January 2021 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ICTCore Version 1.1.0 Released
by Tahir Almas 07 Jan '21

07 Jan '21

Pleased to announce the release of ICTCore Version 1.1.0 https://github.com/ictinnovations/ictdialer. ICTCore is unified communciations development framework developed over freeswitch, Following you will find more details https://www.ictinnovations.com/ICTCore-Version-1.1.0-released-an-open-sourc… Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT

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FROM/P-Asserted-Identity and anonymous
by Grant Bagdasarian | CM.com 07 Jan '21

07 Jan '21

Hi, I'm changing the value of the From URI, in the dialplan, based on whether the Privacy: id header is present. If it's present, the value of the From URI is set to sip:anonymous AT anonymous.invalid, else it is set to sip:PHONE_NUMBER AT X.X.X.X. This works fine, but for some reason, during the bridging of the call using a gateway (with cid_type set to pid), Freeswitch sets the host part of the P-Asserted-Identity URI to anonymous.invalid (only during anonymous calls of course). Is there a way to make Freeswitch leave the host part of the P-Asserted-Identity header alone and set it to its IP address, while still utilizing the cid_type variable? Actual: From: <sip:anonymous AT anonymous.invalid> Privacy: id P-Asserted-Identity: "1234" <sip:1234 AT anonymous.invalid> Expected: From: <sip:anonymous AT anonymous.invalid> Privacy: id P-Asserted-Identity: "1234" <sip:1234 AT X.X.X.X> Removing the cid_type variable (or set to none), and inserting a P-Asserted-Identity header manually in the required format works fine. Also, when using the cid_type=pid, is it possible to add the user=phone param to the P-Asserted-Identity URI? Or does this also require setting the header manually? Regards, Grant

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Is [say] app removed from freeswitch?
by 王聡 07 Jan '21

07 Jan '21

Hello all, We’re trying to let freeswitch read some given words in Japanese, and I found it is possible via [say] app from documents: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+say <https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+say> However, [say] app hadn’t found in version 1.10.5, only [say_string] app was found, but it seems not support Japanese. * module [mod_say_ja] has loaded: [CONSOLE] switch_loadable_module.c:1803 Successfully Loaded [mod_say_ja] [NOTICE] switch_loadable_module.c:680 Adding Say interface ‘ja' * call [say] in dialplan: [ERR] switch_core_state_machine.c:276 Dialplan [say] not found, skipping * call [say_string] in dialplan: [ERR] switch_xml.c:3274 Can't find language ja. I wonder if [say] is no longer available in freeswitch, or there is a mistake on my deployment. Regards. —————————————————— 王聡 cong.wang.itsherpa(a)gmail.com 〒810−0073 福岡市中央区舞鶴2-3-6 赤坂プライムビル 2F 株式会社アイティーシェルパ

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rtp-timeout-sec VS media_timeout
by Allan Kristensen 06 Jan '21

06 Jan '21

We had some problems with "hanging channels" for our webrtc clients (via kamailio). To solve the problem I tried to use "media_timeout" setting but it didn't really work. So I tried the deprecated "rtp-timeout-sec" and this actually works fine? Not working: <param name="media_timeout" value="300"/> Working: <param name="rtp-timeout-sec" value="300"/> How to reproduce: Make a call using webrtc and just close browser window, after some time freeswitch should close the channel because of missing rtp. Instead it just keeps writing: "switch_rtp.c:853 No audio stun for a long time!" forever... Anyway, it works now....just curious why...a typo or bug? /Allan Using: FreeSWITCH Version 1.10.5-release+git~20200818T185121Z~25569c1631~64bit

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Re: [Freeswitch-users] Rings Versus Seconds
by Volodymyr Fedorov 06 Jan '21

06 Jan '21

Hi Ring tone duration depends on the country you live in: https://en.wikipedia.org/wiki/Ringing_tone You may try to do calculations by your own. BR, Vova On Tue, Jan 5, 2021 at 10:57 PM KitchM via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > > > ---------- Forwarded message ---------- > From: KitchM <kitchm(a)tutanota.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Thu, 24 Dec 2020 10:15:31 -0700 (MST) > Subject: Rings Versus Seconds > The system appears to use seconds instead of rings to measure incoming call > timing. Is that correct? > > If so, is there a way to change things to use number or rings instead? > > > > -- > Sent from: http://freeswitch-users.2379917.n2.nabble.com/ > > > > > ---------- Forwarded message ---------- > From: KitchM via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Tue, 05 Jan 2021 13:57:28 -0800 (PST) > Subject: [Freeswitch-users] Rings Versus Seconds > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com -- Best regards, Volodymyr

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Error for emails related to faxing
by Richard-Freeswitch 06 Jan '21

06 Jan '21

We have users who report email failures related to fax notifications. I'm digging for further information on how to reproduce the issue, but so far the only error in the FreeSWITCH log is: EMAIL NOT SENT, error [Cannot open tmp file In spandsp.conf.xml, spool-dir is set to /tmp. The directory exists and is world readable and writeable. fs_cli -x version returns: FreeSWITCH Version 1.10.5-release-17-25569c1631~64bit (-release-17-25569c1631 64bit) Any idea what would cause the EMAIL NOT SENT error?

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TLS 1.3 with verto
by David P 05 Jan '21

05 Jan '21

We use FS 10.5 on Debian for verto calls with setting sip_tls_version=tlsv1.2 and an apache2 reverse proxy for WebSocket logins so there's no port number in the WSS url so there should be no problem with restrictive firewalls for WSS login. In recent months we've seen some of the login attempts timeout, and there's nothing in the FS log at debug level to indicate why. But after a few minutes, verto's reattempts succeed in logging in. Has anyone else experienced this and found the cause? I thought I found the reason in our apache error.log, because it shows that some access attempts use TLSv1.3. I tried to get apache to reject these by switching its config to... SSLProtocol -all +TLSv1.2 ...but the TLSv1.3 attempts still behave the same. (Also, although these attempts appear in error.log, there's no hint about why they are in this log instead of access.log.) Can someone confirm that FS 10.5 doesn't yet support TLSv1.3? Cheers, David

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Inverted FROM/TO headers for BYE requests
by Unai Rodriguez 05 Jan '21

05 Jan '21

Dear List, We are having FreeSWITCH behind AVAYA infrastructure and when FreeSWITCH sends a BYE the From and the To headers are inverted. For example this call was initiated by John Smith however the BYE request has the From and To inverted: send 619 bytes to tcp/[10.125.15.122]:5060 at 16:57:04.079105: ------------------------------------------------------------------------ BYE sip:5188@10.125.15.122:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.155.20.104;branch=z9hG4bKa99Q7j7crjK9K Max-Forwards: 70 From: <sip:1581@10.155.20.104>;tag=Ze68N90tZ1D0c To: "John Smith" <sip:5188@example.com>;tag=54578355~d340a1b2-9d48-464d-9060-03cf872440b9-97644820 Call-ID: 2ad6b00-ff419a57-85bc53-66199b0a(a)10.125.15.122 CSeq: 30379120 BYE User-Agent: FreeSWITCH-mod_sofia/1.8.6~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 These seems to work for Cisco and other Telephony products but not the particular AVAYA system we're working with. Is there any way to modify this behavior and/or is this a known issue? Thank you so much -- unai

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Automatically authenticating client
by Martyn Davies 05 Jan '21

05 Jan '21

I'm trying to support a downstream client (actually a phone on a VoIP gateway) that must work without sending a registration to Freeswitch. I have a definition in the client directory that looks like this: <include> <user id="0101" cidr="192.168.1.0/24"> <variables> : <variable name="user_context" value="default"/> So, I'm assuming that if the source address matches 192.168.1.x, an invite from 0101 will automatically be authenticated, and the user context will be set to 'default' to go through the dialplan. However, what seems to happen is that the dialplan gets executed in the public context, 0101 doesn't match anything, and so the call fails. I also tried adding entries to the acl.conf file, e.g. e.g. <list name="lan" default="allow"> <node type="allow" cidr="192.168.1.0/24"/> </list> or <list name="localnet.auto" default="allow"> <node type="allow" cidr="192.168.1.0/24"/> </list> or adding an 'allow' line to the existing 'domains' list. When I run 'reloadacl', it looks like the rules are being picked up, e.g. 2021-01-05 11:06:07.631090 [NOTICE] switch_utils.c:642 Adding 192.168.1.0/24 (allow) [] to list localnet.auto 2021-01-05 11:06:07.631090 [NOTICE] switch_utils.c:642 Adding 192.168.1.97/32 (allow) [0101(a)192.168.1.241] to list domains However, none of this works, and I still end up in the public context when a call is placed. I would appreciate any examples or advice that people could offer to get my client authenticated, or could anyone give me pointers on how to debug the authentication process when the invite arrives? Regards, Martyn

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