- FreeSWITCH-users - FreeSWITCH Mailing Lists

Just in case you hadn't seen this ... beware DDOS
by Lawrence Conroy 10 Mar '22

10 Mar '22

Hi Folks, I haven’t seen a mail on the list regarding this problem. Everyone may well know of this already, but just.in case … Yes, it IS O/T for freeswitch, but list users may have such kit or interact with these systems, so ... see <https://www.akamai.com/blog/security/phone-home-ddos-attack-vector> As if the boys from Gaza doing lawnmower attacks were not irritating enough, we have these to deal with. Sigh. all the best Lawrence

1 0

Idiot question regarding ufw
by Alex Lake 09 Mar '22

09 Mar '22

Hi all. So I'm using ufw on my Debian 11 box, but have been having trouble trying to get it to stop calls from a certain source IP. eg. If I want to cut off an IP address from registration and placing calls via the box, using "ufw insert 1 deny...." doesn't seem to have an effect. I get the idea that this is because the simple way in which ufw manipulates iptables doesn't allow for existing connections to be terminated. How *should* I achieve my aims here?!

1 0

fs_cli not starting on fresh installation on Debian
by Rajesh Kanade****CTR 07 Mar '22

07 Mar '22

Hi all, I have installed the FreeSWITCH version: 1.10.7-release-19-883d2cb662~64bit (-release-19-883d2cb662 64bit) on Debian GNU/Linux 11 (bullseye) version. 'service freeswitch status' command shows freeswitch as running. When I run fs_cli command from the server itself ( connecting to 127.0.0.1) I get the following error -> [ERROR] fs_cli.c:1691 main() Error Connecting [] I have tried manually creating fs_cli.conf as suggested on the net but it did not help. I read a few places that this could be because IPv4 is disabled by default on debian but did not have any instructions how I could fix this. I am a newbie to FS so any help would be appreciated.

4 3

Lua access to ESL
by Alex Lake 04 Mar '22

04 Mar '22

Using a freeswitch build from a package and want to access esl from a lua script. Is this possible? Or do I have to go the compiled route? FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit Cheers, Alex

3 2

Delay after RTP-Pause by FreeSwitch
by Christian Berger 03 Mar '22

03 Mar '22

Hi, we have an odd problem which is unfortunately not reproducible. We have a call going into a callcentre-queue. At the moment the call is picked up, FreeSwitch stops sending RTP to the caller for about 2.8 seconds. When FreeSwitch sends RTP again, it sends the data it got from the called party during those 2.8 seconds. This stream stays delayed by that amount. Does anybody have any idea why FreeSwitch holds on to those voice frames for such a long time? Thanks Christian Christian Berger foncloud GmbH & Co KG Hahlweg 2a 36093 Künzell Tel: / Fax: +49 661 968990-99 Email: Christian.Berger(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.

1 0

Copying From header
by Adiseshu Channasamudhram 01 Mar '22

01 Mar '22

Hello FS Team, How can I copy the From header in INVITE from incoming leg to the outgoing leg? Basically i have the carrier send the From header as below [not the clean way] From: <sip:1234567890@1.19.10.3>;isup-oli=00;tag=3418741140365741_c2b08.2.4.1638259348522.0_6724417_21166610 I want to make sure i send out the oli in the outgoing INVITE From header. Note that in the incoming From , the oli is outside the uri Thanks a lot in advance Regards Adises

2 4

Re: [Freeswitch-users] PCAP help and many call failures PLEASE HELP
by mayamatakeshi 24 Feb '22

24 Feb '22

On Thu, Feb 24, 2022 at 12:27 AM Sean Devoy <sdevoy(a)bizfocused.com> wrote: > > > I have debug level FS logs. There are so many lines it is difficult to > follow a single call. > > > Do you have the logs related to the capture file? If yes, then search for the Call-IDs of the troubled call: Leg1: Call-ID: f0544e98-378154a6(a)192.168.2.111 Leg2: Call-ID: 26e3398f-0e97-123b-3b83-16f3d912a996 For example, I did a test call with Call-ID: 090abe39-188c-4f06-b371-e8ffe3d862f8 So I grepped the freeswitch.log file with: [root@lab002 ~]$ grep 090abe39-188c-4f06-b371-e8ffe3d862f8 /usr/local/freeswitch/log/freeswitch.log 4b860aaf-b481-4e26-8886-37321c2fc3f3 2022-02-24 06:49:17.131915 [INFO] sofia.c:10414 sofia/external_in/123412341234(a)test.com receiving invite from 127.0.0.1:5060 version: 1.10.7-dev git dd24113 2021-08-25 17:37:19Z 64bit call-id: 090abe39-188c-4f06-b371-e8ffe3d862f8 The 4b860aaf-b481-4e26-8886-37321c2fc3f3 above is the channel uuid (the unique identifier of the channel in freeswitch). The freeswitch log file will include that channel uuid in all log lines related to that channel. So you can grep your freeswitch.log file for it. If no clue about the problem is found, do the same for the other leg. > This happens are about 50% of ALL calls. > > Here is the dial plan: > > <extension name="ten_or eleven_digiti_dial" > > > <condition field="destination_number" expression=" > ^(?:1|\+1)?([2-9]\d\d[2-9]\d{6})$"> > > <action application="set" data="ringback=${us-ring}"/> > > <action application="set" data=" > effective_caller_id_number=${outbound_caller_id_number}"/> > > <action application="set" data="ani=${outbound_caller_id_number}"/> > > <action application="set" data=" > origination_caller_id_number=${outbound_caller_id_number}"/> > > <action application="set" data="effective_caller_id_name=REDACTED"/> > > <action application="set" data= > sip_from_uri=${outbound_caller_id_number}@ > <sip_from_uri=$%7boutbound_caller_id_number%7d(a)64.136.173.30>REDACTED/> > > <action application="set" data=" > sip_from_user=${outbound_caller_id_number}"/> > > <action application="set" data="sip_from_host=REDACTED"/> > > <action application="set" data="sip_h_X-CID=${sip_call_id}"/> > > > > <action application="set" data="continue_on_fail=true"/> > > <action application="set" data="hangup_after_bridge=true"/> > > <action application="set" data="sip_h_X-CID=${sip_call_id}"/> > > <action application="bridge_export" data=" > hold_music=/etc/freeswitch/recordings/REDACTED-moh.wav"/> > > <action application="bridge" data= > {sip_from_uri=${outbound_caller_id_number}@REDACTED,sip_from_user=${outbound_caller_id_number},sip_from_host=REDACTED}sofia/gateway/voip-innovations-outbound/1$1 > <%7bsip_from_uri=$%7boutbound_caller_id_number%7d@64.136.173.30,sip_from_user=$%7boutbound_caller_id_number%7d,sip_from_host=64.136.173.30%7dsofia/gateway/voip-innovations-outbound/1$1> > /> > > </condition> > > </extension> > > > OK. So there is nothing setting an answer timeout (so it would default to 60 seconds). The variable controlling it would be this one: https://freeswitch.org/confluence/display/FREESWITCH/call_timeout

2 1

call duration for bash script
by thomas peterseil 23 Feb '22

23 Feb '22

hello freeswitch-users, i would like to run a bash script after a call and for this script i need the duration of that call. i found the variable "billsec", but i couldn´t find any dialplan examples how to use this variable. can someone give me a hint how i can hand over the duration of the call to a bash script in the dialplan. thank you very much! best regards, thomas

3 2

Mitel/Astra phone UDP and NAT issue
by Sean Devoy 23 Feb '22

23 Feb '22

Hi, At a customer site behind NAT we have three brands of phones: Mitel, CISCO, and Yealink. Cisco and Yealink work like a charm. If the Mitel phone is set to UDP it does not respond to the "Not Authorized" response to its "Registration", presumably because it did not make it back through NAT. Using TCP it works. Any ideas for NAT support with Mitel? Regards, Sean Devoy VP Operations and Development Business Focused Internet Systems, Inc.

1 0

RE: [Freeswitch-users] PCAP help and many call failures PLEASE HELP
by Sean Devoy 23 Feb '22

23 Feb '22

Since about half of the calls fail and half work, could it be that the calls that are answered quicker are making it under some time limit and the others are not? What parameters should I look at in FS for outbound call answer time limit?? Thanks, Sean -----Original Message----- From: FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org> On Behalf Of Sean Devoy via FreeSWITCH-users Sent: Tuesday, February 22, 2022 11:57 AM To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] PCAP help and many call failures PLEASE HELP _________________________________________________________________________ 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

1 0