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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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Opensips integration questions
by Schneur Rosenberg 12 Oct '16

12 Oct '16

Hi, I'm pretty new to Freeswitch, I built a complete solution using OpenSIPS/Asterisk and I'm in the process of replacing Asterisk with Freeswitch. My current system is built that OpenSIPS act as a registrar and as a load balancer, the rest is done on a cluster of Asterisk machines, which as I said before is being rewritten to Freeswitch, in Freeswitch I'm using xml_curl for my dialplan and call routing, I'm using Intralanman project to handle the sip users. With Asterisk I had OpensSIPS as a trunk and I know that everything coming from that trunk, is either a DID or a authenticated outgoing call, Asterisk and OpenSIPS shared the user table, Asterisk recognized the sip user name from the FROM field and used all the variables etc from its own sip user table as if the user has registered directly to itself, I had some issues with blind transfers but I was able to do some manipulations and DB queries to handle it fine. I tried doing the same with Freeswitch, I let the ACL allow the ip from Freeswitch, but Freeswitch would not use the variables from the user like Asterisk did, for example accountcode, user_context etc, I was able to partially fix that by having my web server do a db query and then using the "set" application set all the variables, but I'm not sure if its the proper solution, does anyone have a better solution? will setting my OpenSIPS as a gateway instead of merely allowing it in with ACL work better? right now I'm using the application "execute_extension" to send it to the proper context that I've pulled from the DB, I dont like the hack and if there is a better solution I would like to know. One more major problem that I have is NAT related, when I place a outgoing call for example from a phone which is behind a NAT (all of our customers phones are behind NAT) the call connects and audio works 2 ways, but when the called party hangs up, the carrier sends the BYE to Freeswitch, freeswitch then sends it to my OpenSIPS which is supposed to send it to my clients phone, but Freeswich writes the RURI with the natted address, the RURI is BYE sip:101@192.168.0.11:5062 SIP/2.0. and my OpenSIPS does not know what to do with it and ignores it, my Asterisk replaced the RURI with the received address in the VIA, so same RURI would be BYE sip:101@64.69.47.175:5062 SIP/2.0. which worked fine, same problem happens with all subsequent requests if user does not hang up, I need a solution for this major issue. thank you S. Rosenberg

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Freeswitch is not receiving audio unless and until send RTP to other end
by Naveen Tamanam 12 Oct '16

12 Oct '16

Hi, Recently I have found the following scenario, Freeswitch is not receiving audio unless and until we send RTP(audio) to other end. I have taken both recording and pcap for the call which is not having outgoing audio. Surprisingly, there is a incoming audio in pcap but not in recording. There is no NAT on machine. Is this any kind of known bug ? -- Thanks & Regards, Naveen Tamanam

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DTMF mode
by Yu Boot 12 Oct '16

12 Oct '16

FS receives DTMF tones from A-leg in RFC2833 mode, but generates SIP INFO to B-leg instead of RFC2833. How to fix this?

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mod_lcr with mssql backend
by Guillermo Ruiz Camauer 11 Oct '16

11 Oct '16

Does anyone have mod_lcr running against a MSSQL database through ODBC? What modifications did you have to make? -- Guillermo Ruiz Camauer

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Using SIPS and not SIP TLS
by Emrah 11 Oct '16

11 Oct '16

Hi there, I am in the middle of an argument with a softphone vendor over how to handle SIP and TLS. They support TLS with SIP and TLS with SIPS. If I pick TLS with SIPS, the softphone can lookup my SRV records and register on my FS box. However, it comes through with a transport=tcp as a contact parameter. If I use TLS with SIP, the softphone will ignore my _SIPS._TCP records but come through with a transport=tls. TLS with SIP and TLS with SIPS are almost identical with SIPS ensuring, in theory, that all signaling hops will use TLS. This, however, requires that FreeSWITCH uses a SIPS uri scheme to contact my endpoint and encapsulate in TLS regardless of the transport parameter. What do you suggest? Here is my ticket with them: https://www.acrobits.net/hesk/ticket.php?track=YQL-TRS-J3M4 Best, Emrah

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Re: [Freeswitch-users] Playback on Congestion
by Stanislav Sinyagin 11 Oct '16

11 Oct '16

Then you need some scripting, like Lua or an external ESL agent. On 10 Oct 2016 18:49, "Richard Gleason" <rgleason(a)voipnetwork.net> wrote: But it does not allow to specify rejection codes, for example I want to have two different recordings for SIP 404 and SIP 480. On Mon, Oct 10, 2016 at 3:12 AM, Stanislav Sinyagin <ssinyagin(a)gmail.com> wrote: > You can see an example of using continue_on_fail here: > http://wiki.sangoma.com/NSC-SIP-Trunking-Load-Balancing-And-Failover > > In your case, you play back a message instead of the second bridge. > > > > On Mon, Oct 10, 2016 at 5:15 AM, Richard Gleason > <rgleason(a)voipnetwork.net> wrote: > > Hello Guys, > > > > Need some help with a dial plan that I'm working on. > > > > The goal is to play a pre-recorded message to the originator on > congestion. > > > > For example, if GW replies back to FreeSWITCH with code SIP 404, I want > to > > trigger a pre-recorded message and play it back to the originator. > > > > Dial plan uses option bridge. > > > > If it is not difuclt can some one provide an example of dial plan for > this > > action. > > > > Thanks. > > > > -- > > Best Regards, > > Richard Gleason > > > > ____________________________________________________________ > _____________ > > 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 > -- Best Regards, Richard Gleason _________________________________________________________________________ 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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Playback on Congestion
by Richard Gleason 10 Oct '16

10 Oct '16

Hello Guys, Need some help with a dial plan that I'm working on. The goal is to play a pre-recorded message to the originator on congestion. For example, if GW replies back to FreeSWITCH with code SIP 404, I want to trigger a pre-recorded message and play it back to the originator. Dial plan uses option bridge. If it is not difuclt can some one provide an example of dial plan for this action. Thanks. -- Best Regards, Richard Gleason

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