February 2022 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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>

4 7

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

04 Feb '22

1 0

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

2 2

Playback file stopping abruptly from freeswitch 1.10.5 and onwards
by Shahnawaj Khan 02 Feb '22

02 Feb '22

Hi All, Recently we have moved from freeswitch 1.10.3 to 1.10.5. and we observed that while executing the following command through ESL Command Execute playback(random.wav) freeswitch is stopping the playback earlier than the file playback duration.In this setup we are using freeswitch to generate the call as well. Our setup looks similar to this. Freeswitch_1------>opensips------>Freeswitch_2 Freeswitch_1: a) Originate and park b) playback some file on park (Using command though ESL) Freeswitch_2: a) playback some file on park (Using command though ESL) In Attached logs freeswitch has stopped the playback after 3 seconds but the file is of 10 seconds. But in another scenario where we replace the freeswitch_1 with a manual call using a softphone, Freeswitch_2 plays the file for the whole 10 seconds. Please let me know if anyone has faced the same issue or any insight regarding it. We observed the same issue with 1.10.7 but working fine in our old 1.10.3 setup. Thanks & Regards, Shahnawaj

1 0

SRTP calls randomly have no audio
by Gledi Musa 01 Feb '22

01 Feb '22

Hello, I'm using an instance of FreeSWITCH Version 1.10.7-release-19 in a Debian 11 VM, with OpenSSL 1.1.1k. The carrier only supports TLSv1, so I have hardcoded it in vars.xml and openssl.cnf to use only it, instead of TLSv1.2. On random calls there is no way audio, and without any changes, the next calls work, and then some other calls have no way audio. The logs don't throw any errors, but in the cases where there is no audio, I see that the DTLS negotiation doesn't go past HANDSHAKE, and stays like that for minutes, without throwing any errors. I have tried to set "legacyDTLS=1" as a channel variable, but that doesn't seem to have any effect. Any idea how to approach this?

1 0