January 2021 - FreeSWITCH-users - FreeSWITCH Mailing Lists

[FreeSWITCH sched_hangup does not execute 'execute_extension']
by Bilal Abbasi 20 Jan '21

20 Jan '21

Hi Users, I want to build a scenario in which if the B-Party hangup the call, A-Party can still be able to process through the dialplan. This works fine if B-Party hangup the call, but if the B-Party is hanup forcefully via sched_hangup (timeout), call gets terminated i-e A-Party is not able to reach 'execute_extension' application. Here is my dialplan snippet https://justpaste.it/2ppra I have attached as an image as well. Regards Bilal Abbasi

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register with FQDN or IP
by Joli Martinez 19 Jan '21

19 Jan '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. On the freeswitch box my internal sip profile looks 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 Thanks

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AUDIO RTP REPORTS ERROR: [Remote Address Error!]
by Malik Ehtasham 19 Jan '21

19 Jan '21

hello everyone, I am using sip.js for webRTC with Freeswitch, I am able to initiate a call from sip.js to other sip extension but fail to answer the incoming call on sip.js, and the following error occurs in the fs_cli. 2021-01-18 21:31:15.701411 [DEBUG] switch_rtp.c:4450 Starting timer [soft] 960 bytes per 20ms 2021-01-18 21:31:20.701423 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Remote Address Error!] 2021-01-18 21:31:20.701423 [NOTICE] switch_core_media.c:9670 Hangup sofia/internal/5de16hg3(a)ccujprafrvr6.invalid [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 2021-01-18 21:31:20.701423 [ERR] sofia.c:8506 RTP Error! -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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Freeswitch crash reloading mod_callcenter
by Kunal Mittal 19 Jan '21

19 Jan '21

Hi, I have the latest version of the freeswitch 1.10.5 running with debian 10.7. It crashes everytime on reloading the mod_callcenter module after 3-4 minutes. Any suggestions? Is it a known bug? Any suggestions on how to troubleshoot this issue? Freeswitch version - 1.10.5 Debian - 10.7 Kunal

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AWS - Audio Delay - rtp timer
by Jacob Greene 19 Jan '21

19 Jan '21

Hello everyone, Recently, I am running into an issue with some severe audio delay with some freeswitch boxes running on AWS. It gets to the point where audio is delayed up to 10 seconds in both directions. The calls start off fine and slowly get worse. The only way I've been able to fix this is by disabling the rtp timer on the sofia profile(rtp-timer-name=none). Making this change immediately fixes the issue. No obvious system bottlenecks, 1/5/15 min load averages all less 1 on dual core CPUs, tons of free memory, no I/O issues on the nic, etc. I suspect this might have something to do with some weird CPU sharing AWS is doing on their hypervisor, but I'm not sure. I can't find a ton of info on exactly what the rtp timer does, other than disabling it "disables asynchronous rtp" and "makes freeswitch handle media the same way as asterisks". I'm not really sure what this means. I've read it's less effecient and come across a couple very old post on the mailing list of Anthony steering people away from disabling the rtp-timer. Has anyone had a similar problem? Or does anyone have some more information/resources regarding exactly what the function of the rtp timer is? I'm trying to get in front of this. What are the drawbacks of not using an rtp timer? Thanks for reading!

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Using last_bridge_hangup_cause
by olle＠zaark.com 18 Jan '21

18 Jan '21

Hi, I have a need of using last_bridge_hangup cause in a condition statement to activate a second call leg but I can't get it to work. The log line prints the correct values of last_bridge_hangup_cause, "NO_ANSWER" , but when I use it in the condition it's not working and the same if I try to assign it to another variable. Here is a code snippet I use in the dialplan: <action application="set" data="hangup_after_bridge=false"/> <action application="bridge" data="sofia/gateway/mygateway_1/1234567"/> <action application="log" data="INFO Last bridge hangup ${last_bridge_hangup_cause}"/> .. Do some stuff ... <condition field="${last_bridge_hangup_cause}" expression="NO_ANSWER"> <action application="bridge" data="sofia/gateway/ mygateway_2/1234567"/> </condition> I would prefer using the dialplan without implementing this in lua if possible. Any help appreciated /Olle

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external ip and Nat
by Kakiman Expert 15 Jan '21

15 Jan '21

Hello I am working on freeSWITCH 1.10.2 with webRTC. I have a private address on my server, that is translated on a static public IP in order to reach webRTC/FS through Internet. In the vars.xml, i have then indicated this public IP as *external_sip_ip* and *external_rtp_ip *and everything is working. Now, I also want to be able to reach webRTC with its private IP, so what is the value to configure in vars.xml external_sip and external_rtp ? I can not put both public and private IP. thanks for your helpful answers Joe

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call / session stats
by Kakiman Expert 12 Jan '21

12 Jan '21

Hello, I am working on freeSWITCH 1.10.2 (with webRTC / OPUS) and looking for some calls statistics. Indeed, I saw that we can have real time stats (show calls/session...), but I don't find any way to have it later For example, having the registration or calls for a daily basis, or for a specific period. I just saw that with "show status" we have such a statistic but without any period/range information. 637 session(s) since startup / 0 session(s) - peak 4, last 5min 0 Is there a solution to have these reports ? thanks Joe

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play ringtone until connecting media
by אריה קלטר 10 Jan '21

10 Jan '21

Hello After play_and_get_digits, in some cases i am bridging the call to external server. The remote server answers with 200 OK after something like 0.3 second But, the remote server starting to send media only after 2.1s Any idea how to play ringtone (or another wav file) until the remote server really sends media? Regards Arye

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ICTDialer Version 4.0 released
by Tahir Almas 08 Jan '21

08 Jan '21

Pleased to announce the release of ICTDialer Version 4.0 https://github.com/ictinnovations/ictdialer . ICTDialer is a freeswitch based unified communications auto dialling front end for mass communications and telemarketing needs. Fore more details , please click following link https://www.ictinnovations.com/ICTDialer-Version-4.0-released-open-source-f… Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT

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