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FreeSWITCH sources
by Social Boh 02 Jan '19

02 Jan '19

Hello, right now I can not download the sources of FreeSWITCH git clone -b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git fatal: unable to access 'https://freeswitch.org/stash/scm/fs/freeswitch.git/': Failed to connect to 2803:d000:fffe::151: Network is unreachable Thank you Regards -- --- I'm SoCIaL, MayBe

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mod_http_cache and mod_httapi Corrupted wav file after download
by Babak Yakhchali 02 Jan '19

02 Jan '19

Hi I'm using freeswitch 1.8.4 64 bit build using visual studio 2017 on win 10. When using playback with the file on disk the file plays ok. but if using http based sound urls which use http_cache or httapi the file playback is distorted. I attached the file and the downloaded cached file. The sizes are different!! everything works fine on debian

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Module's architecture image
by David Villasmil 02 Jan '19

02 Jan '19

Hello all, Years ago, Anthony sent out an image explaining FS' module architecture. I had it somewhere but can't find it now... Anyone got it? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Disabling 200 OK with contact headers of all the lines
by sagar malam 31 Dec '18

31 Dec '18

Hello , I am using FS SLA feature and it works very well.However i am facing an issue of registrations getting dropped as explained below : There are 3 phones registered with same extension number.Sofia is configured with "sip-expires-max-deviation" to randomise registration through expiry header in contact header.All the phones are Polycom. In case of shared lines FS adds contact header of all the lines(or phones with same extension) in 200 OK as shown below due to which all the phones are reading expiry timer from first contact header only.So in below example,Phone re registers after 379 seconds(first contact header) instead of 88 seconds(third contact header) leading to registration expiry on FS. Reason why phones are always reading expiry from first contact header is same Public IP(contact is re written by Proxy in front of FS) for all three phones which is confusing phone to identify its own contact header. Is there any way to configure FS to not send contact headers of all the registrations but only one that belongs to the line itself ? or any other way to fix it. ============================200 OK for register packet ============= 2018/12/08 13:36:35.426099 10.50.7.251:5070 -> 10.50.7.253:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 66.160.237.253:5060 ;branch=z9hG4bK9b46.f9b7a68125108f411537617d02bd48f8.0;received=10.50.7.253 Via: SIP/2.0/UDP 198.136.236.1:5060 ;rport=5060;received=10.50.8.1;branch=z9hG4bK9b46.e6f1aa3817a238f499aa3aafa4217425.0 Via: SIP/2.0/UDP 172.16.1.11;rport=1426;received=71.239.113.14;branch=z9hG4bK1df2302cD062D28F From: "Main Line" <sip:398@example.com>;tag=22CE54C0-84BCCF23 To: <sip:398@example.com>;tag=e8Fa2tND4U80D Call-ID: d66f0f0592c09746b903406f312eb0c2 CSeq: 590 REGISTER Contact: <sip:398@71.239.113.14:1071 ;alias=10.50.8.1~5060~1;x-nat=yes;pv-ip=172.16.1.12;pb-ip=71.239.113.14;pb-pt=1071;mac-address=64167f2ec274;transp t=tcp>;expires=379 Contact: <sip:398@71.239.113.14:56478 ;alias=10.50.8.1~5060~1;x-nat=yes;pv-ip=172.16.1.13;pb-ip=71.239.113.14;pb-pt=56478;mac-address=64167f2ebd54;tran ort=tcp>;expires=245 Contact: <sip:398@71.239.113.14:1426 ;alias=10.50.8.1~5060~1;x-nat=yes;pv-ip=172.16.1.11;pb-ip=71.239.113.14;pb-pt=1426;transport=udp;mac-address=64167 eb0c2>;expires=88 Date: Sat, 08 Dec 2018 08:06:35 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Content-Length: 0 =========================================================== -- Thanks, Sagar

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Controlling ptime in INVITE
by Joseph Waite 31 Dec '18

31 Dec '18

Evening All. I am having an issue, currently with Cisco SPA phones, where they are trying to use a prime of 30, however one of our providers does not support this and fails the call. Customer does not notice as call fails over to another provider as it should, however the failover provider is more expensive so not ideal for us. We don’t notice this unless we examine cdr logs. Is there a way, either to tell freeswitch to send a response to the phone negotiating time of 20 or to refuse the call if it comes in with time of 30? Regards

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hangup after 500 seconds of recording
by Abaci B 30 Dec '18

30 Dec '18

Hi, I noticed recently that when recording in freeswitch from a Verizon Wireless 4G LTE phone the call will hangup after 500 seconds of recording, I would guess they have a new media timeout detection system that ignores silence. I tried using setting "record_waste_resources=true" which usually solves the problem, but that didn't help, was wondering if anyone else experiences this issue or knows of a solution to the problem. Thanks

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[Zombie Ports Listening]
by Bilal Abbasi 29 Dec '18

29 Dec '18

Hi Users, I just installed FS 1.8.2, I enabled the bare minimal configurations needed for my platform, but i can see see two ports listening. I searched them and found out that its msrp ports, but i do not have any configurations/modules for these ports and they are still there, can somebody guide me how to put them down. tcp 0 0 0.0.0.0:2855 0.0.0.0:* LISTEN 26243/freeswitch tcp 0 0 0.0.0.0:2856 0.0.0.0:* LISTEN 26243/freeswitch I could not find any configuration where i can exclude them. Regards Abbasi

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FreeSWITCH library & dependency for Debian 8
by Alaadin Abdurrahman 28 Dec '18

28 Dec '18

Hello, i am newbie to freeswitch, i have installed freeswitch following the installation instruction on the FreeSWITCH 1.8 book everything works great. what i am trying to a accomplish is to install it from source code as other linux application so i would like to have a list of all the library dependency for Debian 8 i have seen to many tutorial on the internet but i would like to have list written by the freeswitch developers. best regards

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Inviting both normal and webrtc users
by Allan Kristensen 27 Dec '18

27 Dec '18

Hello, I'm trying to implement calling Webrtc clients (through kamailio proxy), which needs ICE, DTLS, etc. in SDP and so we are adding "media_webrtc=true" when calling and it works fine. But once we do this, many of our other sip clients complain because they only support/accept insecure RTP (RTP/AVP). So question is, shouldn't it be possible to offer both secure and insecure RTP in the INVITE from FS? If not then, would it advisable/possible to not send SDP in the initial invite from FS (we don't need early media for inbound anyway) and then wait for the client to offer the SDP instead ? (I'm afraid the ice process might cause a little silence interval at the start of the call because of the late media setup and user would always expect media to be up when answering the call) We really want to have generic way of handling of clients if possible, to avoid client detection (as currently only kamailio has this info, doing the location services) or any special config items on sip accounts. Best regards, Allan

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Merry Christmas and Happy New Year
by Andrew Keil 24 Dec '18

24 Dec '18

To the FreeSWITCH team, I just wanted to pass on my regards to the entire FreeSWITCH team. Merry Christmas and Happy New Year! It has been a great day for Visytel's solution, running on top of FreeSWITCH 1.8.2 (on Windows Server 2016). Today (Live on TV) a major charity donation service in the UK raised over GBP 120,000.00 using my solution (ie. transactions done over the phone using DTMF) successfully with no errors (all in just 3 hours). Fully PCI SAQ D compliant. Can't ask for more! Warmest Regards, Andrew Keil Visytel Pty Ltd https://www.visytel.com

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