August 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

180 with SDP converted to 183
by Volodymyr Fedorov 13 Aug '16

13 Aug '16

Hi community! I have simple question on B - leg I`m getting 180 with SDP: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 77.120.xxx.xxx;received=77.120.xxx.xxx;branch=z9hG4bKQQD0rvjpZecrB;rport=5060 From: "380XX555555" <sip:380XX555555@77.120.xxx.xxx>;tag=ym07v0XXr79ZQ To: <sip:0XX444444@62.221.xxx.xxx>;tag=d68e8a0 Call-ID: 5913dfbf-dbf1-1234-cfb6-002590773bfa CSeq: 95208960 INVITE Server: VegaTelecom Supported: timer,100rel Contact: <sip:0XX444444@62.221.xxx.xxx :5060;vtservice=b2buaservlet.siptosip;transport=udp> Content-Type: application/sdp Content-Length: 284 v=0 o=MG4000|2.0 323258 331209 IN IP4 62.221.xxx.xxx s=- c=IN IP4 62.221.xxx.xxx t=0 0 m=audio 48910 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0 a=X-vrzcap:identification bin=DSR2879 Prot=mgcp App=MG which is converted to the 183: 2016/08/13 15:07:29.954212 77.120.xxx.xxx:5060 -> 10.10.16.127:5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.16.127;branch=z9hG4bKnjrrsr6jgI3hgw5UEO6x From: <sip:85000@cc.out.co.com:5060>;tag=AwpvzqA To: <sip:0XX444444@77.120.xxx.xxx:5060>;tag=XB7eU5ctUyKDc Call-ID: fd0e5ed58368eb3abfe6ac75f500eb8d(a)10.10.16.127 CSeq: 1 INVITE Contact: <sip:0XX444444@77.120.xxx.xxx:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.26-37~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 243 Remote-Party-ID: "0XX444444" <sip:0XX444444@77.120.xxx.xxx >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1471072637 1471072638 IN IP4 77.120.xxx.xxx s=FreeSWITCH c=IN IP4 77.120.xxx.xxx t=0 0 m=audio 17412 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 So question is: Are there any possibility to keep 180 with SDP or it will be always converted to 183 ? Thank you in advance. -- Best regards, Volodymyr

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answer supervision
by David Villasmil Govea 13 Aug '16

13 Aug '16

Hello Guys, I've got this provider who is answering the call immediately after a 183, systematically, even if the call hasn't been answered yet. Is there any way of detecting busy, ringing and voice with fs? Maybe with "tone_detect" for busy and ringing and something else for voice? Never done this, never had the need. But the provider can't change this... Thanks for your help! David -- DVG -- Imagination is more important than knowledge Albert Einstein ᐧ

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LibSS7 2.0 with Freeswitch
by Mehdi Shirazi 13 Aug '16

13 Aug '16

Hi I want to use LibSS7 2.0 with freeswitch. There is a DAHDI internal loopback feature that can connect a minimal Asterisk instance with Freeswitch instance in same machine over internal DAHDI loopback E1 PRI. E1(SS7) -->DAHDI-->Asterisk-->PRI--> internal DAHDI loopback-->Freeswitch -Do you think this approach is more stable than converting SS7 to sip with Asterisk because Asterisk have minimal modules and just uses DAHDI internal native bridging ? -freeswitch debian binaries already included freetdm module for libpri ? Regards M.Shirazi

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Verto + Cordova
by Luiz Fernando Softov 13 Aug '16

13 Aug '16

Hi, I'm trying to create a Cordova APP using verto.min.js. I just make a register and a call, tried in browser (Chrome) and this work fine. But, when I make a android app, this wont work, since function navigator.getUserMedia(...) isn't native in Android. There is another native function to do the same thing? Has someone here already done this? If yes, what i need? I already googled it, without success yet. ps, I will make a IOS and WP too, if this work.

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server never sends "183 ringing back"
by Cj B 12 Aug '16

12 Aug '16

Hi all, Over the past two weeks all of a sudden Vitelity has been sending me TONS of call failed notifications, but I am seeing the calls hit my server and ring and they always drop at exactly 0:30’s if the called party doesn’t answer. I opened a ticket with Vit and they said that I’m not sending back “183 ringing back” to them. How can I send them this 183 ringing back? Thanks!

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Freeswitch replies with 200 ok on re-invite
by Jurijs Ivolga 12 Aug '16

12 Aug '16

Hi, I already asked this question on IRC channel. When my sip client sends reinvite to Freeswitch. Freeswitch instead of forwarding this reinvite to destination extension, just replies with 200 ok. Is it how it should work? How to force Freeswitch to forward reinvite to destination extension? With kind regards, Jurijs

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Fwd: Freeswitch call via WebRTC not work with Chrome 52
by Hai Bui Duc Ha 12 Aug '16

12 Aug '16

Hi all, - I have problem with Freeswitch call via WebRTC on Chrome 52, I can't establish the call and server appear log: *2016-07-31 17:28:01.977603 [ERR] switch_rtp.c:2917 audio Handshake failure 1* - I have already researched on Internet: https://groups.google.com/forum/?fromgroups#!topic/bigbluebutton-dev/5lR5WW… - They told that: I need to update OpenSSL version to 1.0.2 or later (currently is 1.0.1e-2+deb7u21) - Because, my OS(Debian Wheezy 7.11) don't have OpenSSL 1.0.2 on the repositories, so that I built this lib from source openssl-1.0.2h and also built libcurl 7.42. - After that, I re-built freeswitch like the guide at link https://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#FreeSWITCH… - I use FreeSWITCH Version 1.4.26+git~20151119T204804Z~66cd7d6b1a~64bit - But now, I still not work. You have any idea ? Thank for your support! Best regards, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876

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VertoJS Not working in Firefox
by Murugan Pandian 10 Aug '16

10 Aug '16

Hai, When i use VertoJS with chrome working fine no issue ,But when i use VertoJS with firefox call hangup automatically. OS: OSX FF version: 48 I attached my browser log

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Conditions / time of day
by Rick Jarvis 10 Aug '16

10 Aug '16

I have to admit, I’ve always struggled with conditions. Here I’m trying to have a different close time on a Friday to the rest of the week. Ignore the resulting actions, that’s just for testing. It appears the issue is matching the variable. I’ve tried export instead of set as well. Can anyone tell me where I’m going wrong, or if this should work? <include> <extension name="Time-of-day" continue="true"> <condition wday="6" time-of-day="09:00-16:30" break="never"> <action application="set" data="officestatus=open"/> <anti-action application="set" data="officestatus=closed"/> </condition> <condition wday="2-5" time-of-day="09:00-17:00"> <action application="set" data="officestatus=open"/> </condition> </extension> <extension name="mainin"> <condition field="${officestatus}" expression="^open$"> <action application="answer"/> <action application="playback" data="ivr/ivr-in_line.wav"/> </condition> </extension> <extension name="vmin"> <condition field="${officestatus}" expression="^closed$"> <action application="answer"/> <action application="playback" data="ivr/ivr-im_sorry.wav"/> </condition> </extension> </include> R

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export variable from bind_digit_action called dialplan
by Trever L. Adams 10 Aug '16

10 Aug '16

I have the following in the dialplan entry that starts recording: <action application="export" inline="true" data="rec_file=$${recordings_dir}/${caller_id_number}.${callscreen_destination_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> With set or export it only shows up on the leg that has the bind_digit_action setup. I need to see rec_file as a channel/session variable no matter whether the call is incoming or outgoing. (Both have the bind_digit_action set so only the internal user can activate it.) I have a script called from api_hangup_hook that needs to see this as it is used to setup call records in Odoo, if a recording took place, it is supposed to log an entry. Right now, it doesn't on incoming due to the bind_digit_action not setting a channel/session variable if it isn't the aleg. Any help would be greatly appreciated. Trever

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