January 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

start_dtmf false positives
by Roman Dissauer 14 Mar '18

14 Mar '18

Hi Guys, we have troubles reliably recognizing DTMF tones by using dialplan application start_dtmf. From our carriers we get some calls with inband DTMF and some with RFC2833. To our customers we need to always do RFC2833. Therefore we check SDP information and enable start_dtmf when RFC2833 is not available on the incoming call. When start_dtmf is executed we have a lot of false positives mostly at the start of audio but also during the call. Is there a way to prevent false DTMF detection? Does anyone have same issues and a solution for it? Kind regards Roman Dissauer RDI SOLUTIONS e.U. Rosasgasse 13/24 1120 Wien, Österreich T: +43 1 3530349 - 10 F: +43 1 3530349 - 99 roman.dissauer(a)rdi.at www.rdi.at

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How to bridge a call to an extension without the display updating?
by Michael Avers 13 Mar '18

13 Mar '18

Hello, I'm trying to create a random dialplan extension (say 5050) that when dialed into by a registered extension would bridge the call to another registered user (say 3000), but not change the display or P-Asserted-Identity. Right now on the bridge it sends to the calling extension *P-Asserted-Identity: "5050" <sip:3000@sip.domain>.* I tried setting all the possible values for sip_cid_type and setting ignore_display_updates to false, but it still sends the P-Asserted update as shown above. The bridge itself I tried both with user/3000(a)sip.domain and ${sofia_contact(566055(a)rsi.solidphone.net)} - same result. This is a problem because A. the display on the phone changes to "3000" and more importantly, B. if the calling user later hits redial it will dial 3000 instead of 5050. Any way to achieve what I'm trying to do? Thanks Mike

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Re: [Freeswitch-users] [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]
by Raimundo Pérez Nieves 26 Feb '18

26 Feb '18

Yes I checked. Somebody told me a great solution, it was start freeswitch going to usr/bin and executing ./freeswitch. That start freeswitch in console mode and shows you every problem. It was just an extension tag unclosed. Thanks anyway Enviado desde mi iPhone El 24/01/2018, a la(s) 18:05, Bilal Abbasi <bilaln018(a)gmail.com> escribió: > Does the freeswitch process started? > check that. > >> On Wed, Jan 24, 2018 at 8:50 PM, Raimundo Pérez Nieves <raimundo.perez.cuba(a)gmail.com> wrote: >> Hi guys, I put shutdown in freeswitch but did't change nothing and when I fs_cli I get this error [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error] >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org >

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Set b-leg CallerID using from in a 300 Multiple Choices reply
by Joseph Waite 12 Feb '18

12 Feb '18

Hi Guys I am trying to use the From field in a 300 Multiple Choices message to set the outbound CallerID in the re-directed b-leg So call comes into FreeSwitch, gets sent out to a SIP Redirect server which replies with a 300 Multiple choices. I need to pull the number from the From field of this 300 Response and use it to set the From field, raid & Paid fields in the re-directed invite. Any help would be very much appreciated. Regards

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mod_lua Too many open files
by António Silva 01 Feb '18

01 Feb '18

hi, After a few weeks running fs without issues, i got the error "Too many open files", i understand that i reach the limit of allowed open files in the system. But it must be trigger for some "zombie fds" opened.. Is there a way to get the "zombie" fds open by fs? I use lua scripts to check db mapping, so in every script i make sure to call dbh:release() before returning. I also use esl sockets to check FS events. in the running process i have the following limits: cat /proc/10975/limits Limit Soft Limit Hard Limit Units Max cpu time unlimited unlimited seconds Max file size unlimited unlimited bytes Max data size unlimited unlimited bytes Max stack size 245760 8388608 bytes Max core file size unlimited unlimited bytes Max resident set unlimited unlimited bytes Max processes unlimited unlimited processes Max open files 999999 999999 files Max locked memory 65536 65536 bytes Max address space unlimited unlimited bytes Max file locks unlimited unlimited locks Max pending signals 128123 128123 signals Max msgqueue size 819200 819200 bytes Max nice priority 0 0 Max realtime priority unlimited unlimited Max realtime timeout unlimited unlimited us currently i see the number of files change: while :; do echo -n "$(date +'%F %T') "; lsof -p $(cat /var/run/freeswitch/freeswitch.pid) | wc -l; sleep 5; done; 2018-01-22 17:20:46 78938 2018-01-22 17:20:53 78920 2018-01-22 17:20:59 78902 2018-01-22 17:21:06 78911 2018-01-22 17:21:12 78883 2018-01-22 17:21:18 78883 2018-01-22 17:21:25 78899 2018-01-22 17:21:31 78913 2018-01-22 17:21:37 78925 2018-01-22 17:21:44 78911 2018-01-22 17:21:50 78913 2018-01-22 17:21:56 78935 I see lot of entries: COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME freeswitc 2429 root *257u 0000 0,11 0 10734 anon_inode freeswitc 2429 root *258u 0000 0,11 0 10734 anon_inode freeswitc 2429 root *260u 0000 0,11 0 10734 anon_inode freeswitc 2429 root *261u 0000 0,11 0 10734 anon_inode freeswitc 2429 root *262u 0000 0,11 0 10734 anon_inode freeswitc 2429 root *264u 0000 0,11 0 10734 anon_inode freeswitc 2429 root *265u 0000 0,11 0 10734 anon_inode freeswitc 2429 root *266u 0000 0,11 0 10734 anon_inode grep "0000" openfiles | wc -l 78951 but I don't understand the output for the TYPE= "0000"... thanks for the help. -- Saludos / Regards / Cumprimentos António Silva

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FreeSWITCH behind NAT
by jungle boogie 01 Feb '18

01 Feb '18

Hi All, I have some questions about this page and what folks do when freeswitch is behind NAT: https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal 1. Do ports need to be opened on the firewall under all circumstances? UDP ports for RTP, for instance. 2. Is this always a good idea to enable in sip_profiles/internal.xml? <param name="nat-options-ping" value="true"/> 3. Is it necessary/recommended to have STUN enabled in vars.xml AND setup the nat-options-ping? 4. my sip_profile/internal.xml has this: <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> Is this an improvement over what's in confluence of: <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> 5. If the endpoints are configured to connect using TCP, does any of this change what's above? thanks!

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Trouble connecting Sangoma Vega 50 BRI to Freeswitch
by badhills 01 Feb '18

01 Feb '18

Hi, i have trouble connecting my Sangoma vega 50 BRI to Freeswitch. I setup the Vega in quick setup, with user and password. Then i setup the gateway in freeswitch. But both sides are getting forbidden and unauthorized messages. In the freeswitch log is a dns failure, but dns seems to working great. I can't find issues there. Unfortunately, I haven't found anything suitable yet and I'm getting desperate. It can't be that hard. If I configure extensions, it works fine and I can make internal calls with 2 or more phones. The connection to the gateway isn't very different, is it? Under Asterisk with Freepbx I already got this to work, but I wanted to try it with FreeSwitch. Had anybody a working connection and could explain it a lil bit to me? kind regards, Markus -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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The next version of FreeSWITCH will be...?
by Nick Giannak III 01 Feb '18

01 Feb '18

Greetings. Last year, there was a lot of talk about FreeSWITCH 1.8, especially given that a new set of prompts from Allison Smith was going to be included with that version. As an audio guy, I'm excited about this by comparison to the prompts we already have, but I digress. However, a lot of that talk seems to have quietened down, and version 1.9 is shown as development on the FreeSWITCH website. And yet, no 1.8 release materialized. So I'm asking for a status report...I'm confused about what's coming next. Please alleviate my confusion? Thanks in advance, Nick

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Deliberately adding delay to all calls
by Erik M. Devane - Comms Guy 31 Jan '18

31 Jan '18

I've been working on using FreeSWITCH as a bridge between different devices, and I need to delay all calls by 2-3 seconds in order for the external systems to keep up. This delay should happen at the start of a call, and all subsequent audio should sound fine, just late. I know that doing this sounds silly - I'm sorry about that. Here are my approaches: 1) Eavesdrop on a call send through delay_echo - this works for SIP clients but I have to do an absolute_codec_string as it seems that CODEC support is limited, and I'm often getting no delay at all with my PortAudio clients. 2) Use lua to do a session freeswitch.msleep on answer - exactly what Confluence tells me to avoid - if I sleep but don't consume audio, I think I can get the desired result. 3) Mess with the jitterbuffer - I haven't had any success with this yet. 4) Increase the buffer in mod_PortAudio. Writing this out makes me sure that I'm missing something - a way of getting delay_echo to work for all calls, or a way to pause session media to get the desired effect. Can you please help?

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(no subject)
by Ali Haider 31 Jan '18

31 Jan '18

The following packages have unmet dependencies: freeswitch-meta-all : Depends: freeswitch-music Depends: freeswitch-sounds Depends: freeswitch-mod-memcache (= 1.6.20~37~987c9b9-1~jessie+1) but it is not going to be installed Depends: freeswitch-mod-kazoo (= 1.6.20~37~987c9b9-1~jessie+1) but it is not going to be installed E: Unable to correct problems, you have held broken packages. where can i find dependes packages

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