- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Bye does not hangup
by Raimundo Pérez Nieves 25 May '18

25 May '18

The problem I am having is that the bye in sip trace isn’t hanging up the call, It keeps waiting to another bye from another ip, look the time between that bye from sip trace and the hangup that FS shows in debug. Thanks for your time

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Multiple external profiles
by Andrew Colin 25 May '18

25 May '18

Hi Guys If i am running 2 external sip profiles both running port 5080 but with different ip's how do i tell a specific gateway to use which sip profile?

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Configuring timer-T1X64
by Atul Thosar 25 May '18

25 May '18

Hi All, I want to change the timer B of INVITE Client Transaction. So I set following in conf/sip_profiles/external.xml <param name="timer-T1X64" value="4000"/> Restarted freeswitch and observed following debug line in logs 2018-05-10 16:57:43.799719 [DEBUG] sofia.c:4629 timer-T1X64 [4000] However, INVITE Client Transaction is not timing out in 4 sec. It still takes 32 sec to timed out. What could be the reason? Did I miss anything? Btw I am using external Lua client to control the freeswitch call flow. My Lua script connects to freeswitch over ESL connection and fires following command to make outbound call [DEBUG Thu May 10 17:00:17 2018] : command for outbound call is: lua ~freeswitch.Session('{origination_uuid=197422a3-e29e-452a-bf4e-86a6895a8cd5,sip_invite_route_uri=<sip:148.147.174.99;lr=on;ftag=K9ZF9aXv8ra1e>}sofia/external/ sip:18447770001@domain.com', freeswitch.Session('0b4aaab6-cee8-1236-d897-005056b308ba')) 0b4aaab6-cee8-1236-d897-005056b308ba is call-id of my incoming call to freeswitch. I am using following version of freeswitch freeswitch@as80> version FreeSWITCH Version 1.8.0~64bit ( 64bit) -- BR, Atul

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Bye does not hangup
by Raimundo Pérez Nieves 25 May '18

25 May '18

Hi, I have this particular problem. When I turn on sip trace global, I get a bye from legB but it does not hangup the call, how can I capture this bye and hangup. I attach sip trace Thanks

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FreeSWITCH offering SRTP on Re-INVITE
by Andrew Cassidy 24 May '18

24 May '18

Good afternoon All, I have experienced the following issue with Zoiper specifically (I have a support ticket open with them currently) and was wondering if it's something that could/should be fixed FreeSWITCH side. If rtp_secure_media is set to optional, and SRTP is disabled on Zoiper, when FreeSWITCH sends the Re-INVITE, it sends crypto lines. Zoiper then replies with a 514 Unsupported Media Type and the timer refresh fails, causing Zoiper to hang up the call. My current feeling is that as Zoiper ignored the crypto on the initial INVITE it should be them that then handles the Re-INVITE consistently by also ignoring the crypto lines but I'm not familiar enough with the RFCs to make a judgement. The workaround is to either enable SRTP in Zoiper or don't set rtp_secure_media to optional when Zoiper is in use. Kind regards, -- *Andrew Cassidy BSc (Hons) MBCS* Managing Director 0330 44 55 960 <+443304455960> andrew(a)cassidyweb.co.uk www.cassidyweb.co.uk

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Freeswitch in google cloud platform
by Karanath sachidanandan, Sanooj 24 May '18

24 May '18

Hi Guys , Anybody using Freeswitch in Google cloud platform ? I am trying to have a multi region freeswitch installation in google cloud with Sip load balancer Kamailio . But challenge is to have a multi region load balancing. Google’s TCP load balancer looks promising but don’t support Sip ports(5060, 5061) Any thoughts on that or any other suggestion? Thanks in advance https://cloud.google.com/compute/docs/load-balancing/tcp-ssl/tcp-proxy Regards Sanooj ________________________________ The information in this Internet Email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this Email by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients any opinions or advice contained in this Email are subject to the terms and conditions expressed in any applicable governing The Home Depot terms of business or client engagement letter. The Home Depot disclaims all responsibility and liability for the accuracy and content of this attachment and for any damages or losses arising from any inaccuracies, errors, viruses, e.g., worms, trojan horses, etc., or other items of a destructive nature, which may be contained in this attachment and shall not be liable for direct, indirect, consequential or special damages in connection with this e-mail message or its attachment.

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Trouble Shooting Hung Sofia Profile
by Geoff Mina 24 May '18

24 May '18

Anyone have any pointers on the best way to troubleshoot hung Sofia profiles? We are on 1.6.19 installed via Yum on CentOS 7 and have had consistent system hangs every few days. You can see the SIP message come in on the siptrace logs and then there is never a response. Process restart required to correct. CLI is functional. Nothing interesting in logs. Any pointers greatly appreciated. Thanks. -- GEOFF MINA Chief Executive Officer Connect First / Contact Center Solutions, Built Better. 2545 Central Ave #200, Boulder, CO 80301 720.335.5924 Connect First / Contact Center Solutions, Built Better www.connectfirst.com This email and any files transmitted with it are confidential and are intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error, please notify the system manager.

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DUAL NIC
by Markus Hammerer 24 May '18

24 May '18

Helo I have a Server with two nics And the freeswitch works with the wrong nic How can i configure this with the other NIC? Where i can do this? Thanks Markus

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how to make FS to listen for 2 RTP streams on one call ?
by Evtim Ivanov 23 May '18

23 May '18

Hello all, I'm trying to implement this (shown in the picture) scenario, but to use FS instead of OrekaTR: There are several small bumps on the road, which I successfully overcame, but finally I met a challenge which is bigger, than my skills, so I need the community help. The problem is, that RtpProxy can't mix the media, so instead of one mixed stream on the forked call, I have 2 separate RTP streams. I have no idea how to make FS to listen both of them. Is this possible and how? Any advice is highly appreciated. Best! Evtim

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Adding SIP Headers in Javascript
by Preston Hagar 23 May '18

23 May '18

Quick Version: Is there some special way to set custom SIP headers when you are bridging a call from Javascript instead of XML? Full Details: We am using Voice Operator Panel (VOP) as a receptionist console. We have multiple external numbers that all funnel into a few receptionist desks. If I add a SIP Diversion header, then VOP will use that to show the receptionist which number the caller dialed to reach them, which would help things out a lot. If I add it in an XML dialplan and then do a direct bridge to one of the receptionist numbers, it works great (snip of working direct dialplan): <action application="set" data="sip_h_Diversion=<sip:${ destination_number}(a)pbx.example.org >;privacy=off;reason=unconditional;counter=1;screen=no"/> <action application="bridge" data="user/1000(a)pbx.example.org"/> The problem is I need a bit of advanced logic to figure out which receptionist are online and then ring those, then fail over to other numbers and ultimately IVR if no one answers, so I instead pass calls to them off to javascript, so the bridge line above becomes something link: <action application="javascript" data="${jslib}/frontdesk.js ${dialed_extension} ${domain_name} ${destination_number}" /> I've tried setting the SIP header in the XML above the javascript line and then exporting it <action application="set" data="sip_h_Diversion=<sip:${ destination_number}(a)pbx.example.org>;privacy=off;reason=unconditional;counter=1;screen=no"/> <action application="export" data="sip_h_Diversion" /> and setting it in the Javascript: var diversionHeader = 'sip_h_Diversion=<sip:' + $destination_number + '@ pbx.example.org>;privacy=off;reason=unconditional;counter=1;screen=no'; session.execute('set',diversionHeader); session.execute('export','sip_h_Diversion'); but in both cases (or with both in place), the SIP header is never written. Is there some magic to setting SIP headers in Javascript? I can't seem to find any examples or documentation about it. Thanks!

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