by Tom Parrott 29 Aug '18

29 Aug '18

Hi Sergey, Can you explain what you mean? Did you mean you would like RTCP stats added? Thanks Tom On 2018-08-29 13:54, freeswitch-users-request(a)lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. Re: JIRA FS-9657 - Inbound HEP SIP capture for TLS transport > (Sergey Safarov) > 2. Re: rtp_has_crypto (Social Boh) > 3. Attended Transfer to Call Center Queue when agent answers at > the same time the transfer is completed (Shaun Stokes) > 4. Re: Attended Transfer to Call Center Queue when agent answers > at the same time the transfer is completed (Shaun Stokes) > _______________________________________________ > 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 > http://www.freeswitch.org

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rtp_has_crypto
by Social Boh 29 Aug '18

29 Aug '18

Hello list, I'm testing this dialplan Block: <extension name="hold_music"> <condition field="destination_number" expression="^9664$"/> <condition field="${rtp_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"> <action application="answer"/> <action application="execute_extension" data="is_secure XML features"/> <action application="playback" data="$${hold_music}"/> <anti-action application="set" data="zrtp_secure_media=true"/> <anti-action application="answer"/> <anti-action application="playback" data="silence_stream://2000"/> <anti-action application="execute_extension" data="is_zrtp_secure XML features"/> <anti-action application="playback" data="$${hold_music}"/> </condition> </extension> using Bria Softphone; his SDP look like: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BtJ/e81Ed/6DMngVpHxmi+F5WHZQt89R9V1PYv27 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:BtJ/e81Ed/6DMngVpHxmi+F5WHZQt89R9V1PYv27 Why don't match with: ^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$ Dialplan: sofia/internal/1003(a)144.202.15.148 Regex (FAIL) [hold_music] ${rtp_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=on-false Thank you Regards -- --- I'm SoCIaL, MayBe

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REFER does not hit the dialplan on attended transfer
by Peter Steinbach 29 Aug '18

29 Aug '18

I try to find out a solution for the following: * A calls B (Dialplan A to B is executed with A as external number and B as local phone) * B is initiating an attended transfer to C (local phone), Dialplan B to C is executed * B is then executing the attended transfer to C, B sends a REFER to Freeswitch with C in the Refer header) * Freeswitch detects attended transfer, does not trigger a dialplan and redirects the call directly to user/C * Problem: C still sees B as caller_id in his phone display instead of A, dependent on the phone Type (Snom or Yealink) So I am wondering, if I could get the REFER to trigger a dialplan, so I can control the displyed numbers? I searched a bit and found the variables "force_transfer_dialplan" an "force_transfer_context". Looking into the code in switch_ivr.c, I can see that the variables are preset with "XML", resp. "default". Question: Can I use these variables to hit the dialplan on REFER. And if yes, how can I do this? -- With kind regards Peter Steinbach

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JIRA FS-9657 - Inbound HEP SIP capture for TLS transport
by Tom Parrott 28 Aug '18

28 Aug '18

Hi, I've found that Freeswitch does not allow capturing of SIP packets (using the HEP capture-server feature), when the SIP connection is over encrypted TLS transport. Outbound packets are correctly send to the HEP capture-server, even when over an encrypted TLS transport. This has previously been logged, and closed by someone else on this JIRA case. https://freeswitch.org/jira/browse/FS-9657 However I have looked into it and submitted a 3-line patch to get this working properly. I've posted my notes on the JIRA case and emailed the assignee directly, however have not gotten a response yet. My concern is that the JIRA case is already marked as closed and may be forgotten if its not re-opened. Please can this be looked into. Thanks Tom

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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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