- FreeSWITCH-users - FreeSWITCH Mailing Lists

Add sip-h-P-Charge-Info
by Aina Mestre 31 Oct '22

31 Oct '22

Good afternoon, I have a freeswitch with pyfreebilling system and I would like to add the sip header "sip-h-P_Charge_Info" on the INVITE. I made the debug and saw that when lua was getting the variables was not able to obtain all data and I noticed that even if the trunk is connected with peers, and I receive and answer correctly the OPTIONS, when i make a fs_cli -x "sofia status", there is no connection [cid:image006.png@01D8EA23.CB52DD30] Do you know why it appears as if there is no users and i can have incoming and outgoing calls? Best regards

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limit max registrations
by Kakiman Expert 26 Oct '22

26 Oct '22

Hello Is there a way to limit the number of registrations on a freeswitch ? I mean not limit a number of registration for an extension, but globally, on the freeswitch level. (for example, i want at maximum 100 users registering on the freeswitch) thanks

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How to display switch_snprintf logs
by Andrew Reed 25 Oct '22

25 Oct '22

Hello, I’m debugging an issue where I’m not receiving events in some cases after a call is hung up. I’ve set “linger 60” and in the cases that it is successful I see the log “Socket Linger 60”, but in the cases where it fails I don’t see that log, making me think that the socket is not lingering correctly. I noticed while looking through the freeswitch code on github that there were some debug logs using “switch_snprintf” and I was wondering how to turn these on so that I can see more detail on what events are being sent and whether linger is being set correctly. Thanks, Andrew Reed

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Re: [Freeswitch-users] Issue with Enterprise originate and 487 "ORIGINATOR_CANCEL".
by Christian Berger 25 Oct '22

25 Oct '22

Thanks :) Am 14.01.22 um 16:15 schrieb Zvonimir Bužanić via FreeSWITCH-users: > Hi, I've created pull request: > > https://github.com/signalwire/freeswitch/pull/1520 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.

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switch_snprintf output
by Unai Rodriguez 19 Oct '22

19 Oct '22

Hello, I noticed in some parts of the freeswitch code there is logging using switch_snprintf. How can I see these in fs_cli or the freeswitch.log? Thank you, Unai Rodriguez

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Milion dollar question about CDR
by Gregor Nanger 14 Oct '22

14 Oct '22

Hi! We are using xml_cdr to process and save CDRS. But somehow process_cdr variable is broken with the latest version. I have the following scenario: process_cdr=true (so that both legs are posted to cdr URL) continue_on_fail bridge to gateway 1 continue_on_fail bridge to gateway 2 If first gateway fails FS goes to second one. Everything works ok. But only last cdr for gateway 2 is posted. Also only one callflow in XML. This worked fine. It used to post 2 xmls for everytime FS goes out to gateway. But now I have to fake enterprise originate that I get 2 cdrs. I append :_: on the end of the bridge. So, bottom line, without enterprise originate, only one XML and only last cdr, with fake enterprise originate 2 cdrs for both calls out to gateways Anyone noticed this behaviour, any advice?

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Video don't work over 3G/4G mobile data after upgrade to 1.10.7
by José David Jurado Alonso 11 Oct '22

11 Oct '22

Hi, The video is not displayed over 3G/4G mobile data after upgrade to 1.10.7 from 1.8.5 when call from 2N intercom device to linphone mobile app. In reverse order, calling from the linphone app to the intercom, everything works correctly. Audio works well in both cases. Really, the video did not work either using a WiFi connection but I have solved it by adding the parameter "bypass_media=true" before the bridge. However, it still does not establish the video on mobile data network. I have tried many NAT, media and others params configurations without success. I have used version 1.8.5 for years without any problems. The need for the FS update comes because in version 1.8.5 the BYE does not send to intercom device when calls are made from the linphone mobile app to the intercom device and in 1.10.7 it does work correctly Environment: * OS: Debian GNU/Linux 9.8 (stretch) on AWS EC2 * FreeSWITCH version (OK): 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) * FreeSWITCH version (KO): 1.10.7-release-19-883d2cb662~64bit (-release-19-883d2cb662 64bit) * Intercom device: 2N Helios * Linphone app version: Try with several versions 4.X.X and 5.X.X with same result * Video codec: H264 FS log traces: * Trace OK (over wifi): freeswitch_video_OK_wifi.log<https://github.com/signalwire/freeswitch/files/9754205/freeswitch_video_OK_…> * Trace KO (over 4G): freeswitch_video_KO_mobile4g.log<https://github.com/signalwire/freeswitch/files/9754203/freeswitch_video_KO_…> Configurations: * vars.log<https://github.com/signalwire/freeswitch/files/9754159/vars.log> * sip_profiles_external.log<https://github.com/signalwire/freeswitch/files/9754238/external.log> (I use the external profile to SIP) * dialplan_public.log<https://github.com/signalwire/freeswitch/files/9754186/public.log> * dialplan_default.log<https://github.com/signalwire/freeswitch/files/9754187/default.log> * dialplan_default_opening.log<https://github.com/signalwire/freeswitch/files/9754188/default_opening.log> The public dialplan transfer to default when call is from intercom to mobile and transfer to default_opening when call is from mobile to intercom. Issue: https://github.com/signalwire/freeswitch/issues/1828 [https://opengraph.githubassets.com/5be74d27029bc09a51f2c2fd7107922dbc67beb1…]<https://github.com/signalwire/freeswitch/issues/1828> Video don't work over 3G/4G mobile data after upgrade to 1.10.7 · Issue #1828 · signalwire/freeswitch<https://github.com/signalwire/freeswitch/issues/1828> Description: The video is not displayed over 3G/4G mobile data after upgrade to 1.10.7 from 1.8.5 when call from 2N intercom device to linphone mobile app. In reverse order, calling from the linpho... github.com Thanks!

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Want to attend Cluecon 2022 with us?
by Samir Doshi 07 Oct '22

07 Oct '22

Hi FreeSWITCH, I hope you are well. I just wanted to know if you are interested in visiting ClueCon 2022, Chicago, USA with us. Please let me know as soon as possible. Best regards, Inextrix Technologies Pvt. Ltd. [https://6r4dd.r.ag.d.sendibm3.com/mk/cl/f/jNuN7CkViLKhANb1nNwB1wFozuLL2_Sne… [Click here to unsubscribe](https://6r4dd.r.ag.d.sendibm3.com/mk/un/BUliLEECDGB6mciJObMmXp…

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FreeTDM not building
by Peter Hartmann 06 Oct '22

06 Oct '22

Hi folks, I updated my git source, compiled and installed. When I ran make install, I got several warnings that ftmod_* modules were existing and essentially not installed by this command. This is all in spite of ../../libs/freetdm/mod_freetdm being uncommented in modules.conf. When I look at them, indeed they were compiled when I updated in march. Any ideas why they aren't building now? This is some yak shaving because I am suspecting an issue with an old Sangoma analog card and hoping an update might help. Thanks in advance! Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com (212)203-8870 Public Key ID 0x1acfea8b5d88088d If I can't explain it to you in plain language, that means I don't understand it.

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Generate a new variable to reference in dialplan
by Aina Mestre 05 Oct '22

05 Oct '22

Good morning, I defined in my directory (/usr/local/freeswitch/conf/directory) a variable called accountcode (it's okey if it has to be another name because this is already in use) with value 7777. What I need is that in the dialplan is set as a variable so that i can use it as a SIP_Header for my call as sip-h-P-Charge-Info, like follows: <extension name="saliente"> <condition field="destination_number" expression="^(\d+)$" break="on-true"> <action application="set" data="effective_caller_id_number=${caller_id_number}"/> <action application="set"><![CDATA[sip_h_P-Charge-Info=${accountcode}]]></action> <action application="bridge" data="sofia/gateway/NP/$1"/> </condition> </extension> Is it that posible? How can I solve it? Thank you in advance

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