March 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Debian package freeswitch-all
by Stanislav Sinyagin 05 Mar '15

05 Mar '15

hi, I just noted that "freeswitch-all" package does not contain speex and opus. And of course "freeswitch-mod-speex" and "freeswitch-mod-opus" conflict with it (I know that "freeswitch-meta-all" would solve the problem, but I like the single-package approach). Is it by intent or was it just missed out? thanks

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freeswitch got killed
by Aqs Younas 05 Mar '15

05 Mar '15

Hi, users. I am playing streams with mod_vlc, but some streams make my switch killed. I am using the lasted git version. FreeSWITCH Version 1.5.15b+git~20150224T205826Z~4909cdb7fb~64bit (git 4909cdb 2015-02-24 20:58:26Z 64bit) Logs that i see are these, also log files is attached. 2015-03-02 05:37:49.554042 [NOTICE] switch_core_session.c:3000 Execute log(${cur}) 2015-03-02 05:37:49.554042 [NOTICE] switch_core_session.c:3000 Execute set(episode=0${last_matching_digits}) 2015-03-02 05:37:49.554042 [NOTICE] switch_core_session.c:3000 Execute curl( http://206.225.05.12/rd_api/api/inboundcampaign/get_extension post ext=${episode}&did=${dst}) 2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute log(${curl_response_data}) 2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute set(error=No) 2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute set(cur=${curl_response_data}) 2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute log(${cur}) 2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute transfer(${cur} XML play) 2015-03-02 05:37:49.594042 [NOTICE] switch_ivr.c:1861 Transfer sofia/external/19546006100@69.27.168.33:5060 to XML[ 95.81.147.3/rfimonde/all/rfimonde-64k.mp3@play] 2015-03-02 05:37:50.094042 [INFO] mod_dialplan_xml.c:635 Processing 19546006100 <19546006100>->95.81.147.3/rfimonde/all/rfimonde-64k.mp3 in context play 2015-03-02 05:37:50.094042 [INFO] switch_ivr_async.c:212 Digit parser DPTOOLS: Setting realm to 'moderator' 2015-03-02 05:37:50.114043 [NOTICE] mod_vlc.c:192 VLC Path is http http://95.81.147.3/rfimonde/all/rfimonde-64k.mp3 [0x25e099b8] access_http access: Raw-audio server found, mp3 demuxer selected [0x7fa3fc2f69a8] mpgatofixed32 audio converter error: libmad error: bad main_data_begin pointer 2015-03-02 05:45:38.494035 [NOTICE] sofia.c:952 Hangup sofia/external/ 18034805839@69.27.168.71:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2015-03-02 05:45:38.534034 [INFO] mod_json_cdr.c:271 Process [f7c5c71a-438b-4c56-938e-34cb19766fd6.cdr.json] 2015-03-02 05:45:38.914042 [NOTICE] switch_core_session.c:1641 Session 2591 (sofia/external/18034805839@69.27.168.71:5060) Ended 2015-03-02 05:45:38.914042 [NOTICE] switch_core_session.c:1645 Close Channel sofia/external/18034805839@69.27.168.71:5060 [CS_DESTROY] [0x7fa47bf55668] Killed What is believe is that freeswitch must not be killed even if stream is bad.

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unable to call in freeswitch IPv6
by Shisheer Teli 05 Mar '15

05 Mar '15

Hi Team, My freeswitch server is on IPv6, and now i am able register extension with IPv6 in freeswitch. but i am unable to call from IPv6 extensions.. can help ..? Regards, shisheer T

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Help with mod xml_curl
by Antony Aguirre Morales 05 Mar '15

05 Mar '15

Hi I have 1 freeswitch xml_curl directory configured and working properly , now I want to add another freeswitch and see the same BD for the 2 fs . The problem I have is that the same management extensions in the 2 freeswitch but they are 2 different ips , how can I do so that you can see the same directory without the problem domain . Anyone have an idea? example fs1 -> domain 1.1.1.1 fs2 -> domain 1.1.1.2 Have in common the extension 1000 but I can now register with a single domain either that of the fs1 and fs2 . regards. r

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Real-time billing application for the FreeSWITCH (mod_lua, mod_perl or ESL)
by Sami Montour 04 Mar '15

04 Mar '15

Hello Everyone, I am a long time user of OpenSIPS and have recently been tasked with developing a call control/real-time billing application for the FreeSWITCH. I would like to develop the application in either C/C++, Perl or lua (totally new to lua) in this order. I have used ESL to develop some basic functionality of the application using Perl. For instance, when the call comes in, I pass over control from FreeSWITCH to the call control application using outbound TCP socket. Then the call control application listens for CHANNEL_ANSWER event to start billing process and CHANNEL_HANGUP event to stop billing. The application does some database work as well to retrieve and check user status and balance and all that stuff. My main concern is with load testing because of the TCP connection between the FreeSWITCH and call control application and also TCP connection to database to retrieve subscriber info. The basic functionality seems to work fine but haven't tested it with high call volume, which I am about to do. The last couple of days I have been experimenting with mod_lua but can't find a method under "Session" to get information related to CHANNEL_ANSWER and CHANNEL_HANGUP events. The "session:geVariable" does not have variables related to session answer time or hangup time, something similar to "$e->getHeader("Caller-Channel-Answered-Time")" in ESL. I must be missing something very simple here. Again, my experience with mod_lua is only a couple of days old. I am new Freeswitch and thought I would seek the advice of all the experts and people with more experience with Freeswitch. For this type of application, am I better off with mod_lua, or mod_perl, or shall I stick with ESL using Perl? How about ESL with C/C++? Will I gain some performance gain there? Any feedback would be appreciated. Thanks. Sam

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Freeswitch and sipML
by Naveen Khanna 04 Mar '15

04 Mar '15

Hi, I work on freeswitch to develop call centre solutions for customers. I am using browser based sipML client & facing the problem of excessive amount of sessions that do not close decently. This probably leads to hanging of my application. Has anyone faced such problem & can someone suggest an effective solution / safeguard. I would prefer using browser because it is effective for pop up applications in call centre environment. Regards, Naveen Khanna

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Copying FreeSWITCH installation
by Oleg Stolyar 04 Mar '15

04 Mar '15

Hi guys, I have a probably silly question. If I have a fully functioning FS installation on (for instance) CentOS. Will it work I simply copy the /usr/loca/freeswitch directory to another CentOS machine with identical setup? Or are there other places in the system (outside of the /usr/local/freeswitch diectory) that FS has files or settings in?

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Incoming Fax problems with bad rows
by Raphael Lechner 04 Mar '15

04 Mar '15

Hi, We have a problem that some incomings fax have many bad rows, from at least 2 different customer and therefore not all lines are readable. We tried first with to remove the ATA Device and changed that, that we receive them by mod_spandsp but the problem still exists. Attached the log file: Sender 1:https://pastebin.freeswitch.org/23960 <https://pastebin.freeswitch.org/23960> Sender 2: https://pastebin.freeswitch.org/23961 <https://pastebin.freeswitch.org/23961> FreeSWITCH version: 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git 507a0f2 2014-12-29 18:59:51Z 64bit) We use a patton ISDN/VoIP Gateway. Any hint how we can resolve/debug that? Thank you, Raphael

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$${sounds_dir} variable
by Richard Mace 04 Mar '15

04 Mar '15

Hi all, Could someone please let me know which file contains the defined location for the variable $${sounds_dir} Thanks Richard

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Re: [Freeswitch-users] ICE SDP Params
by Rob Moore 04 Mar '15

04 Mar '15

Hi Brian, I thought as much, WebRTC isn’t something we are trying to use at the moment (although im sure we’ll find a use for it in the not too distant future.) Invites are created using the bridge application in XML dialplan. I have SIP pcaps but I don’t have and Freeswitch traces at the moment as the issue is only appearing once in say 500 calls on our production system so it can be a little awkward to pin down detailed tracing. I’m working on getting an example today and will post back as soon as possible. Is there any way to disable WebRTC entirely? That could be worth a try whilst I get a test setup for this scenario. From: Brian West <brian(a)freeswitch.org<mailto:brian@freeswitch.org>> Date: Mon, Mar 2, 2015 at 6:15 PM Subject: Re: [Freeswitch-users] ICE SDP Params To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> This is all related to WebRTC, how are you creating the invite? Logs would be helpful./b

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