June 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

group_confirm_key=exec : freeswitch start exec on 183-session progress instead of 200-OK..
by Julien Terrasson 19 Jun '19

19 Jun '19

Good morning, I'm using freeswitch Version 1.8.2 -3-a98a958ac3 64bit, and i have an issue with the following scenario : I bridge an outgoing call through a PSTN gateway, using group_confirm_key = exec, mainly to ask the called party confirmation before establishing the call. To do this, i use group_confirm_key=exec to start a lua script that : * Plays a vocal prompt to warn the calleR party that he has to wait for his call to be accepted. * Plays a vocal prompt to request the calleD to press a DTMF to confirm that he accept the call. This worked fine until i start dialing numbers that make the PSTN gateway to return a 183-session progress before 200-OK. When this happen, the group_confirm script is triggered on reception of 183-session progress : before the calleD party actually has answered the call ! Maybe i'm missing something, but I don't get why group_confirm_key=exec start on 183-session progress, as per my understanding, confirmation message should be requested only when B party has answered (200-OK). Does somebody knows how group_confirm_key=exec is supposed to work ? Should this be reported as an issue ? Regards, J.Terrasson

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How to install V8?
by Yehavi Bourvine 19 Jun '19

19 Jun '19

Hello, I am still running version 1.6, since I have a lot of JavaScript scripts which I cal from Freeswitch using mod_v8. In order to upgrade to 1.8 I need to install the V8 library. I've tried on Centos-7 and Ubuntu-18, but the install package is either old or not existent. I've tried installing from google's source code, but the configure script does not find it. How do I install V8? Thanks, __Yehavi:

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Alternate Voicemail Outgoing Message
by Sean Devoy 17 Jun '19

17 Jun '19

Hi, Is there a way to control the outgoing message when directing a call to VM? An Exec wants his VM calls to go to his office manager but with a different outgoing message than her own VM. Thanks, Sean

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SDP BUNDLE support
by Sowmya Venkat 15 Jun '19

15 Jun '19

Hi, I am trying to understand the expected behavior of latest FS when the client sends SDP offer with RTP BUNDLE and uses the exact same IP + port combination for both audio and video media components. Is BUNDLE ( https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-39) supported by FS? If not, can you pls let me know the when it is planned to add the support for the same? I am seeing the below errors in fs_cli. Can you pls confirm if the issue is because BUNDLE is not supported by FS? 2019-06-12 07:01:03.143578 [DEBUG] switch_core_media.c:3236 alternate payload received (received 96, expecting 111) 2019-06-12 07:01:03.143578 [WARNING] switch_core_media.c:3259 Could not change to payload type 96, ignoring... 2019-06-12 06:52:36.463638 [ERR] mod_opus.c:921 Decoder Error: buffer too small fs:4080 plc:false! 2019-06-12 06:52:36.463638 [ERR] switch_core_io.c:633 Codec RAW Signed Linear (16 bit) decoder error! [9] 2019-06-12 06:52:36.543581 [ERR] mod_opus.c:921 Decoder Error: corrupted stream fs:4080 plc:false! Here are the relevant SDP m and a lines from the client's offer: a=group:BUNDLE audio video .... m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 a=mid:audio a=rtcp-mux a=rtpmap:111 opus/48000/2 a=candidate:1 1 udp 2013266431 35.162.237.109 38673 typ host .... m=video 9 UDP/TLS/RTP/SAVPF 96 98 100 102 127 125 108 a=mid:video a=rtcp-mux a=rtpmap:96 VP8/90000 a=candidate:1 1 udp 2013266431 35.162.237.109 38673 typ host Thanks Sowmya

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ICTFax version 4.0 , Open source Fax over Ip software
by Tahir Almas 14 Jun '19

14 Jun '19

We are pleased to announce that ICTFax 4.0 has been released developed over freeswitch , ictcore and angular framework. New features include ATA / Extension support and REST API http://ictfax.org/ictfax-4.0-released-with-ata-api-support Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT

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DTLS 1.2 support
by Sowmya Venkat 14 Jun '19

14 Jun '19

Hi, Can you please let me know when it is planned to add DTLS Ver 1.2 support in Freeswitch ? Thanks Sowmya

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Mod XML Curl and IVR
by Sean Devoy 13 Jun '19

13 Jun '19

Is there a way to handle IVR's from MOD XML CURL? I have the directory and dialplan handle using my DB, but I don't see how to configure the IVR menus like this. Thanks, Sean

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Privacy headers selection
by Maciej Bylica 13 Jun '19

13 Jun '19

Hello I am running Freeswitch Version 1.6.20+git~20180507T185611Z~43a9feb7f8~64bit The goal i want to achieve is to send only PAI + Privacy headers to my traffic terminator. The problem i have to deal with is that in some circumstances PPI header is send instead. My configuration is: ... <action application="set" data="origination_privacy=screen"/> <action application="bridge" data="{max_forwards=70,originate_timeout=180,sip_contact_user=Test,ignore_display_updates=true,sip_cid_type=pid}sofia/gateway/terminator/+$1"/> ... The issue is when incoming INVITE has RPID with screen=no. when FS send PPI out instead of PAI. - Remote-Party-ID: "49111223344" <sip:49111223344@a1.b1.c1.d1 >;privacy=off;screen=no - Privacy: none - P-Preferred-Identity: <sip:49111223344@a1.b2.c2.d2> What can i do to achieve my goal? Do i need to tune my configuration? It is said https://freeswitch.org/confluence/display/FREESWITCH/sip_cid_type that "you must set privacy flag, otherwise P-Preferred-Identity will be inserted instead of P-Asserted-Identity.", but as you see origination_privacy flag is set to screen. Thanks in advance Mac

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sofia - wrong (external) IP address in contact for local calls
by Dušan Dragić 13 Jun '19

13 Jun '19

Hi, So, after not using freeswitch for a few years, I just did a clean install of Freeswitch 1.8.6 on debian 9 using packages from the repo and have a small issue maybe someone can help me with. I'm using the vanilla config here to keep it simple, also Freeswitch is started with -nonat. The problem I encountered is the contact field for local calls (on the same IP subnet as freeswitch) on the internal profile contains the external IP address which makes the phone send the response to the wrong IP. You can find attached a complete debug log with sip trace of a call between ext 1007 (192.168.0.66) and ext 1008 (192.168.0.65). Freeswitch is 192.168.0.140. The profile contains the default params: <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_rtp_ip}"/> This is what I get from sofia status: freeswitch@deb9> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 192.168.0.140,192.168.0.140 Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.0.140 Ext-RTP-IP 93.86.XX.XX SIP-IP 192.168.0.140 Ext-SIP-IP 93.86.XX.XX URL sip:mod_sofia@93.86.XX.XX:5060 BIND-URL sip:mod_sofia@93.86.XX.XX:5060;maddr=192.168.0.140;transport=udp,tcp WS-BIND-URL sip:mod_sofia@192.168.0.140:5066;transport=ws WSS-BIND-URL sips:mod_sofia@192.168.0.140:7443;transport=wss HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,PCMU,PCMA,H264,VP8 CODECS OUT OPUS,G722,PCMU,PCMA,H264,VP8 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 2 Is this the correct behavior? If I remove ext-rtp-ip and ext-sip-ip params from the config it works, but am I missing something obvious or is this a bug? -- Dušan Dragić

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call forward, rtp issue
by devang nathwani 13 Jun '19

13 Jun '19

Hello, Callee end(connected through private profile) playing IVR and after that the call forwards, at that time Freeswitch(private profile) getting reinvite from callee end but that invite not reaching to public profile and at caller end, leg a shows MoH in bridge. After that callee and caller end not connected through sip or rtp. Any help regarding what params should be used on private or on public facing profiles to avoid this? However if there is no reinvite from callee party, call works as it should, without issue. Thanks in advance.

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