- FreeSWITCH-users - FreeSWITCH Mailing Lists

ICE_LITE FS-to-FS issues
by David Villasmil 09 Feb '22

09 Feb '22

Hello folks, I enabled ice_lite on FS and when the caller and callee are on the same FS, all is good. But if they are on different FS there’s no audio, although signaling “looks” good. The path is: A->FS->FS->B Anyone else seeing this? Thanks all! David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Something like a dial hook in bridge available?
by Bernd Krueger-Knauber 08 Feb '22

08 Feb '22

Hi freeswitcher, I run into a problem: We developed an iOS app with push notifications for our freeswitch pbx. So far it is working, but ... The ring group script (lua) directly call the bridge command including delay values. It is not possible to wait inside the iOS app after a push notification. To do it apple conform I need a hook, or something else, which is directly called before the invite stuff from sofia happens. Up to now I found only a hangup hook, but not a 'soon start dialing' hook. Is there any chance to come into the game, place a push notification, wait until the device is registered (with a timout of course) and then start the invite stuff? Best regards, Bernd

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Lua 5.4
by Trever L. Adams 08 Feb '22

08 Feb '22

Does FreeSWITCH work with lua 5.4.3 and later? When 5.4 came out I remember it causing FreeSWITCH to crash. Thank you. Trever

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 188, Issue 12
by Adiseshu Channasamudhram 07 Feb '22

07 Feb '22

Hello Henning how can i include a condition so that it applies these exports only if the >From is from a particular IP? Thanks a for your input by the way - helped a lot Regards Adi ________________________________ From: FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org> on behalf of freeswitch-users-request(a)lists.freeswitch.org <freeswitch-users-request(a)lists.freeswitch.org> Sent: Monday, February 7, 2022 3:20 PM To: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Subject: FreeSWITCH-users Digest, Vol 188, Issue 12 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..."

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(no subject)
by Adiseshu Channasamudhram 07 Feb '22

07 Feb '22

Hello Freeswitch Team Can one of you please let me know how i can copy the P-Asserted-Identity header from the incoming INVITE to the outgoing INVITE? I did go through most of the articles but none of them helped. I have tried the below so far in my dialplan <action application="set" data="sip_h_P-Asserted-Identity=${sip_P-Asserted_Identity}"/> <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/Mygateway/$1"/> <action application="bridge" data="{sip_cid_type=pid,origination_privacy=screen}sofia/internal/${destination_number}@${distributor(cust-site_p)}"/> Thank you in advance Regards, Adi

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ice_lite dynamically?
by David Villasmil 04 Feb '22

04 Feb '22

Hello all, I'm trying to enable/disable ice_lite dynamically by setting it on the xml dialplan, but nothing's happening. If i set it globally on vars.xml it does work, but i can't modify it on the dialplan... is this not possible? Thanks and Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Leg 480 kills all other legs?
by David Villasmil 04 Feb '22

04 Feb '22

Hello all, I’m calling multiple destinations separated by | One of the destinations is returning 480 and fs is canceling all other legs. I have continue_on_fail to true (I also tried with 480 and “Temp Unava…”) It was my understanding fs would just ignore that leg (no media is received) and allow the other legs continue… What am I missing? Thanks all! David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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cidlookup not working
by Joli Martinez 04 Feb '22

04 Feb '22

Hello, I have enabled the module cidlookup and added my config to the cidlookup.conf.xml file. I have also reloaded the module as well as restarted freeswitch. Freeswitch does not attempt to send any queries out to opencnam. I am doing a tcpdump on their IP's from the box and it is not even attempting to go out to them on port 443 or 53. I am running FS 1.10.7 on Debian 11. Any suggestions? <configuration name="cidlookup.conf" description="cidlookup Configuration"> <settings>  <param name=”url” value=”https://api.opencnam.com/v3/ phone/+${caller_id_number}?account_sid=<SID>&auth_token=<TOKEN>”/> </settings> </configuration>

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Multi-tenant not working
by Mircea Botoca-Huh 04 Feb '22

04 Feb '22

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No hang up, transfers or hold
by fs 03 Feb '22

03 Feb '22

My thought is that others may have run into this and have some pointers to share. I have Freeswitch 4.2.5 Branch 4.2 Switch 1.6.20 64 bit on Linux FusionPBX 3.16.0-11-amd64 #1 SMP Debian 3.16.84-1 (2020-06-09) x86_64 GNU/Linux. I'm using default settings as far as I know/recall. Obviously something must not be as it should be. My phones are a Polycom 650 and a Yealink T48G. My external test phone is a cell phone. It had only been used with a single extension for years until now when I added a second phone and extension. Before today I had noticed that hold was not working properly but did not use it. Attempting to transfer external calls between the two fails in the middle of the transfer. I press the transfer button and type the extension and then hit send but the call does not arrive and the call is hung up on the internal side but the external party does not know until something times out. If I put a call on hold it may be lost as well. Brief holds work. If I call out and then try to transfer that call the same thing happens. If I conference the other extension on an inbound call the external party is not getting any audio once the conference is initiated. If I then hangup on one extension the other is not aware that the call is over. The same is true for the external party. I can call the other extension and answer, when I hang up on either one the other does not know it's gone. It does not matter which phone originates or receives. At least audio works well. :) I've placed a FS log file here: https://drive.google.com/file/d/1VdoLqoPhr5mUE1ky1i2SJjXFChp9w-nM/view?usp=… The call duration is 2 minutes from 9:30 to 9:32. IP's and phone numbers have been replaced and placed within < > brackets. Internal IP's retains the last 8-bit number. --- fs

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