November 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

segfault when reloading profile
by Juan Pablo L 11 Nov '18

11 Nov '18

Hello, i am getting segfaults when i reload a sip profile, i know it has something to do with the profile itself because when i use the default vanilla configuration and i reload a profile it does not segfault, i have checked every single option in the profile file looking for errors or anything and i find nothing i can fix or change, i m using: FS: FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git a98a958 2018-09-26 17:55:25Z 64bit) linux: Linux sbc02 4.9.0-8-amd64 #1 SMP Debian 4.9.110-3+deb9u6 (2018-10-08) x86_64 GNU/Linux when analysing the core file this is what i get in gdb: Core was generated by `freeswitch -nc'. Program terminated with signal SIGSEGV, Segmentation fault. #0 switch_ci_hashfunc_default (char_key=0x0, klen=klen@entry=0x7f7249e229c8) at src/switch_apr.c:120 120 for (p = key; *p; p++) { [Current thread is 1 (Thread 0x7f7249e23700 (LWP 3029))] (gdb) bt #0 switch_ci_hashfunc_default (char_key=0x0, klen=klen@entry=0x7f7249e229c8) at src/switch_apr.c:120 #1 0x00007f720a5effd6 in xml_directory_db_search (section=<optimized out>, tag_name=<optimized out>, key_name=<optimized out>, key_value=<optimized out>, event_params=0x7f722400ae80, user_data=<optimized out>) at xml_directory.c:152 #2 0x00007f72538973eb in switch_xml_locate (section=section@entry=0x7f724a0570e0 "directory", tag_name=tag_name@entry=0x0, key_name=key_name@entry=0x0, key_value=key_value@entry=0x0, root=root@entry=0x7f7249e22bd0, node=node@entry=0x7f7249e22bd8, params=0x7f722400ae80, clone=SWITCH_FALSE) at src/switch_xml.c:1695 #3 0x00007f7249f40f92 in launch_sofia_worker_thread (profile=profile@entry=0x7f71f401a160) at sofia.c:3062 #4 0x00007f7249f4ac19 in sofia_profile_thread_run (thread=<optimized out>, obj=0x7f71f401a160) at sofia.c:3385 #5 0x00007f7253b06e07 in dummy_worker (opaque=0x7f71f401c258) at threadproc/unix/thread.c:151 #6 0x00007f7252615494 in start_thread (arg=0x7f7249e23700) at pthread_create.c:333 #7 0x00007f7252357acf in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:97 any help or direction would be appreciated ... thank!

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TESTING
by Andrey Wolk 11 Nov '18

11 Nov '18

Test message.

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Re: [Freeswitch-users] Multiple Diversion headers
by Sjoerd Boomstra 09 Nov '18

09 Nov '18

Hi, Yes I’m interested in your patch; I’d like to be able to send multiple headers (Diversion in my case). For receiving there’s already a solution with the param parse-all-invite-headers=true. Best regards, Sjoerd. Van: Vallimamod Abdullah - SIP Solutions [mailto:vma@sip.solutions] Verzonden: woensdag 22 augustus 2018 19:02 Aan: Sjoerd Boomstra Onderwerp: Re: [Freeswitch-users] Multiple Diversion headers Hi, I have faced the same issue when passing certification tests with Orange-FR. There is a bug in freeswitch that makes it impossible to manage multiple headers with same name, either receiving or sending; the last value overwrites all the previous ones. I wrote a patch to correct it. I didn't get the chance to publish it yet, but if you are interested I can share it with you. Btw, I provide freeswitch consulting services for carriers. If you have any need, feel free to ping me. Best Regards, -- Vallimamod Abdullah SIP Solutions vma(a)sip.solutions<mailto:vma@sip.solutions> +33 6 62 60 68 97 linkedin.com/in/vallimamod<http://linkedin.com/in/vallimamod> . On 22 Aug 2018, at 16:10, Sjoerd Boomstra <S.Boomstra(a)telecats.nl<mailto:S.Boomstra@telecats.nl>> wrote: Hi All, I’m struggling with multiple Diversion headers. In our case we have an incoming call with multiple diversion headers, we do some processing and want to add a diversion header ourselves to the outgoing call. What I’ve found so far is that for inbound calls one needs to set the profile param parse-all-invite-headers=true, this results in a sip_i_Diversion variable that uses the ARRAY:: syntax and has all the values I want. So far so good. Example: ARRAY::<sip:Carol@uas2.isp.com>;privacy=off;reason=unconditional;counter=2;screen=no|:<sip:Bert@uas2.isp.com>;privacy="full";reason=user-busy;counter=3;screen=no|:<sip:Bob@uas1.isp.com>;reason=do-not-disturb;counter=1 For the outbound call it’s more difficult, I cannot get FreeSwitch to produce multiple Diversion headers. The only syntax I’ve found is using comma’s in the value, like this: <action application="export" data="sip_h_Diversion=<sip:Carol@uas2.isp.com>;privacy=off;reason=unconditional;counter=2;screen=no,<sip:Bert@uas2.isp.com>;privacy="full";reason=user-busy;counter=3;screen=no,<sip:Bob@uas1.isp.com>;reason=do-not-disturb;counter=1,<sip:Bert@uas2.isp.com>;reason=user-busy"/> This results in a single diversion header with the contents I put in. From the RFC’s I understand that this is technically equivalent to 4 (in my case) Diversion headers, is this true? Is there a way I can convince FS to generate multiple separate diversion headers? Best regards, Sjoerd. BTW I’ve tested with FS 1.6.20 and 1.4.26

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H.323 Connection
by satoh.osamu＠nttdata-sbc.co.jp 09 Nov '18

09 Nov '18

Hello, I am trying to connect to FreeSWITCH by H.323 protcol. I built,installed and loaded the module of mod_h323. but it not work properly. I installed it as follows. (OS, Version info) OS: CentOS Linux release 7.5.1804 (Core) FreeSWITCH: 1.6.2 PTlib: 2.10.9.2 h323plus: 1.27.0 Configure was reference to the following. https://freeswitch.org/confluence/display/FREESWITCH/mod_h323 It is in the following state.(When Call to SIP Extension) - It seems to connection is successfly(Connection established) on Client, but it can not hear any sound. - In Server of FreeSWITCH, the CPU load abnormally increases, and there is the case that service of Freeswith is stopped. I looking for logs of past mailing lists, there are cases of success and failure. http://lists.freeswitch.org/pipermail/freeswitch-users/2016-September/12256… <http://lists.freeswitch.org/pipermail/freeswitch-users/2016-June/120918.html>http://lists.freeswitch.org/pipermail/freeswitch-users/2016-June/120896.html Can you tell me what settings and installation procedures can be successfully connected? Best Regards, Satoh

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One way audio case between FS and Chrome on WebRTC.
by MARAND, Remi 08 Nov '18

08 Nov '18

Hello, I am trying to validate FS as a SIP to WebRTC Gateway in our lab environment. I started in middle October and cannot understand this One Way Audio trouble. I must thank the Freeswitch team and contributors for this very impressive work. FreeSWITCH Version 1.8.1-2-4f54cff36a~64bit (-2-4f54cff36a 64bit) On system: SMP Debian 4.9.110-3+deb9u6 (2018-10-08) x86_64 GNU/Linux Openssl version : OpenSSL 1.1.0f 25 May 2017 Chrome version: 69.0 (I tried with different version and with Firefox with the same trouble). The wss part is ok with sip.js and verto.js The Ice negotiation is ok, I use sometimes local networks and sometimes web, I have had to authorize networks in the candidate ACL and domain ACL (acl.conf.xml) The result is the same on both topology. DTLS negotiation is OK, and there is UDP streams between Chrome (or Firefox) and FS in both ways. There is no audio in the direction from FS to Chrome, the other direction is OK. The simplest test is to call the 5000 number from the Chrome client, I send you a paste bin and pcap trace for this call. Should you give me information element to progress on this, what is really mandatory in the sip_profile/internal.xml and external.xml files, and in directory/default/1000.xml for a WebRTC call ?? What should be the good options in fs_cli to see if the encryption of RTP packets is ok or not.. ? Do you think that I have to reinstall a Freeswitch from the current branch ? @IP for FS : 192.168.145.67 @IP for Chrome : 10.70.54.43 Link on the pastebin : https://pastebin.freeswitch.org/view/09a72087 The pcap trace joined is done on Freeswitch on the same call. Thank you for helping me !! Best regards. [Prosodie-signature]<http://www.prosodie.com/> Rémi Marand - Product Owner - Pod Connect. PROSODIE - Marketing & Produit Tél. : +33 (0)1.46.84.12.77 / 06.87.72.53.25 rmarand(a)prosodie.com<mailto:rmarand@prosodie.com> This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

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Memory increase because of multiple instances of freeswitch
by António Silva 08 Nov '18

08 Nov '18

Hi all, I notice a strange behaviour on machine due to increase of memory, when i went to see that was the process consuming the memory i notice that freeswitch have multiple process running: root 2543 90.4 4.8 2178716 1587484 ? S<sl Sep27 36619:46 /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs root 7858 0.0 5.0 2289392 1649484 ? SN Oct19 0:00 \_ /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs root 30626 0.0 4.4 2172632 1464684 ? SN Oct23 0:00 \_ /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs root 4505 0.0 4.9 2336544 1621768 ? SN 10:43 0:00 \_ /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs root 22557 0.0 4.9 2336548 1636604 ? SN 11:02 0:00 \_ /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs and today: root 21823 93.9 4.8 2154128 1581064 ? S<sl Oct26 17781:45 /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs root 18417 0.0 4.9 2247264 1610108 ? SN 11:09 0:00 \_ /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs what could be the cause of this? anyone experience the same? As for the service everything is running ok, is just that the memory keeps increasing.. i restart freewitch when i start to reach my memory limit. I'm running fs 1.8.2 on debian jessie. -- Saludos / Regards / Cumprimentos António Silva

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mod_verto and verto communicator on Debian 9 -> WS SETUP FAILED
by Alexander Haugg 08 Nov '18

08 Nov '18

Hi, I try to setup the mod_verto – verto communicator scenario. The Certificate is OK “https://freeswitch.org/confluence/display/FREESWITCH/WebRTC#WebRTC-InstallCertificates” But for the sockets IP:8081 and IP:8082 I get the error message “WS SETUP FAILED” The question… Why the ws setup fails? vhosts is not configured, I am using the mini_httpd as http server. Thanks a lot Alex

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mod_fail2ban links
by Guillermo Ruiz Camauer 07 Nov '18

07 Nov '18

I am trying to install mod_fail2ban but both of the links that appear on Confluence are not working. Does anyone have working links? Has this module been abandoned? -- Guillermo Ruiz Camauer

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How to handle acceptance or rejection for SIP SUBSCRIBE requests (subscriptions)
by Florent Krieg 05 Nov '18

05 Nov '18

Hello everybody, How can I handle acceptance/rejection for subscribe requests? The idea is to apply 'rights' on users, and make for instance user1 able to subscribe to user2's presence (eg a colleague) but unable to subscribe to user3's presence (eg the company boss)? I currently have two ideas of doing so, but none fully satisfies me because they're dirty tricks and not built-in. My ideas: 1/ handle user directory with xml_curl => when sending the request with sip_auth_method=SUBSCRIBE I check From/To and the rights: if allowed I reply the user profile, otherwise I reply "not found" (erf! what sip-level trick...) 2/ write an external script that subscribe to freeswitch events related to presence, accept any subscription, after each successful subscription the script checks the database entry: if allowed don't touch anything, if not allowed then delete the entry from the db (super-erf! db-level trick) Would there be other solutions that I would have missed? I mean, proper and clean solutions. Thanks a lot for your help :) Florent

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DTMF Issue - Unexpected digits are receiving
by Chandramouli P 05 Nov '18

05 Nov '18

Hello, We developed an application (FreeSwitch) that receives pressed digit by the end user on mobile phone dialer through DTMF and some action will be performed. Everything is working fine till now. But, we noticed one issue recently. From the FreeSwitch log, we came to know that FreeSwitch is received the letter i.e. "A" from DTMF, instead of digit. We tried to regenerate this issue. But, when we are trying to regenerate, FreeSwitch is receiving the digit from DTMF as usual. I would like to know How FreeSwitch is received the letter i.e. "A", instead of digit. Which key was pressed by the end user on mobile phone dialer? Thank you. Best Regards, Chandramouli.

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