- FreeSWITCH-users - FreeSWITCH Mailing Lists

Multi-homed box - strange NAT question
by Jim Miller 02 Feb '21

02 Feb '21

Hi Folks I'm running FreeSWITCH Version 1.10.3-release~64bit (-release 64bit) on a FreeBSD 12.1 box. The issue I'm having is related to NAT, I'm sure no one has ever seen a post on this topic.... My configuration is a box that is multi homed with an Internet facing interface and a private IP LAN interface. The clients (Polycoms) are on the private LAN interface but behind a NAT (pfsense) on this subnet. If I have the clients route directly to the FS box's private LAN without NAT I can make this work but as soon as I NAT them (which I need to for other reasons) I don't see the registrations show up with fs_path or the other variables like I might expect. I've fiddled with the apply-nat-acl variable to no avail. Thoughts? Thanks Jim

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Re: [Freeswitch-users] Huge Jitter after "Incorrect Timestamp"
by Stephan Reich 02 Feb '21

02 Feb '21

Hello, we have problems with latency on some calls. In a pcap I see this entry. "Incorrect Timestamp" After this paket we get several seconds delay in the call. Are there parameters that fix this problem ? Stephan Reich Leiter Technik foncloud GmbH & Co KG Hahlweg 2a 36093 Künzell Tel: +49 661 968990 20 / Fax: +49 661 968990-99 Email: stephan.reich(a)foncloud.net Web: www.foncloud.net P.S.: Wussten Sie schon? Unter https://www.foncloud.net/wissen finden Sie zahlreiche Informationen und hilfreiche Artikel rund um unsere Produkte und Services. Registergericht: Amtsgericht Fulda, Persönlich haftende Gesellschafterin der foncloud GmbH&Co.KG: Global Brain Network GmbHGeschäftsführer der Global Brain Network GmbH: Peter Krug Sitz der Gesellschaft: Künzell. Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

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users registering from IP or FQDN
by Joli Martinez 01 Feb '21

01 Feb '21

Hello, I am running a small freeswitch system with several phones, users register with the IP of the freeswitch box. Since they are now working from home more, they need to be able to connect remotely. I have built a kamailio proxy and exposed it. I created an FQDN sip.mydomain.com and pointed it to the kamailio box. The registration comes in and kamailio sends it to freeswitch. I have added my doemain underneath the internal sip profile like this. I have also reloaded it. <domain name="all" alias="false" parse="false"/> <domain name="sip.mydomain.com" alias="true" parse="true"/> Registrations from the public internet still fail, but users registering with the IP work. What am I missing? 2021-01-19 12:19:37.480591 [WARNING] sofia_reg.c:1794 SIP auth challenge (REGISTER) on sofia profile 'internal' for [222(a)Sip.mydomain.com] from ip 1.1.1.1 2021-01-19 12:19:37.540512 [WARNING] sofia_reg.c:2930 Can't find user [ 222(a)Sip.mydomain.com] from 1.1.1.1 You must define a domain called 'Sip.mydomain.com' in your directory and add a user with the id="222" attribute and you must configure your device to use the proper domain in its authentication credentials. 2021-01-19 12:19:37.540512 [WARNING] sofia_reg.c:1739 SIP auth failure (REGISTER) on sofia profile 'internal' for [222(a)Sip.mydomain.com] from ip 1.1.1.1

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Scanners and botnet vulnerability
by Marc Bernard 01 Feb '21

01 Feb '21

Hello All, Is anyone else noticing that there is more and more scanners attempting brute force with no reply to auth request resulting in logging a lot of abandoned calls ? Scenario: - A scanner send an INVITE|REGISTER with no credentials - Freeswitch responds with authentication request and a challenge is send to logs; " 2021-01-25 12:27:39.306075 [WARNING] sofia_reg.c:1792 SIP auth challenge (REGISTER) on sofia profile 'public' for [1730(a)1.2.3.4] from ip 5.6.7.8" - Scanner does not respond - After a while, Freeswitch logs the following: 2ae23e93-c929-4089-a594-8e7af633ca88 2021-01-25 12:28:37.506078 [WARNING] switch_core_state_machine.c:687 2ae23e93-c929-4089-a594-8e7af633ca88 sofia/public/1730(a)1.2.3.4 Abandoned -- In our case, we made fail2ban more sensitive to auth failures logs which does not get triggered because of the scanner not even trying to send credentials. Wouldn't it make more sense for this log to include the IP of sip client that abandoned the call (5.6.7.8) instead of only the IP of the sip profile (1.2.3.4) ? This would allow us to have Fail2ban block this scenario more aggressively. Thoughts ?

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Freeswitch call recording
by Rasheed Kalapurackal 01 Feb '21

01 Feb '21

Dear All , i am a newbie just started trying and learning Freeswitch platform. Until now my area of focus was call recording for various telephony platforms like Avaya, cisco , and many more.. i would like to know if there is any API available in Freeswitch platform for recording call by a third party application , by streaming the voice as a duplicate RTP stream to the IP of Call Recording application . what I understood that there is Call Recording API like session recordFile available in the ESL for recording the calls in the system itself. But i couldnt find an option to record from a 3rd party application by getting the duplicate RTP stream. please let me know if any information is available on this query. Thanks and regards Abdul Rasheed

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Wrong Caller ID on Semi-Attended Transfer
by Joshua Rupp 01 Feb '21

01 Feb '21

Hello, We've a problem with the caller id in one transfer scenario: In our Example Party A calls Party B, Party B answers the call and transfers it to Party C. We're talking about the Caller ID in the phone of Party C now. If B is doing an attended or unattended transfer, the caller id ist updating correctly as you would expect it. But there's a third scenario, which is very common for users of older PBX systems: A so called "semi-attended transfer". This means, that Party B is initiating A attended transfer via BLF Key, but hangs up, before Party C has answered the call. In this case, the Caller ID should update to Party A, as soon as B hangs up his phone. But this is not happening: The Caller ID remains on Party B until C answers the call. Only then the caller id is updated. Why is the caller id not updated after B hangs up and before C answers the call? Is there any possibility to change this behaiviour? Thanks for your help Joshua

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Feeeswitch HA with Keepalived database conflict
by edson＠inoutglobal.xyz 01 Feb '21

01 Feb '21

Dears, I am configring FReeswitch HA with Keepalived approach just like this https://freeswitch.org/confluence/display/FREESWITCH/HA+keepalived Freeswitch database is shared with Postgresql-BDR replication. When executing fsctl crash on node 1, node 2 tries to recover call from its recovery table, but it somehow tries to insert it again causing following conclict. How to avoid this behaviour? >ERROR: duplicate key value violates unique constraint "recovery_pkey" < 2021-01-31 12:48:09.394 -03 >DETAIL: Key (uuid)=(66727254-af9d-47ef-ad9f-6f6bbb794eb6) already exists. < 2021-01-31 12:48:09.394 -03 >STATEMENT: insert into recovery (runtime_uuid, technology, profile_name, hostname, uuid, metadata) values ('4a864869-fe7f-4470-9992-20b91d09040f','sofia','internal','freeswitch-bndes ','66727254-af9d-47ef-ad9f-6f6bbb794eb6','<cdr core-uuid="4a864869-fe7f-4470-9992-20b91d09040f" switchname="freeswitch-bndes"> Regards

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Re: [Freeswitch-users] Rings Versus Seconds
by Robert Fitzjohn 28 Jan '21

28 Jan '21

> That is odd to the new users because we are used to setting the number of rings before our answering machine picks up Likely cos the phone is localized/sold in a certain country, the manufacturer has already done the required calculation. On Thu, 28 Jan 2021 at 12:37, KitchM via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: kitchm(a)tutanota.com > To: Joseph Barrero <joe(a)expert.net> > Cc: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Bcc: > Date: Tue, 26 Jan 2021 00:24:41 +0100 (CET) > Subject: Re: [Freeswitch-users] Rings Versus Seconds > Thank you. That clarifies things. > > So as I now understand it, the industry uses seconds to define the length > of time the callee will be hearing any sounds made during the ringing > process of the phone call before he/she picks up the call. Further, the > auto-attendent and/or routing of calls will be based upon counts of seconds > at each step > > Do I understand that correctly? > > That is odd to the new users because we are used to setting the number of > rings before our answering machine picks up. > > -- > Sent with Tutanota, the secure & ad-free mailbox: > https://tutanota.com > > > Jan 25, 2021, 22:53 by joe(a)expert.net: > > The issue is that the length of a ring can be different as heard by the > caller depending on the country/location of the recipient. Also, the > ringing on the recipients' device/phone can also be variable based on their > ring-tone preferences and the make and model of the device/phone. In other > words, 3 rings can consist of 2 different lengths as heard by the caller > and the callee. As such, duration in seconds is really the only > consistent way I know how to measure the timing of a ring. > > On Tue, Jan 5, 2021 at 3:27 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 13:27:42 -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 > > > > > > ---------- Forwarded message ---------- > From: KitchM via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> > To: Joseph Barrero <joe(a)expert.net> > Cc: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Bcc: > Date: Thu, 28 Jan 2021 04:37:26 -0800 (PST) > Subject: Re: [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

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Freeswitch Interop issue with MS Teams
by Arslan Saeed 28 Jan '21

28 Jan '21

Hi All. Has anyone seen any weird signalling behaviour of Freeswitch when interworking with MS Teams server? We have seen a few weird things where freeswitch is apparently not passing the signaling messages from one leg to the other in B2BUA mode. In one of the weird cases, freeswitch decides to not relay the SIP Bye message from the called party to the caller and just consumes it (resulting in the call to stay up from the caller's side). This has happened randomly as well as in one particular call scenario when a feature is activated on ms teams side. However SIP signaling wise we cannot see any difference between a good call (where SIP bye is relayed to caller by freeswitch and the one where it decides to stomp on it). Our freeswitch version is "FreeSWITCH (Version 1.10.5 -release-17-25569c1631 64bit)" I am not including any PCAPS or detailed configuration of freeswitch in the post yet and can provide it if required. Any ideas or link to some useful resources please? Thanks Arslan -- <https://overthewire.com.au/email-link/otwbanner>

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Re: [Freeswitch-users] Rings Versus Seconds
by Joseph Barrero 25 Jan '21

25 Jan '21

The issue is that the length of a ring can be different as heard by the caller depending on the country/location of the recipient. Also, the ringing on the recipients' device/phone can also be variable based on their ring-tone preferences and the make and model of the device/phone. In other words, 3 rings can consist of 2 different lengths as heard by the caller and the callee. As such, duration in seconds is really the only consistent way I know how to measure the timing of a ring. On Tue, Jan 5, 2021 at 3:27 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 13:27:42 -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

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