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codec changed
by amani mansour 31 Mar '16

31 Mar '16

Hi, can i please play back a .pcap file ? because when i converted this file .pcap to new file .wav the codec is changed ,can you help me please ?? best regards amani

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Re: [Freeswitch-users] Memory allocation "switch_core_session_alloc"
by Ahmed habiba 31 Mar '16

31 Mar '16

Mainly I’m not writing a new module, I’m modifying mod_spy to allow spying current active called without providing uuid, mainly I did the below: 1-Creat new hash map to contain all bridged calls(userid@domain as key, and UUID as value). 2-when I receive any spy request I query the hash map, using userid@domain, if I find a uuid and add it as if I get it in command “userspy userid@domain uuid “ 3-then I continue. The point that I try to allocate the memory on session level and once the call get hangup I remove it from the hash map, however this doesn’t work as I expect only one hangup request comes and accordingly one entry from hasp being removed. I attached here the code for your review From: Anthony Minessale <anthony.minessale(a)gmail.com <mailto:anthony.minessale@gmail.com>> Subject: Re: [Freeswitch-users] Memory allocation "switch_core_session_alloc" Date: March 31, 2016 at 1:51:48 AM GMT+3 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Correct. The memory is allocated from the session's pool and will not be valid when the call ends. For persistent memory allocation, create your own pool when the module loads, (or use the one passed to you in the init function) and use switch_core_alloc instead. What module are you writing? On Wed, Mar 30, 2016 at 4:48 PM, Ahmed habiba <ahabiba(a)gmail.com <mailto:ahabiba@gmail.com>> wrote: Hi, I’m trying to use “switch_core_session_alloc” in a new module however I notice in documentation the below note, does this mean that once the session hangup this memory segment will not be valid anymore and I should not use this segment again?, your kind feedback will be appreciated. "Note: the memory will be in scope as long as the session exists" Thanks, Ahmed Habiba. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ <http://freeswitch.org/> ☞ http://cluecon.com/ <http://cluecon.com/> ☞ http://twitter.com/FreeSWITCH <http://twitter.com/FreeSWITCH> ☞ irc.freenode.net <http://irc.freenode.net/> #freeswitch ☞ http://freeswitch.org/g+ <http://freeswitch.org/g+> ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s <https://www.youtube.com/watch?v=9XXgW34t40s> https://www.youtube.com/watch?v=NLaDpGQuZDA <https://www.youtube.com/watch?v=NLaDpGQuZDA>

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Make current problem
by Miha 31 Mar '16

31 Mar '16

Hi trying to upgrade on fly my 1.5 version of FS but getting this: make[1]: *** [libs/libvpx/Makefile] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 You have mail in /var/spool/mail/root Is here any fix solution? tnx miha

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Master and slave warm standby in production
by Dmitry Mordovin 31 Mar '16

31 Mar '16

Dear friends! Anybody use Postgres 9.5 warm standby in production? Which schemes you use? My setup: - 2 DB servers (master and slave) - 4 APP servers (tomcat) Each tomcat server have JDBC connect to master and prepared config for slave. Business task permit switch between DB server on APP servers manually when failover. What is the best scheme in my configuration? BR, Dmitry

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[Mute A-Party]
by Bilal Abbasi 31 Mar '16

31 Mar '16

Hi Users, Do Freeswitch has a setting to mute the A-Party, I only want one way audio (yes many had problems with this..:-P). Any suggestions. Regards Abbasi

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Memory allocation "switch_core_session_alloc"
by Ahmed habiba 30 Mar '16

30 Mar '16

Hi, I’m trying to use “switch_core_session_alloc” in a new module however I notice in documentation the below note, does this mean that once the session hangup this memory segment will not be valid anymore and I should not use this segment again?, your kind feedback will be appreciated. "Note: the memory will be in scope as long as the session exists" Thanks, Ahmed Habiba.

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Re: [Freeswitch-users] Convert pcap file to wav file
by amani mansour 30 Mar '16

30 Mar '16

good night sir , i replaced R by Y i have this problem [image: Images intégrées 1] the file contains rtp named g711.pcap ,you will find it attached and the script to execute to convert this pcap to wav is : ssrc=$(tshark -n -r /home/amani/projet/PFE/g711.pcap -Y rtp -T fields -e rtp.ssrc -Eseparator=, | sort -u | awk 'FNR ==1 {print}') sudo tshark -n -r /home/amani/projet/PFE/g711.pcap -Y rtp -Y "rtp.ssrc == $ssrc" -T fields -e rtp.payload | tee payloads for payload in `cat payloads`; do IFS=:; for byte in $payload; do printf "\\x$byte" >> /home/amani/projet/PFE/g711.raw; done; done sox -t raw -r 8000 -v 4 -c 1 -e mu-law /home/amani/projet/PFE/g711.raw /home/amani/projet/PFE/g711.wav can you please sir execute it ? and tell me where is the problem ? thank you best regards 2016-03-30 22:16 GMT+01:00 amani mansour <amani.mansour2(a)gmail.com>: > good night sir , > i replaced R by Y > i have this problem > [image: Images intégrées 1] > > the file contains rtp named g711.pcap ,you will find it attached > > and the script to execute to convert this pcap to wav is : > ssrc=$(tshark -n -r /home/amani/projet/PFE/g711.pcap -Y rtp -T fields -e > rtp.ssrc -Eseparator=, | sort -u | awk 'FNR ==1 {print}') > sudo tshark -n -r /home/amani/projet/PFE/g711.pcap -Y rtp -Y "rtp.ssrc == > $ssrc" -T fields -e rtp.payload | tee payloads > for payload in `cat payloads`; do IFS=:; for byte in $payload; do printf > "\\x$byte" >> /home/amani/projet/PFE/g711.raw; done; done > sox -t raw -r 8000 -v 4 -c 1 -e mu-law /home/amani/projet/PFE/g711.raw > /home/amani/projet/PFE/g711.wav > > > can you please sir execute it ? and tell me where is the problem ? > thank you > best regards > amani >

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Convert pcap file to wav file
by amani mansour 30 Mar '16

30 Mar '16

Hello , Please i need to convert pcap file to wav file with comman linux in freeswitch to play back this file .wav i found this : 1 step : write a script to convert pcap to wav wich is : ssrc=$(tshark -n -r /home/amani/projet/PFE/test.pcap -R rtp -T fields -e rtp.ssrc -Eseparator=, | sort -u | awk 'FNR ==1 {print}') echo $ssrc sudo tshark -n -r /home/amani/projet/PFE/test.pcap -R rtp -R "rtp.ssrc == $ssrc" -T fields -e rtp.payload | tee payloads for payload in `cat payloads`; do IFS=:; for byte in $payload; do printf "\\x$byte" >> /home/amani/projet/PFE/test.raw; done; done echo ‘sox has converted pcap to wav file’ sox -t raw -r 8000 -v 4 -c 1 -e mu-law /home/amani/projet/PFE/test.raw /home/amani/projet/PFE/test.wav ==> but when i run this script the error is : tshark: -R without -2 is deprecated. For single-pass filtering use -Y. ‘sox has converted pcap to wav file’ sox FAIL formats: can't open input file `/home/amani/projet/PFE/test.raw': No such file or directory Rq: paylods file is empty :( please can you help me ??? thanks with regards amani

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mod_erlang_event outbound timeout in FreeSWITCH 1.6
by Nathan Hartwell 30 Mar '16

30 Mar '16

Hi all, I've been using mod_erlang_event outbound mode with FS 1.4 for ~1 year with no issues but am unable to get it to work on 1.6 or master (on Debian Jessie, installed from apt). When mod_erlang_event attempts to make an outbound connection to my Erlang node it errors with a timeout (relevant logs below). Inbound mode continues to work fine. I am preparing a more detailed bug report with test scripts, etc. but wanted to quickly check whether anyone else on the list has faced similar issues or has suggestions on other things to try. Thanks! Nathan Hartwell Logs: EXECUTE sofia/internal/1001(a)192.168.33.10 erlang(yacc_inbound:launch yacc-dev@localhost) 2016-03-29 23:27:21.745778 [DEBUG] mod_erlang_event.c:1665 enter erlang_outbound_function yacc_inbound:launch yacc-dev@localhost 2016-03-29 23:27:21.745778 [DEBUG] mod_erlang_event.c:1684 Using existing listener for session 2016-03-29 23:27:21.745778 [DEBUG] mod_erlang_event.c:1696 Creating new spawned session for listener 2016-03-29 23:27:21.745778 [DEBUG] mod_erlang_event.c:1540 Added session to listener 2016-03-29 23:27:21.745778 [DEBUG] mod_erlang_event.c:1563 rpc call: yacc_inbound:launch(Ref) 2016-03-29 23:27:21.745778 [DEBUG] mod_erlang_event.c:1575 Waiting for reply 9.0.0@freeswitch@localhost 3eb319cd-883c-4918-9d42-395e7683585b 2016-03-29 23:27:26.765708 [WARNING] mod_erlang_event.c:1581 Timed out when waiting for outbound pid 9.0.0@freeswitch@localhost 3eb319cd-883c-4918-9d42-395e7683585b Dialplan: <extension name="yacc_queue"> <condition field="destination_number" expression="^5007$"> <action application="set" data="yacc_queue=advisor"/> <action application="erlang" data="yacc_inbound:launch yacc-dev@localhost"/> </condition> </extension>

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Routing outbound calls to user extension
by Murat Knecht 30 Mar '16

30 Mar '16

Hello, I want to route outbound calls to my user-extension-connected softphone, and would like to know which mechanism to use. The IVR system I develop makes calls and then reacts to keypresses (reverse DTMF). For testing purposes, I'd like to route those outbound calls to my softphone — the advantage of that is that the IVR system can remain unchanged, calling numbers in the US or Philippines, while I still test with my local softphone. Regarding my setup, I have a FS 1.2.24 with two softphones connected as user extensions 1001 and 1002. Calling between those phones works, and I can originate calls to them, with user/1001 for example. Then, I have an external gateway “lan” and a dialplan setup, which I want to use to originate calls from the IVR (those are just the default configs, as they should not matter, except the bridge / routing config inside). If I do a call to a Philippines number: originate sofia/external/lan/639123298392 &echo() Currently, this gives me: [WARNING] mod_sofia.c:5089 Cannot locate registered user lan/63282932@external I don't understand why an *external* number would need to or could be registered. To address my original requirement, I tried adding an unconditional bridge to user/1001 to the dialplan, which is otherwise empty. <extension name="lan_pstn"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="user/1001"/> </condition> </extension> This has no effect. (Still the above error.) If I add a hostname to the number, an INVITE is sent, and then I get a different error: originate sofia/external/lan/1001@hostname &echo() [NOTICE] sofia.c:6702 Hangup sofia/external/lan/1001@hostname [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] My question is: How can I configure this gateway that all outbound calls are accepted (nobody needs to be registered) and are then routed to a user extension, i.e. never leave the “local setup”? Thank you, Murat P.S.: Needed to resubscribe, sorry if this ends up duplicate.

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