March 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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concurrent calls capacity planning
by Niranjan kl 30 Mar '16

30 Mar '16

We started using freeswitch for one of our products. We use esl to control calls. I am trying to come up with numbers to do capacity planning. I ran my tests on 8 core cpu 16 GB Ram server. This is on ubuntu 14.04 linux (kernel 3.13) distro on amazon web services. I used sipp to do load tests. I used freeswitch's external profile (single profile only) as I will have to deal with third party SIP trunk providers and G711 is the codec I used. When I executed 500 concurrent calls by creating 500 calls at the same time, things work fine. I see linux's load staying close to 3. When I do 1000 concurrent calls by creating 1000 calls at the same time, linux's load shoots to 75 and sometimes reaching 100. Earlier, I did the same test on 4 core 8 GB server, load shot up to 75 for 500 concurrent calls. When I searched for freeswitch installation on aws, I read that timer configuration and low latency kernel changes are recommended. But It also says it is not needed for 3.x kernel. So, Is this needed to extract good performance from hardware on aws ? Instead of adding hardware blindly, I would like to figure out is there any other way I can profile this and understand the relationship between hardware and concurrent calls number. Thanks, Niranjan KL

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play back wireshark parol
by amani mansour 30 Mar '16

30 Mar '16

Hi, I have a wireshark file i want to play the parol in this file like voicemail like for example : <action application="playback" date {loops=1}tone_stream://path=${conf_dir}/tetris.ttml"/> this example of tetris music with regards amani

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Re: [Freeswitch-users] Ringing don't exist
by Ahmed habiba 30 Mar '16

30 Mar '16

The only way you will get 180/183 is that there is an endpoint that take time/ or ring on the other side, but what I see it is like an Internal IVR calling with immediate response. my question why do you need to see 180/183, if so you can defined another soft phone and make it rings, what is you business case. From: amani mansour <amani.mansour2(a)gmail.com <mailto:amani.mansour2@gmail.com>> Subject: Re: [Freeswitch-users] Ringing don't exist Date: March 30, 2016 at 10:59:50 AM GMT+3 To: Ahmed habiba <ahabiba(a)gmail.com <mailto:ahabiba@gmail.com>>, 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>> Can anyone assist me please ? Thank you. 2016-03-30 8:02 GMT+01:00 amani mansour <amani.mansour2(a)gmail.com <mailto:amani.mansour2@gmail.com>>: Good morning Mr Ahmed , Thank you sir , i modifie the ring back but no change in the tcpdumps but I think (probably sure) that the problem is in the bridge : i don't know what i must set in data = user/ ... or sofia/gateway/... remarque that the extension 600 is registred when i added an external gateway like this : sip-sofia/external <include> <gateway name="gateway_to_600.xml"> <param name="proxy" value="192.168.3.5"/> <param name="username" value="600"/> <param name="password" value="$${default_password}"/> <param name="register" value="true"/> </gateway> </include> 2016-03-30 6:44 GMT+01:00 Ahmed habiba <ahabiba(a)gmail.com <mailto:ahabiba@gmail.com>>: Hi, the configuration you did will not generate any 180/183, as per your configuration you answer immediately once you get invite, if you are trying something like ring-back-tone you can try the below, this will allow some music to play while the other phone is ringing <action application="set" data="ringback=[data]"/> Thanks, Ahmed Habiba. From: amani mansour <amani.mansour2(a)gmail.com <mailto:amani.mansour2@gmail.com>> Subject: Re: [Freeswitch-users] Ringing don't exist Date: March 30, 2016 at 12:12:43 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>> Hi Mr Bote the problem is not in the number look when i do in fs_cli : sofia status internal reg ==> i have 2 numbers registred : 1000 and 1011 but only 1000 is configured by softphone 1011 : i did it using external gateway when i dial 1011 from the softphene where registred the number 1000 ,, so the number 1000 is registred in the FS NOT IN THE SOFT PHONE and now i want to call the number 1000 only when it ring (180) or the method progress sesiion arrive i listen a short music example titris music now i receive the all methodes done well also i listen the music but i dont receive the method nor 180 nor 183 can you help me please with regards amani 2016-03-29 21:56 GMT+01:00 Bote Man <bote_radio(a)botecomm.com <mailto:bote_radio@botecomm.com>>: Any device or soft phone that you want to register to FS should have its own entry in conf/directory/default/*.xml This contains its authentication ID and password to register it to FreeSWITCH. Then you need an entry in conf/dialplan/default.xml or other dialplan that tells FS how to route that number. The Vanilla configuration files available for FS come with directory files for users 1000.xml through 1019.xml and a default dialplan entry that calls 1000-1019. So it will not know how to route 600 nor 423 until you add routes for those numbers in the dialplan. Or you could change 423 to 1001 and enter its credentials into 1001.xml and it should call both directions. --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence <http://freeswitch.org/confluence> From: amani mansour Sent: Tuesday, 29 March, 2016 16:04 Subject: Re: [Freeswitch-users] Ringing don't exist Mr i have : 1 FS A(virtual machine 1)--------------------------------------------->in my conputer i installed a soft phone ( number 1000 is regestred from the soft phone ) i did a gateway which permet to register another number 423 in freeswitch A so now i have 2 numbers registred in FS but only one is regestred by the soft phone i want to dial 423 from the soft phone 1000 normally the call will be done normally ) but i don't have the methode ring i wish that i have well describe the problem t thank you sir ,i wish you can help me best regards amani 2016-03-29 17:58 GMT+01:00 Michael Jerris <mike(a)jerris.com <mailto:mike@jerris.com>>: What device is not sending the 180? why is it not sending it? On Mar 29, 2016, at 12:19 PM, amani mansour <amani.mansour2(a)gmail.com <mailto:amani.mansour2@gmail.com>> wrote: yes i know mr Brian ,in fact i asked why ? 2016-03-29 17:14 GMT+01:00 Brian West <brian(a)freeswitch.org <mailto:brian@freeswitch.org>>: There is no 180/183 in there, just a 200OK /b On Tue, Mar 29, 2016 at 10:59 AM, amani mansour <amani.mansour2(a)gmail.com <mailto:amani.mansour2@gmail.com>> wrote: Hi, I need to make a call from my soft phone to another freeswitch ,when the call is established (i meant (INVITE ,OK , RINGING or Progress.dession .) and after the ring is done to the freeswitch i will listen for example a song ,for that i make an external gateway ,but the step "ringing or progress session " is not estableshed i need to call the extension 600 <extension 600.PNG> <tcpdump.PNG> thanks with regards amani _________________________________________________________________________ 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/> _______________________________________________ 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/>

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Verto client on different server than freeswitch
by Rutu Patel 30 Mar '16

30 Mar '16

Hi, We are using 2 freeswitch servers with mod_verto implemented. We have web client verto demo on different server than freeswitch server. For sofia registrations we use shared database using odbc connection. Is it possible to use shared database concept with verto registration or is there any other way to achieve this ? -- Thanks, Rutu Patel <http://www.inextrix.com>

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Attended Transfer with one-way audio
by Alex Pierry 30 Mar '16

30 Mar '16

Hi All, I'm facing a strange behaviour in FS after a Attended Transfer. For the A leg listen the C leg, C needs to press some DTMF key. Otherwise only C leg can listen A leg. My scenario is: A (external) ---> TDM/SIP Gateway ---> FS ---> B (extension) ---> Attended Transfer ---> C (extension). Both extensions are SIP Phones registered in FS. If the transfer is Blind, the audio works fine. Other test: If the A leg is also a local extension, the audio on Attended Transfer works fine too. An important point is that all transfers are made pressing "Flash" key in SIP Phone not by att_xfer in Dialplan. So, any clue to start? Thank you, Alex. <http://pt.slideshare.net/marketingteclan> <http://blog.teclan.com.br/>

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json_cdr url separator
by Alexandru Covalschi 30 Mar '16

30 Mar '16

How can I send json to multiple servers simultaneously using json_cdr? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator tel: +37367398493 web: http://abriss.solutions/ <http://abs-telecom.com/>

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