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Re: [Freeswitch-users] *.tmp.xml.fsxml Files
by Luke Milbourne 03 Oct '16

03 Oct '16

Hi Josh, Hope your keeping well mate. I know about that one :) These ones are suffixed ".tmp.xml.fsxml" with what looks like a UUID as there filename. But they're all empty....just wondered if anyone else knows what they're for. Thanks, Luke On 3 October 2016 at 10:43, <freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. > > *.tmp.xml.fsxml Files (Luke Milbourne) > 2. Re: *.tmp.xml.fsxml Files (Joshua Gigg) > 3. issue with ilbc and ptime between a call between webrtc > client and freeswitch (Caragea Silviu) > 4. Re: Registration Unauthorized then 481 Call Does Not Exist > (Benjamin Cropley) > > > ---------- Forwarded message ---------- > From: Luke Milbourne <luke.milbourne(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Mon, 3 Oct 2016 09:37:56 +0100 > Subject: [Freeswitch-users] *.tmp.xml.fsxml Files > Hi All, > > Can anyone confirm what the purpose of these files are in FreeSWITCH's log > directory? > > They're all empty so no clues in the actual files. > > Kind regards, > > -- > Luke Milbourne > <luke.milbourne(a)gmail.com> > > Tel: 07857154817 > Google Talk/Email: luke.milbourne(a)gmail.com > > > ---------- Forwarded message ---------- > From: Joshua Gigg <giggsey(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 03 Oct 2016 08:58:41 +0000 > Subject: Re: [Freeswitch-users] *.tmp.xml.fsxml Files > In the freeswitch.xml, there is a comment: > > > The preprocessor will compile the full xml document to > ${prefix}/log/freeswitch.xml.fsxml > > https://freeswitch.org/stash/projects/FS/repos/freeswitch/ > browse/conf/vanilla/freeswitch.xml#30 > > On Mon, 3 Oct 2016 at 09:40 Luke Milbourne <luke.milbourne(a)gmail.com> > wrote: > > Hi All, > > Can anyone confirm what the purpose of these files are in FreeSWITCH's log > directory? > > They're all empty so no clues in the actual files. > > Kind regards, > > -- > Luke Milbourne > <luke.milbourne(a)gmail.com> > > Tel: 07857154817 > Google Talk/Email: luke.milbourne(a)gmail.com > _________________________________________________________________________ > 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 > > > > ---------- Forwarded message ---------- > From: Caragea Silviu <silviu.cpp(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 3 Oct 2016 12:40:19 +0300 > Subject: [Freeswitch-users] issue with ilbc and ptime between a call > between webrtc client and freeswitch > Hello, > > I have a call coming from FS and going via sip to a webrtc compliant > client. > Works fine as time the codec is not ILBC. If codec is ILBC is failing > because > Webrtc consider the codec as being invalid because of the ptime line. > > Inside their code: > > bool IsILBCRateValid(int rate, int frame_size_samples) { > if (((frame_size_samples == 240) || (frame_size_samples == 480)) && > (rate == 13300)) { > return true; > } else if (((frame_size_samples == 160) || (frame_size_samples == 320)) > && > (rate == 15200)) { > return true; > } else { > return false; > } > } > > This is failing because rate will be 13300 and frame_size_samples will be > 160. > > frame_size_samples = (8000 / 1000) * 20 which is 160. > > How I can make fs to advertise correctly that the codec ILBC is 30 ms one > and not 20 ? > > Here is the sdp that's generated: > > v=0 > o=FreeSWITCH 1475470534 1475470535 IN IP4 91.231.138.218 > s=FreeSWITCH > c=IN IP4 91.XXX.XXX.XXX > t=0 0 > a=msid-semantic: WMS 5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0v > a=end-of-candidates > m=audio 16430 RTP/SAVPF 102 0 103 101 104 > a=rtpmap:102 iLBC/8000 > a=fmtp:102 mode=30 > a=rtpmap:0 PCMU/8000 > a=rtpmap:103 isac/16000 > a=fmtp:103 ibitrate=32000;maxbitrate=53400 > a=rtpmap:101 telephone-event/8000 > a=rtpmap:104 telephone-event/16000 > a=fingerprint:sha-256 B9:24:01:90:AD:E6:43:60:10:F9: > CA:3A:85:86:26:91:A6:41:A8:77:3B:F0:39:20:3C:50:0F:A3:5E:1F:48:03 > a=setup:actpass > a=rtcp-mux > a=rtcp:16430 IN IP4 91.xxx.xxxx.xxx > a=ssrc:4159978388 cname:tTLcMVJxIsF1gi8j > a=ssrc:4159978388 msid:5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0v a0 > a=ssrc:4159978388 mslabel:5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0v > a=ssrc:4159978388 label:5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0va0 > a=ice-ufrag:VWQeyw6k2xxshnNB > a=ice-pwd:6oDJad6JwBACKzMvlsAKOwUH > a=candidate:8788212768 1 udp 659136 91.XXX.XXX.XXX16430 typ host > generation 0 > a=candidate:8788212768 2 udp 659136 91.XXX.XXX.XXX16430 typ host > generation 0 > a=silenceSupp:off - - - - > a=ptime:20 > > Silviu > > > ---------- Forwarded message ---------- > From: Benjamin Cropley <benjamin.cropley(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 3 Oct 2016 10:42:42 +0100 > Subject: Re: [Freeswitch-users] Registration Unauthorized then 481 Call > Does Not Exist > 1. 50.97.212.23 sends 103.12.214.115 a REGISTER > 2. 103.12.214.115 spots that no credentials were supplied, and so supplies > a nonce etc. in a WWW-Authenticate header in a 401 packet > 3. 50.97.212.23 sends an ACK to acknowledge this request for credentials > 4. 50.97.212.23 sends 103.12.214.115 a REGISTER with an Authorization > header > 5. 103.12.214.115 sends a 481 which means 103.12.214.115 has received the > REGISTER 'outside of a dialog'. > 6. 50.97.212.23 sends an ACK to acknowledge the error > > This is wierd, because based on the trace alone 103.12.214.115 should see > the REGISTER with Authorization as part of the previous dialog. Or at the > very least, treat it as a new request. > > I would look at some actual logs on 103.12.214.115 for more information. > > Good luck, > Ben > > On Sun, Oct 2, 2016 at 1:59 PM, Tanim <tanim05(a)gmail.com> wrote: > >> Hi, >> >> >> >> When I try to registration its giving following log. Its only happen a >> specific dialer but that dialer also work other softswitch. It’s getting >> problem only with freeswitch. My freeswitch version is 1.6.9 >> >> I can’t figure out why its happen. How can I overcome this issue and is >> there any sip message issue or not? So that I can ask those dialer company >> to fix their issue. >> >> >> >> Thanks in advance. >> >> >> >> >> >> REGISTER sip:103.12.214.115:5060 SIP/2.0 >> >> Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea >> >> Max-Forwards: 70 >> >> From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie >> >> To: <sip:12345678@103.12.214.115:5060> >> >> Call-ID: smvpcaakieaaaaaaaaa >> >> CSeq: 1 REGISTER >> >> Contact: <sip:wmaeaccba@50.97.212.23:1310> >> >> Expires: 3600 >> >> Content-Length: 0 >> >> >> >> SIP/2.0 401 Unauthorized >> >> Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea >> >> From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie >> >> To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe >> >> Call-ID: smvpcaakieaaaaaaaaa >> >> CSeq: 1 REGISTER >> >> User-Agent: WHaT Rocks! >> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, PRACK, NOTIFY >> >> Supported: precondition, 100rel, path, replaces >> >> WWW-Authenticate: Digest realm="103.12.214.115", >> nonce="2a9726d3-fac2-4d7b-9773-771b60a35e7a", algorithm=MD5, qop="auth" >> >> Content-Length: 0 >> >> >> >> ACK sip:103.12.214.115:5060 SIP/2.0 >> >> Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea >> >> Max-Forwards: 70 >> >> From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie >> >> To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe >> >> Call-ID: smvpcaakieaaaaaaaaa >> >> CSeq: 1 ACK >> >> Contact: <sip:wmaeaccba@50.97.212.23:1310> >> >> Content-Length: 0 >> >> >> >> REGISTER sip:103.12.214.115:5060 SIP/2.0 >> >> Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea >> >> Max-Forwards: 70 >> >> From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie >> >> To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe >> >> Call-ID: smvpcaakieaaaaaaaaa >> >> CSeq: 2 REGISTER >> >> Contact: <sip:wmaeaccba@50.97.212.23:1310> >> >> Expires: 3600 >> >> Authorization: Digest username="12345678",realm="103 >> .12.214.115",nonce="2a9726d3-fac2-4d7b-9773-771b60a35e7a",uri="sip: >> 103.12.214.115:5060",algorithm=MD5,response="73c3be >> ac11531911319f05807714c047" >> >> Content-Length: 0 >> >> >> >> SIP/2.0 481 Call Does Not Exist >> >> Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea >> >> From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie >> >> To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe >> >> Call-ID: smvpcaakieaaaaaaaaa >> >> CSeq: 2 REGISTER >> >> User-Agent: WHaT Rocks! >> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, PRACK, NOTIFY >> >> Supported: precondition, 100rel, path, replaces >> >> Content-Length: 0 >> >> >> >> ACK sip:103.12.214.115:5060 SIP/2.0 >> >> Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea >> >> Max-Forwards: 70 >> >> From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie >> >> To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe >> >> Call-ID: smvpcaakieaaaaaaaaa >> >> CSeq: 2 ACK >> >> Contact: <sip:wmaeaccba@50.97.212.23:1310> >> >> Content-Length: 0 >> >> >> >> Regards >> >> Minat >> >> _________________________________________________________________________ >> 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 >> > > > > -- > All the best, > Ben Cropley > 07539 366 905 > > _______________________________________________ > 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 > > -- Luke Milbourne <luke.milbourne(a)gmail.com> Tel: 07857154817 Google Talk/Email: luke.milbourne(a)gmail.com

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*.tmp.xml.fsxml Files
by Luke Milbourne 03 Oct '16

03 Oct '16

Hi All, Can anyone confirm what the purpose of these files are in FreeSWITCH's log directory? They're all empty so no clues in the actual files. Kind regards, -- Luke Milbourne <luke.milbourne(a)gmail.com> Tel: 07857154817 Google Talk/Email: luke.milbourne(a)gmail.com

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B-leg continuously Ringing
by Murugan Pandian 03 Oct '16

03 Oct '16

Hi, When i try to make outbound PSTN call using Gateway ,My phone ring continually i am getting ring until unless i disconnect call from a leg,if i disconnect call from my phone again i am getting ring to my phone . Bellow is my dial plan <extension name="outbound call"> <condition field="destination_number" expression="^919xxxxxxxx$"> <action application="bridge" data="sofia/gateway/flowroute/919xxxxxxxx"/> <action application="hangup"/> </condition> </extension> Regard's, Pandian

2 1

Bridging calls to other server
by Caio Assis 03 Oct '16

03 Oct '16

Good afternoon. The situation I'm facing is this: I have a server that has 2 network interfaces (eth0 and eth1), the former with an external static IP set, and the ladder with a internal IP set. I have one more server on my private network that bridges calls to my gsm gateway (which means it's set with a internal IP as well), and I have a VoIP account to bridge other calls. The problem I'm facing is: when I try to bridge a call to my gateway, in my private network, the IP freeswitch is sending is the external IP, so the call is denied by my gateway. With the VoIP I don't have any problem, since the IP external meets my voip's external IP. Just to give you an example, it's like this. Main Server (eth0: 192.168.1.50, eth1: 123.123.123.123) Main Server bridges calls to Gateway (192.168.1.255). The gateway recieves an request coming from 123.123.123.123, and it's set to decline external IP, so the call gets a forbidden signal. Main Server bridges calls to VoIP provider(177.177.177.177). VoIP accepts my calls, since the request it receives is coming from the external IP. I need Freeswitch to send the call to the gateway using the internal IP and voip using external IP. I use asterisk, and i set a parameter brindaddr=0.0.0.0 to accept request coming from both internal and external IPs. All of my asterisk configurations work fine. Thanks.

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Registration Unauthorized then 481 Call Does Not Exist
by Tanim 03 Oct '16

03 Oct '16

Hi, When I try to registration its giving following log. Its only happen a specific dialer but that dialer also work other softswitch. It's getting problem only with freeswitch. My freeswitch version is 1.6.9 I can't figure out why its happen. How can I overcome this issue and is there any sip message issue or not? So that I can ask those dialer company to fix their issue. Thanks in advance. REGISTER sip:103.12.214.115:5060 SIP/2.0 Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea Max-Forwards: 70 From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie To: <sip:12345678@103.12.214.115:5060> Call-ID: smvpcaakieaaaaaaaaa CSeq: 1 REGISTER Contact: <sip:wmaeaccba@50.97.212.23:1310> Expires: 3600 Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe Call-ID: smvpcaakieaaaaaaaaa CSeq: 1 REGISTER User-Agent: WHaT Rocks! Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY Supported: precondition, 100rel, path, replaces WWW-Authenticate: Digest realm="103.12.214.115", nonce="2a9726d3-fac2-4d7b-9773-771b60a35e7a", algorithm=MD5, qop="auth" Content-Length: 0 ACK sip:103.12.214.115:5060 SIP/2.0 Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea Max-Forwards: 70 From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe Call-ID: smvpcaakieaaaaaaaaa CSeq: 1 ACK Contact: <sip:wmaeaccba@50.97.212.23:1310> Content-Length: 0 REGISTER sip:103.12.214.115:5060 SIP/2.0 Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea Max-Forwards: 70 From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe Call-ID: smvpcaakieaaaaaaaaa CSeq: 2 REGISTER Contact: <sip:wmaeaccba@50.97.212.23:1310> Expires: 3600 Authorization: Digest username="12345678",realm="103.12.214.115",nonce="2a9726d3-fac2-4d7b-9773-77 1b60a35e7a",uri="sip:103.12.214.115:5060",algorithm=MD5,response="73c3beac11 531911319f05807714c047" Content-Length: 0 SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe Call-ID: smvpcaakieaaaaaaaaa CSeq: 2 REGISTER User-Agent: WHaT Rocks! Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY Supported: precondition, 100rel, path, replaces Content-Length: 0 ACK sip:103.12.214.115:5060 SIP/2.0 Via: SIP/2.0/UDP 50.97.212.23:1310;branch=z9hG4bKhabaiiea Max-Forwards: 70 From: <sip:12345678@103.12.214.115:5060>;tag=a34xd9anboejegocoaaoeie To: <sip:12345678@103.12.214.115:5060>;tag=FavjcHKUKtXFe Call-ID: smvpcaakieaaaaaaaaa CSeq: 2 ACK Contact: <sip:wmaeaccba@50.97.212.23:1310> Content-Length: 0 Regards Minat

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Are is it bug?
by Sergey Safarov 01 Oct '16

01 Oct '16

I cannot resolve IPv6 hostname via FS console. Are it is bug? [root@localhost6 ~]# nslookup -type=aaaa ipv6.cybernode.com Server: ::1 Address: ::1#53 Non-authoritative answer: ipv6.cybernode.com has AAAA address 2001:470:1:1b9::31 [root@localhost6 ~]# fs_cli -x "host_lookup ipv6.cybernode.com" -ERR no reply [root@localhost6 ~]#

4 8

Dialplan expression matching
by Vladyslav Zakhozhai 30 Sep '16

30 Sep '16

Hi, I've mentioned that pattern matching in my dialplan works incorrectly but I can't figure out the main reason. For example I have the following dialplan: <context name="internal"> <extension name="echo_test"> <condition field="destination_number" expression="^(echo|9999)$"> Some action </condition> </extension> <extension name="is_internal_call"> <condition field="destination_number" expression=".*"/> <condition field="${sip_h_X-Exten-Type}" expression="internal"> Some action </condition> </extension> <extension name="ua_local"> <condition field="destination_number" experession="^[2-5][0-9]{6}$"> Some action </condition </extension> <extension name="ua_mobile"> <condition field="destination_number" experession="^0[0-9]{9}$"> <action application="bridge" data="sofia/gateway/golden/${destination_number}"/> </condition> </extension> </context> 1. Call to "echo" - OK (echo_test) 2. Call to some extension with custom header X-Exten-Type - OK (is_internal_call) 3. Call to 5930000 - OK (ua_local) 4. Call to 0930000000 - FAILED to match ua_mobile extension. The last call matches ua_local. But why? The result of regex dp_tool freeswitch@vlakas> regex 5930000|^[2-5][0-9]{6}$ true freeswitch@vlakas> regex 0930000000|^[2-5][0-9]{6}$ false So, everything is correct. I can't understand why almost everyting goes to ua_local extension. I appreciate you help. -- Best regards, Vladyslav Zakhozhai email: v.zakhozhai(a)gmail.com tel.: +380(93) 757-21-61

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Debian8 FS-1.6.10 simple bridging and high CPU load
by Mikhail Demekhov 30 Sep '16

30 Sep '16

Hello! I have installed freeswitch using apt-get on my debian 8 sersver from https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie block under title 'latest release branch:' My server hardware: 2xCPU AMD Opteron(tm) Processor 6262 HE 64 GB RAM I'm using calls with bridging from voipgw1(2,3) to gw4(5,6), G711A on both leg A and leg B core.db in TMPFS Through my server takes about 125 simultaneous calls CPU load about 125%! It is very high in my opinion. How to fix it? top PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 13587 freeswi+ 39 19 485252 200020 16284 S 110,1 0,3 3062:38 freeswitch top -H PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 13589 freeswi+ 39 19 485252 200692 16284 S 3,0 0,3 55:08.10 freeswitch 13820 freeswi+ 39 19 485252 200692 16284 S 1,6 0,3 0:00.21 freeswitch 13606 freeswi+ 39 19 485252 200692 16284 S 1,3 0,3 27:11.91 freeswitch 12817 freeswi+ 39 19 485252 200692 16284 S 1,3 0,3 0:05.72 freeswitch 12858 freeswi+ 39 19 485252 200692 16284 S 1,3 0,3 0:06.25 freeswitch 10917 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:16.64 freeswitch 11059 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:15.60 freeswitch 11651 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:12.20 freeswitch 11868 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:11.82 freeswitch 11940 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:11.56 freeswitch 12066 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:10.54 freeswitch 12851 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:05.78 freeswitch 12852 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:05.53 freeswitch 13153 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:03.76 freeswitch 13155 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:03.65 freeswitch 13226 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:03.45 freeswitch 13423 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:02.24 freeswitch 13431 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:02.47 freeswitch 13549 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:01.67 freeswitch 13702 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:00.99 freeswitch 13719 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:00.77 freeswitch 13763 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:00.56 freeswitch 13783 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:00.52 freeswitch 13821 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:00.27 freeswitch 13822 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:00.17 freeswitch 13851 freeswi+ 39 19 485252 200692 16284 S 1,0 0,3 0:00.07 freeswitch 13834 root 20 0 26172 3488 2528 R 1,0 0,0 0:00.36 top 3 root 20 0 0 0 0 S 0,7 0,0 18:40.94 ksoftirqd/0 8 root 20 0 0 0 0 S 0,7 0,0 57:06.30 rcu_sched 13599 freeswi+ 39 19 485252 200692 16284 S 0,7 0,3 27:53.60 freeswitch 13602 freeswi+ 39 19 485252 200692 16284 S 0,7 0,3 10:13.13 freeswitch 10297 freeswi+ 39 19 485252 200692 16284 S 0,7 0,3 0:20.80 freeswitch 10613 freeswi+ 39 19 485252 200692 16284 S 0,7 0,3 0:18.69 freeswitch 10838 freeswi+ 39 19 485252 200692 16284 S 0,7 0,3 0:17.52 freeswitch 11192 freeswi+ 39 19 485252 200692 16284 S 0,7 0,3 0:14.49 freeswitch 11409 freeswi+ 39 19 485252 200692 16284 S 0,7 0,3 0:14.54 freeswitch 11450 freeswi+ 39 19 485252 200692 16284 S 0,7 0,3 0:14.57 freeswitch 11474 freeswi+ 39 19 485252 200692 16284 S 0,7 0,3 0:14.26 .......... Rgards, Demekhov Mikhail

5 11

Could not able to listen audio notification via WebRTC.
by devang nathwani 30 Sep '16

30 Sep '16

Hi All, I have setup WebRTC using verto module of freeswitch and we have done evrything successfullt just the onething we are experiencing since long time and could not get any solution. I want to play a account balance when ever any user try to make call and if there is a no balance in acccount need to play "Not enough credit in account". Now the problem is Freeswitch actually playing those notification which we can able to listen properly if we make call using softphone but i could not able to listen it via WebRTC web phone. any one can help me to fix this issue?

2 1

Chrome 54 and Verto
by Chad Phillips 29 Sep '16

29 Sep '16

Testing ahead against Chrome 54 beta using Verto in the latest 1.6 release, I’m getting a lot of websocket disconnects, as well as Verto hanging in the newCall() method. Curious if anyone else has run into this yet before I start digging in to investigate? Chad

2 1