July 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

internal number blacklist
by Matteo Cazzador 02 Jul '15

02 Jul '15

HI, i'm a novice, i'm trying to make a dynamic blacklist using mod_blacklist. My scope is block call from internal number to external . I need to create dynamic blacklist of internal number, not a blacklist of external number. Example i need to block dynamically internal number call to external payment number (over a billing target) Someone can help me please. Thanks a lot. -- Rispetta l'ambiente: se non ti è necessario, non stampare questa mail. ****************************************** Ing. Matteo Cazzador Email: mcazzador(a)gmail.com ******************************************

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Pass RPID from B-leg to A-leg
by Volodymyr Fedorov 02 Jul '15

02 Jul '15

Hello,Community. Are there any ability to keep RPID which I get in 180 Ringing on A-leg ? Maybe some kind of variable. B-leg RPID: Remote-Party-ID: "fedorov volodymyr" sip:44419@77.120.80.7 >;party=called;privacy=off;screen=no What freeswitch pass back to A-leg: Remote-Party-ID: "9944419" <sip:9944419@77.120.80.7 >;party=calling;privacy=off;screen=no Thanks! -- Cheers !

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So you wanna setup your own CA for WSS/SSL/TLS?
by Brian West 02 Jul '15

02 Jul '15

Someone should probably turn this into a nice how-to: Here is how I did it. wget http://www.openssl.org/contrib/ssl.ca-0.1.tar.gz tar zxfv ssl.ca-0.1.tar.gz cd ssl.ca-0.1/ perl -i -pe 's/md5/sha1/g' *.sh perl -i -pe 's/2048/2048/g' *.sh ./new-root-ca.sh ./new-server-cert.sh self.bkw.org ./sign-server-cert.sh self.bkw.org cat self.bkw.org.crt self.bkw.org.key > /usr/local/freeswitch/certs/wss.pem Setup Apache: default-ssl: SSLCertificateFile /usr/local/freeswitch/certs/wss.pem SSLCertificateKeyFile /usr/local/freeswitch/certs/wss.pem SSLCertificateChainFile /usr/local/freeswitch/certs/wss.pem Setup Sofia TLS: cat self.bkw.org.crt self.bkw.org.key > /usr/local/freeswitch/certs/agent.pem cat ca.crt > /usr/local/freeswitch/certs/cafile.pem vars.xml: <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/> Restart FreeSWITCH. Now make sure your system has ca.crt imported so it will trust your new found hotness. TEST: openssl s_client -connect self.bkw.org:443 openssl s_client -connect self.bkw.org:8082 Depending on what you've setup you'll see: subject=/C=US/ST=Oklahoma/L=McAlester/O=Tonka Truck/OU=Secure Web Server/CN= self.bkw.org/emailAddress=brian@bkw.org issuer=/C=US/ST=Oklahoma/L=McAlester/O=Whizzzzzzy Bang Bang/OU=Certification Services Division/CN=WBB Root CA/emailAddress= brian(a)bkw.org Or there abouts. -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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The FreeSWITCH 1.4.20 release is here!
by Ken Rice 02 Jul '15

02 Jul '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1dy0Hes The FreeSWITCH 1.4.20 release is here! The FreeSWITCH 1.4.20 release is here! This is a routine maintenance release and the resources are located here: Tarballs: http://ift.tt/1NwaFd8 Packaging: http://ift.tt/1U8FMQN Security issues: FS-7708 Fixed docs on enabling cert CN/SAN validation New features that were added: FS-7561 [mod_sofia] Add Perfect Forward Secrecy (DHE PFS) FS-7564 [mod_rayo] Added new algorithms for offering calls to clients FS-7623 [mod_amqp] Allow for custom exchange name and type for producers and fixed param name ordering bug caused by exposing these params FS-7720 Improve play_and_detect_speech to set current_application_response channel variable as follows: “USAGE ERROR”: bad application arguments’, “GRAMMAR ERROR”: speech recognizer failed to load grammar, “ASR INIT ERROR”: speech recognizer failed to allocate a session, and “ERROR”: any other errors FS-7743 [mod_skinny] Updated SKINNY on-hook action to hang up all calls on a device, except those in a short list of call states (or perform a blind transfer) and added a hook after completing the hangup operation to start ringing if there is an inbound call active on the device. Improvements in build system, cross platform support, and packaging: FS-7610 Fixed a gcc5 compilation issue FS-7426 Only disable mod_amqp on Debian Squeeze and Wheezy FS-7297 g729 installer The following bugs were squashed: FS-7582 FS-7432 Fixed missing a=setup parameter from answering SDP FS-7650 [mod_verto] Fixed crash when making a call from a verto user with profile-variables in their user profile FS-7678 Fixed for fail_on_single_reject not working with | bridge FS-7612 Fixed invalid json format for callflow key FS-7621 [mod_shout] Fixed a slow interrupt FS-7432 Fixed missing a=setup parameter from answering SDP FS-7573 Fixed 80bit tag support for zrtp FS-7636 Fixed an issue with transfer_after_bridge and park_after_bridge pre-empting transfers FS-7654 Fixed an issue with eavesdrop audio not working correctly with a mixture of mono and stereo FS-7579 [mod_conference] Fixed a bug not allowing suppression of play-file-done FS-7593 [mod_skinny] Fixed a bug where skinny phones would stomp on each other in database when thundering herd occurs FS-7597 [mod_codec2] Fixed encoded_data_len for MODE 2400, it should be 6 bytes. Also replaced 2550 bps bitrate (obsoleted operation mode) by 2400 FS-7604 [fs_cli] Fixed fs_cli tab completion concurrency issues on newer libedit FS-7258 FS-7571 [mod_xml_cdr] Properly encode xml cdr for post to web server FS-7607 Update URLs to reflect https protocol on freeswitch.org websites and update additional URLs to avoid 301 redirects. FS-7479 Fixed a crash caused by large RTP/PCMA packets and resampling FS-7524 [mod_callcenter] Fixing tiers, level and position should default to 1 instead of 0 FS-7622 [mod_amqp] Make sure to close the connections on destroy. Currently the connection is malloc’d from the module pool, so there is nothing to destroy. FS-7689 [mod_lua] Fixed a bug with lua not loading directory configurations FS-7489 [mod_unimrcp] Fixed a TTS Audio Queue Overflow FS-7467 [mod_callcenter] Fixing stuck channels using uuid-standby agents FS-7429 [mod_curl] Fixed a json formatting error

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Re: [Freeswitch-users] Presence Issues with Secure Websockets
by Lyle Pratt 02 Jul '15

02 Jul '15

My sincere apologies on my breach of etiquette. It will not happen again! Thank you for your update to my question on Jira. We will continue to work on the problem and contribute a patch ourselves if possible. -Lyle ---------- Forwarded message ---------- From: Anthony Minessale <anthony.minessale(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Date: Wed, 1 Jul 2015 18:44:46 -0500 Subject: Re: [Freeswitch-users] Presence Issues with Secure Websockets Its against our etiquette guidelines to update the list and Jira about the same thing. It adds 2 things to tend to about the same problem. Once you have filed a JIRA, there is no need to refer to it here. On Wed, Jul 1, 2015 at 10:58 AM, Lyle Pratt <lylepratt(a)gmail.com> wrote: > We opened a JIRA issue. Has anyone looked into this? > Presence events over WSS are basically broken. > https://freeswitch.org/jira/browse/FS-7680 > > Thanks, > Lyle Pratt > BetterVoice.com > > > Michael Jerris <mike@...> writes: > > > > If we are not sending the NOTIFY, please open a jira. Will need to > include full sip > debug with sofia debug and sip trace turned up. > > > > On Jun 10, 2015, at 1:11 PM, Thomas Quintana <thomas- > 78QPtSbfA2Jofqt5WZNmog(a)public.gmane.org> wrote: > > Hey Guys, > > > > We're subscribing to presence events via SIP.js using a secure websocket > (WSS) > and we can see the subscriptions being received by FreeSWITCH with a 202 > Accepted, but never receive any events via the notify event. > Alternatively, when we > move to a plain websocket (WS) everything seems to work as expected. > > > > > > > > After digging around we found that a few people were experiencing > similar issues > with FreeSWITCH and additionally with Kamailio http://lists.sip- > router.org/pipermail/sr-users/2014-July/084157.html. The issue was > resolved with > Kamailio by patching the contact header. In an attempt to achieve the same > results > we have experimented with the settings below but no success. > > > > > > > > > > <param name="NDLB-force-rport" value="safe"/> > > > > > > <param name="NDLB-received-in-nat-reg-contact" value="true"/> > > > > > > We also have an open ticket with the folks over at > SIP.js https://github.com/onsip/SIP.js/issues/184. Any help would be > greatly > appreciated. > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting@... > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users@... > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org >

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Re: [Freeswitch-users] Presence Issues with Secure Websockets
by Lyle Pratt 01 Jul '15

01 Jul '15

We opened a JIRA issue. Has anyone looked into this? Presence events over WSS are basically broken. https://freeswitch.org/jira/browse/FS-7680 Thanks, Lyle Pratt BetterVoice.com Michael Jerris <mike@...> writes: > > If we are not sending the NOTIFY, please open a jira. Will need to include full sip debug with sofia debug and sip trace turned up. > > On Jun 10, 2015, at 1:11 PM, Thomas Quintana <thomas- 78QPtSbfA2Jofqt5WZNmog(a)public.gmane.org> wrote: > Hey Guys, > > We're subscribing to presence events via SIP.js using a secure websocket (WSS) and we can see the subscriptions being received by FreeSWITCH with a 202 Accepted, but never receive any events via the notify event. Alternatively, when we move to a plain websocket (WS) everything seems to work as expected. > > > > After digging around we found that a few people were experiencing similar issues with FreeSWITCH and additionally with Kamailio http://lists.sip- router.org/pipermail/sr-users/2014-July/084157.html. The issue was resolved with Kamailio by patching the contact header. In an attempt to achieve the same results we have experimented with the settings below but no success. > > > > > <param name="NDLB-force-rport" value="safe"/> > > > <param name="NDLB-received-in-nat-reg-contact" value="true"/> > > > We also have an open ticket with the folks over at SIP.js https://github.com/onsip/SIP.js/issues/184. Any help would be greatly appreciated. > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@... > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@... > 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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RFC 4579 and FreeSWITCH
by Rick Block 01 Jul '15

01 Jul '15

A while ago there was a question about using REFER (RFC 4579 section 5.5) to add a new participant to a conference, see http://lists.freeswitch.org/pipermail/freeswitch-users/2012-December/090370… Can someone who is successfully using this post a howto including the relevant config, and exact REFER that works? We're getting a 202 Accepted in response to an out of dialog REFER, but no INVITE is being sent from FreeSWITCH. Our config and REFER are as below. Thanks, Rick Block <extension name="rfc4579_conferences"> <condition field="destination_number" expression="^(conf4579-.+)"> <action application="answer" data="is_conference"/> <action application="conference" data="$1-${domain_name}@cdquality++flags{rfc-4579}"/> </condition> </extension> To: sip:conf4579-1@10.4.0.7;isfocus From: sip:phil@10.4.6.31;tag=329478 Call-ID: 863823 CSeq: 28138 REFER Refer-To: sip:phil@10.4.0.213 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:phil@10.4.6.31 Via: SIP/2.0/UDP HANB65:5060:branch=z9hG4bk457125;rport Content-Length: 0

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no display-update after transfer
by wiebittewas 01 Jul '15

01 Jul '15

when using transfer on the phones, the call will be successfully transfered to the other phone, but it doesn't display the orinal caller-id, but keeps instead that id of the transfering phone. as I can see on the net, the refer-to info from the transfering phone to the server looks good: REFER sip:mod_sofia@192.168.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK850879522;rport From: <sip:24@192.168.1.3:5060>;tag=27328813 To: "27" <sip:27@pbx.priv>;tag=Net73KDB46X6r Call-ID: 33f2a7ee-9a39-1233-eab8-7674fbcada00 CSeq: 77522256 REFER Contact: <sip:24@192.168.1.3:5060> Max-Forwards: 70 Supported: replaces, path, timer Refer-To: <sip:20@pbx.priv?Replaces=1999853991-5060-136%40BJC.BGI.CBI.EC%3Bto-tag%3DQ0cS79ejyracg%3Bfrom-tag%3D219451532> Referred-By: <sip:24@pbx.priv> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 and freeswitch sends an ACCEPTED-msg for this REFER in the freeswitch.log I found entries like: [ALERT] sofia.c:1198 sofia_update_callee_id() sofia/internal/u620@192.168.1.4:5060 Same Callee ID "Outbound Call" <20> which leads to the situation, that no display-update will be send. unfortunately the REFER-handling of freeswitch doesn't seem to offer any possibility to intervene for debugging the available variables, so I have no idea, why freeswitch sees equal callee-ids. any suggestions how to look deeper or how to modifier the variables so that the update will be sent? regards w.

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2 incoming calls on freeswitch with bypass_media
by Jurijs Ivolga 01 Jul '15

01 Jul '15

Hi! I'm not sure that this is right place for this question, but here it is: Is it possible to connect(using uuid_bridge for example) 2 incoming calls on freeswitch and later to trigger re-invite messages, in such way that media will start to go directly between endpoint bypassing freeswitch. I think this is impossible and it is contrary with SIP RFC. Maybe somebody can confirm that this is impossible or not. :) Thank you in advance :)

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Asterisk canreinvite (directmedia) equivalent in freeswitch
by Ričardas Stoma 01 Jul '15

01 Jul '15

Does freeswitch have anything similar to canreinvite (asterisk sip setting)? I know there is bypass_media but it does not work in the same way as canreinvite in asterisk. It looks like bypass_media always tells freeswitch to send media directly between originator and terminator even though they do not have common codecs. I want to use bypass_media only when leg A and leg B have common codecs. If they do not have then freeswitch should stay in media path and handle codec transcoding. Is there an easy way to achieve this in freeswitch?

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