October 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Early Media,Hangup issue in WebRTC
by Murugan Pandian 14 Oct '16

14 Oct '16

Hi, I am using SIPJS for my webrtc client ,When i try to make call pstn, i cant able to hear any early media and if b leg (pstn) disconnect call ,A-leg(sipjs) did't receive bye message. if i use this same scenario in my voipfone(xlite.zoiper) works fine. Regard's Murugan Pandian

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Changing sound file suffixes
by Valter Nogueira 14 Oct '16

14 Oct '16

Is possible change sound file suffixes on FS? I would like to share files that I already use in Asterisk with my FS, like .alaw, instead of .PCMA. Regards, Valter

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Re: [Freeswitch-users] FreeSwitch General Info
by Ahmed Munir 14 Oct '16

14 Oct '16

Thanks Mike and Anthony for you inputs. From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 13 Oct 2016 11:57:04 -0500 > Subject: Re: [Freeswitch-users] FreeSwitch General Info > > > On Thu, Oct 13, 2016 at 11:41 AM, Michael Jerris <mike(a)jerris.com> wrote: > >> >> On Oct 13, 2016, at 12:31 PM, Ahmed Munir <ahmedmunir007(a)gmail.com> >> wrote: >> >> Few things I forgot to ask, the scale-ability I was referring to as >> horizontal scale and thanks for your input on that. >> >> For point#2, mod_curl you mentioned for realtime, will it sustain more >> than 500-1K CPS and will it stress out the httpd/apache service? >> >> >> depends on the box. Making the requests and parsing them has overhead, >> sure. As I said, you need to test on your target hardware with your call >> flows to see for sure how it will size. 1kcps on one box is probably high >> to the point of having issues on single box. Doing so across multiple >> boxes would be advised, i don’t like to put that many calls onto a single >> point of failure. Will it stress the apache service…. that would be a >> factor of what you have to do over there, and 100% dependent on that code >> answering the request, its impossible for me to answer that question >> generically. >> > > > xml_curl is a default module implementing a gateway to retrieve config > from a black box http service. It will only perform as good as that http > service which itself could be a cluster/caching etc. > You can also create custom modules to connect any way you wish into config > and dialplan lookup etc. (I helped write Asterisk realtime, and at the > time it was a bolt on afterthought, FS is lucky that it was designed from > the beginning with the knowledge gained from Asterisk experience) > We have several other abstractions for integrating with external systems > but there are too many to discuss here. > > > > >> >> For point#3, you mean to say Freeswitch doesn't use system files for >> media like Asterisk does? >> >> >> Sure we can use system files for media. You were asking about additional >> sockets, we have media sockets, if you are opening files of course that >> would use file handles as well. >> > > > Sockets are used for RTP, HTTP/WSS, Media Files, etc. The use of many > sockets does not play a role in the performance of software assuming you > tune the kernel to match the load needed. > Most socket related instability comes from the use of the select syscall > which is limited FD numbers below 1024 > > > >> >> For point#4, multi-threading; I forgot to mentioned whether Freeswitch >> supports multi-processing? As Asterisk, it uses/runs on single process >> whereas OpenSIPs/Kamilio, we can set multiple child processes. >> >> >> We are not forking. You could manually launch multiple processes if you >> choose. >> > > > In Linux, there is no longer a difference between threads and processes, > they are identical. (See "man clone") http://man7.org/linux/man-pa > ges/man2/clone.2.html > > > >From "man fork" > > *C library/kernel differences* > Since version 2.3.3, rather than invoking the kernel's *fork*() system > call, the glibc *fork*() wrapper that is provided as part of the NPTL > threading implementation invokes clone(2) <http://man7.org/linux/man-pages/man2/clone.2.html> with flags that provide the > same effect as the traditional system call. (A call to *fork*() is > equivalent to a call to clone(2) <http://man7.org/linux/man-pages/man2/clone.2.html> specifying *flags* as just *SIGCHLD*.) > The glibc wrapper invokes any fork handlers that have been > established using *pthread_atfork*(3). > > > > > > > > >> >> >> From: Michael Jerris <mike(a)jerris.com> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Thu, 13 Oct 2016 10:04:56 -0400 >>> Subject: Re: [Freeswitch-users] FreeSwitch General Info >>> >>> > On Oct 13, 2016, at 9:52 AM, Ahmed Munir <ahmedmunir007(a)gmail.com> >>> wrote: >>> > >>> > Hi, >>> > >>> > Currently I'm browsing different platforms online for open source VoIP >>> technologies as currently using Asterisk as PBX. >>> > >>> > I would like to know following things about FreeSwitch; >>> > >>> > 1- FreeSwitch whether supports realtime integration sip >>> profiles/extensions (using DB) and dialplan? (Just like in Asterisk) >>> >>> yes, most frequently using mod_xml_curl which uses http to pull xml on a >>> per dialplan/directory lookup basis >>> >>> > 2- On average, how many extensions and active calls (CPS) it can >>> sustain? >>> >>> This widely varies based on what you are doing. Calls with no media >>> passed through the switch would be measured much more in calls per second >>> than total calls, and could be in the thousands, calls with heavy video >>> transcoding would be in the double digits on a big box. What you are doing >>> could lie anywhere on that spectrum. Our recommendation is always to test >>> your scenario and figure out sizing for yourself. Typically people with >>> heavy load are very satisfied with our performance.. >>> >>> > 3- For a call channel, will it create or use system sockets >>> (openfile)? As you know that Asterisk create/open and consume the system >>> sockets for each call. >>> >>> Certainly we use sockets for media. We do not use them in the way >>> asterisk does. >>> >>> > 4- Will it supports single thread or multithreading? >>> >>> FreeSWITCH is heavily multithreaded. Each call will use at least 2 >>> threads, one for each call leg. >>> >>> >>> > 5- Is it scale-able in term of setting up as a cluster(s)? >>> >>> Sip itself allows for this. Depends on your exact scenario how to >>> architect in a way to make this scale horizontally, but it can be done in >>> most scenarios >>> >>> > >>> > I'll be glad to hear it from you. >>> > >>> > -- >>> > Regards, >>> > >>> > Ahmed Munir Chohan >>> >> >> >> _________________________________________________________________________ >> 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 >> > > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > <http://freeswitch.org/g+>* > > ClueCon Weekly Development Call > ☎ sip:888@conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > > -- Regards, Ahmed Munir Chohan

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Bridging between two rfc1918 networks
by Serge S. Yuriev 13 Oct '16

13 Oct '16

Hello, Two SIP profiles: External 83.хх Internal 10.23.154.0/24 Via external we are receiving/send calls from/to 172.17.2.0/29 For some reason if we call outside FS sends unmodified addresses in SDP. So we have unroutable address in SDP and one-way audio. If call flows ext to int all working correct. Tried local-network-acl on inside (10.хх) with excluded 172.хх, apply-nat-acl with included 172.xx on either int and ext. Nothing helps :( "Bad one" SDP - from internal to external send 960 bytes to udp/[10.23.154.63]:6060 at 18:16:22.226984: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.23.154.63:6060;branch=z9hG4bKe433fa68b81 From: "IT, Юрьев Сергей" <sip:12550@10.23.154.63>;tag=195594~27154efa-6325-45a2-9e47-67e5d9302ebc-237816120 To: <sip:62987%236546@10.23.154.100>;tag=66NUXXHvB6HBp Call-ID: 86c80-7f71bc46-c44e-3f40000a(a)10.23.154.63 CSeq: 101 INVITE Contact: <sip:mod_sofia@10.23.154.100:6060> User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20160707T165535Z~be13536ac9~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, PRACK, NOTIFY Require: timer Supported: precondition, 100rel, timer, path, replaces Allow-Events: talk, hold, conference, refer Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 180 v=0 o=- 1475853382 2 IN IP4 172.17.2.3 s=- >> c=IN IP4 172.17.2.4 b=AS:64 t=0 0 m=audio 3040 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 And a good one - external to internal send 1162 bytes to udp/[10.23.154.65]:5060 at 12:34:15.132027: ------------------------------------------------------------------------ INVITE sip:12550@10.23.154.65 SIP/2.0 Via: SIP/2.0/UDP 10.23.154.100:6060;rport;branch=z9hG4bKUXyFjDmg8rtmB Max-Forwards: 69 From: "Абонент" <sip:$(caller_id_number)@10.23.154.100>;tag=1agg8aZ7FUUBK To: <sip:12550@10.23.154.65> Call-ID: d8367628-0fc1-4325-998f-3f32f9d3a05b CSeq: 97580363 INVITE Contact: <sip:gw+cucm-65@10.23.154.100:6060;transport=udp;gw=cucm-65> User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20160707T165535Z~be13536ac9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, PRACK, NOTIFY Supported: precondition, 100rel, timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 268 X-FS-Support: update_display,send_info Remote-Party-ID: "Абонент" <sip:$(caller_id_number)@10.23.154.100>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1475804423 1475804424 IN IP4 10.23.154.100 s=FreeSWITCH >> c=IN IP4 10.23.154.100 t=0 0 m=audio 28432 RTP/AVP 8 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 -- Serge S. Yuriev Senior VoIP engineer

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Re: [Freeswitch-users] Inbound or Outbound Socket connection?
by Anis Bedhiafi 13 Oct '16

13 Oct '16

Ok David. There are too things here. I have the dialplan from one side and I have the esl socket from the other side. In the dialplan I should call the esl connection somehow to work on that specefic call in progress. Right? So how can I make this link without an instruction in the dialplan? Does this make sence? On 13 Oct 2016 4:20 pm, "David Villasmil" <david.villasmil.work(a)gmail.com> wrote: You don't need to invoke it, you connect to fs permanently and subscribe to the events you're interested in. When you get the events, you can get all the info you want and act accordingly. On Thu, Oct 13, 2016 at 4:36 PM Anis Bedhiafi <anis.bedhiafi(a)e-volut.io> wrote: > Hi David, > > Thank you for the answer. In the perl script I need to do some logic to > detect the event and send a notify back to the event system. So my concern > is how I make freeswitch invokes the perl script in my dialplan where I > have already a bridge application? > > Regards > > On 13 Oct 2016 3:29 pm, "David Villasmil" <david.villasmil.work(a)gmail.com> > wrote: > > Hello, > > Why not connect to fs via perl esl and filter the events? (Inbound) you > remove the overhead of creating a new connection everytime a call is > executed... > you'd connect to fs from perl and keep the connection open. > > David > On Thu, Oct 13, 2016 at 4:21 PM Anis Bedhiafi <anis.bedhiafi(a)e-volut.io> > wrote: > > Thank you very much. > > I am digging me too. > > Thank you again. > > Best Regards. > > On 13 Oct 2016 3:19 pm, "Valter Nogueira" <valter(a)fastway.com.br> wrote: > > I understood now, when invoking socket with async, it should start a new > thread and continue to next step bridging the call. > > Well, I have never tried such thing before, I will do some testing and > when I get any result I return to you. > > Regards, > > Valter > > 2016-10-13 11:06 GMT-03:00 Anis Bedhiafi <anis.bedhiafi(a)e-volut.io>: > > Dear Valtar, > > Thank you for your answer. > > Yes I am using ESL bind for perl. > > On 13 Oct 2016 2:59 pm, "Valter Nogueira" <valter(a)fastway.com.br> wrote: > > When you are using <action application="socket" data="127.0.0.1:8084 > async"/> it means you are in outbound mode (from the perspective of FS) > > Are you using ESL bind for perl or have you written your own > implementation? > > After accepting a connection on 8084 port, you should issue a > "connect\n\n" back to FS > > Valter > > > 2016-10-12 7:08 GMT-03:00 Anis Bedhiafi <anis.bedhiafi(a)e-volut.io>: > > Hi Guys, > > I guess I have a little confusion here between inbound esl and outbound > esl. > > *My Goal:* What I am trying to do is detect if the call is being > transferred and notify the caller about it. > > *My Scenario:* > > 1- I have an application based on SIP.js client that makes outbound calls > via a registered SIP trunk on my Freeswitch box. I upload a list of phone > numbers to call and hit Launch button. > > 2- Some calls can be transferred by the callee endpoint. I want to detect > this fact. > > 3- According to my research, I can do this using Event Socket library. > 4- To detect if there is a transfer, I can control the channel variable > RDNIS or TRANSFER_TIME. If there is a transfer these variables are not > empty. > 5- Once I detect there is a transfer I want to send a NOTIFY packet to the > client. > > *My confusion is the following:* > > I don't know whether to use an Outbound socket connection or an Inbound > Connection. > > I tried to do this with outbound connection as follows: > > <condition field="destination_number" expression="^1703$"> > <action application="set" data="socket_resume=true"/> > <action application="socket" data="127.0.0.1:8084 async"/> > <action application="bridge" data="sofia/gateway/mygateway/$1"/> > </condition> > > I have a perl server listening on port 8084/tcp. > In this perl script I subscribe for CHANNEL_ANSWER Events and I filter by > the call uuid. > > The problem is when I call the socket connection, the bridge application > never executes. > > How am I supposed to listen for a specific event and make a bridge call? > > Is what I am doing wrong. Please correct me. > > Thank you for your support. > > Kind regards. > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 > > _________________________________________________________________________ > 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 _________________________________________________________________________ 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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Re: [Freeswitch-users] FreeSwitch General Info
by Ahmed Munir 13 Oct '16

13 Oct '16

Few things I forgot to ask, the scale-ability I was referring to as horizontal scale and thanks for your input on that. For point#2, mod_curl you mentioned for realtime, will it sustain more than 500-1K CPS and will it stress out the httpd/apache service? For point#3, you mean to say Freeswitch doesn't use system files for media like Asterisk does? For point#4, multi-threading; I forgot to mentioned whether Freeswitch supports multi-processing? As Asterisk, it uses/runs on single process whereas OpenSIPs/Kamilio, we can set multiple child processes. From: Michael Jerris <mike(a)jerris.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 13 Oct 2016 10:04:56 -0400 > Subject: Re: [Freeswitch-users] FreeSwitch General Info > > > On Oct 13, 2016, at 9:52 AM, Ahmed Munir <ahmedmunir007(a)gmail.com> > wrote: > > > > Hi, > > > > Currently I'm browsing different platforms online for open source VoIP > technologies as currently using Asterisk as PBX. > > > > I would like to know following things about FreeSwitch; > > > > 1- FreeSwitch whether supports realtime integration sip > profiles/extensions (using DB) and dialplan? (Just like in Asterisk) > > yes, most frequently using mod_xml_curl which uses http to pull xml on a > per dialplan/directory lookup basis > > > 2- On average, how many extensions and active calls (CPS) it can sustain? > > This widely varies based on what you are doing. Calls with no media > passed through the switch would be measured much more in calls per second > than total calls, and could be in the thousands, calls with heavy video > transcoding would be in the double digits on a big box. What you are doing > could lie anywhere on that spectrum. Our recommendation is always to test > your scenario and figure out sizing for yourself. Typically people with > heavy load are very satisfied with our performance.. > > > 3- For a call channel, will it create or use system sockets (openfile)? > As you know that Asterisk create/open and consume the system sockets for > each call. > > Certainly we use sockets for media. We do not use them in the way > asterisk does. > > > 4- Will it supports single thread or multithreading? > > FreeSWITCH is heavily multithreaded. Each call will use at least 2 > threads, one for each call leg. > > > > 5- Is it scale-able in term of setting up as a cluster(s)? > > Sip itself allows for this. Depends on your exact scenario how to > architect in a way to make this scale horizontally, but it can be done in > most scenarios > > > > > I'll be glad to hear it from you. > > > > -- > > Regards, > > > > Ahmed Munir Chohan > > _______________________________________________ > 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 > > -- Regards, Ahmed Munir Chohan -- Regards, Ahmed Munir Chohan

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Inbound or Outbound Socket connection?
by Anis Bedhiafi 13 Oct '16

13 Oct '16

Hi Guys, I guess I have a little confusion here between inbound esl and outbound esl. *My Goal:* What I am trying to do is detect if the call is being transferred and notify the caller about it. *My Scenario:* 1- I have an application based on SIP.js client that makes outbound calls via a registered SIP trunk on my Freeswitch box. I upload a list of phone numbers to call and hit Launch button. 2- Some calls can be transferred by the callee endpoint. I want to detect this fact. 3- According to my research, I can do this using Event Socket library. 4- To detect if there is a transfer, I can control the channel variable RDNIS or TRANSFER_TIME. If there is a transfer these variables are not empty. 5- Once I detect there is a transfer I want to send a NOTIFY packet to the client. *My confusion is the following:* I don't know whether to use an Outbound socket connection or an Inbound Connection. I tried to do this with outbound connection as follows: <condition field="destination_number" expression="^1703$"> <action application="set" data="socket_resume=true"/> <action application="socket" data="127.0.0.1:8084 async"/> <action application="bridge" data="sofia/gateway/mygateway/$1"/> </condition> I have a perl server listening on port 8084/tcp. In this perl script I subscribe for CHANNEL_ANSWER Events and I filter by the call uuid. The problem is when I call the socket connection, the bridge application never executes. How am I supposed to listen for a specific event and make a bridge call? Is what I am doing wrong. Please correct me. Thank you for your support. Kind regards.

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 124, Issue 65
by Ahmed Munir 13 Oct '16

13 Oct '16

Thanks Michael, Few things I forgot to ask, the scale-ability I was referring to as horizontal scale and thanks for your input on that. For point#2, mod_curl you mentioned for realtime, will it sustain more than 500-1K CPS and will it stress out the httpd/apache service? For point#3, you mean to say Freeswitch doesn't use system files for media like Asterisk does? For point#4, multi-threading; I forgot to mentioned whether Freeswitch supports multi-processing? As Asterisk, it uses/runs on single process whereas OpenSIPs/Kamilio, we can set multiple child processes. From: Michael Jerris <mike(a)jerris.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 13 Oct 2016 10:04:56 -0400 > Subject: Re: [Freeswitch-users] FreeSwitch General Info > > > On Oct 13, 2016, at 9:52 AM, Ahmed Munir <ahmedmunir007(a)gmail.com> > wrote: > > > > Hi, > > > > Currently I'm browsing different platforms online for open source VoIP > technologies as currently using Asterisk as PBX. > > > > I would like to know following things about FreeSwitch; > > > > 1- FreeSwitch whether supports realtime integration sip > profiles/extensions (using DB) and dialplan? (Just like in Asterisk) > > yes, most frequently using mod_xml_curl which uses http to pull xml on a > per dialplan/directory lookup basis > > > 2- On average, how many extensions and active calls (CPS) it can sustain? > > This widely varies based on what you are doing. Calls with no media > passed through the switch would be measured much more in calls per second > than total calls, and could be in the thousands, calls with heavy video > transcoding would be in the double digits on a big box. What you are doing > could lie anywhere on that spectrum. Our recommendation is always to test > your scenario and figure out sizing for yourself. Typically people with > heavy load are very satisfied with our performance.. > > > 3- For a call channel, will it create or use system sockets (openfile)? > As you know that Asterisk create/open and consume the system sockets for > each call. > > Certainly we use sockets for media. We do not use them in the way > asterisk does. > > > 4- Will it supports single thread or multithreading? > > FreeSWITCH is heavily multithreaded. Each call will use at least 2 > threads, one for each call leg. > > > > 5- Is it scale-able in term of setting up as a cluster(s)? > > Sip itself allows for this. Depends on your exact scenario how to > architect in a way to make this scale horizontally, but it can be done in > most scenarios > > > > > I'll be glad to hear it from you. > > > > -- > > Regards, > > > > Ahmed Munir Chohan > > _______________________________________________ > 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 > > -- Regards, Ahmed Munir Chohan

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FreeSwitch General Info
by Ahmed Munir 13 Oct '16

13 Oct '16

Hi, Currently I'm browsing different platforms online for open source VoIP technologies as currently using Asterisk as PBX. I would like to know following things about FreeSwitch; 1- FreeSwitch whether supports realtime integration sip profiles/extensions (using DB) and dialplan? (Just like in Asterisk) 2- On average, how many extensions and active calls (CPS) it can sustain? 3- For a call channel, will it create or use system sockets (openfile)? As you know that Asterisk create/open and consume the system sockets for each call. 4- Will it supports single thread or multithreading? 5- Is it scale-able in term of setting up as a cluster(s)? I'll be glad to hear it from you. -- Regards, Ahmed Munir Chohan

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Capturing the Current Call Audio
by Leonardo Lima Ribeiro 13 Oct '16

13 Oct '16

Hi! I’m tying to capture the current call audio but I have no idea on how to do that. My main achievement is to get the current call audio and them stream this audio bytes to an API via HTTP Post. I was doing some tries with mod_shout and mod_vlc, and I was successful in the streaming part - I tested some wavs and mp3 files. But the mainly part is that I need the current call audio, and it has no file in the system to stream right? I was thinking if there’s a command that I can get the audio bytes from the current call but I really don’t know, I could not find any option like that. Could you please help me? Thank you, Leonardo Ribeiro

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