From alex at freeswitch.com Thu Nov 1 05:04:19 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Thu, 1 Nov 2018 14:04:19 +0900 Subject: [Freeswitch-users] Freeswitch crashing : Program terminated with signal 11, Segmentation fault. In-Reply-To: References: Message-ID: Hello, FreeSWITCH 1.6.17 is too old to be supported. Try 1.8.2 on Debian (not CentOS). Regards, Alex On Wed, Oct 31, 2018 at 11:01 PM sagar malam wrote: > Hello, > > I am using FS 1.6.17 for PBX system.It was working fine since last one > month.Since last two days it has started crashing randomly.All the core > dump show same error as mentioned below : > > #0 hash (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100) at > ./src/include/private/switch_hashtable_private.h:53 > i = > #1 switch_hashtable_search (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100) > at src/switch_hashtable.c:231 > e = > hashvalue = 2433273728 > #2 0x00007f3edfb59885 in switch_core_hash_find (hash=, > key=key at entry=0x7f3ea0001100 "register") at src/switch_core_hash.c:178 > No locals. > #3 0x00007f3e90d28690 in sofia_glue_find_profile__ (file=file at entry=0x7f3e90e2c27a > "sofia_presence.c", > func=func at entry=0x7f3e90e32500 <__func__.29963> > "actual_sofia_presence_mwi_event_handler", line=line at entry=521, > key=0x7f3ea0001100 "register") at sofia_glue.c:1630 > profile = > > Please find whole core dump : > https://pastebin.com/xP2ziRMX > > System is in production.Please help. > -- > Thanks, > > Sagar > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mouli123 at gmail.com Thu Nov 1 05:53:50 2018 From: mouli123 at gmail.com (Chandramouli P) Date: Thu, 1 Nov 2018 11:23:50 +0530 Subject: [Freeswitch-users] Forward issue - FS is inviting itself In-Reply-To: References: Message-ID: Hello, Any update would be appreciated. Thank you. On Wed, Oct 31, 2018 at 6:06 PM Chandramouli P wrote: > Hello, > > We have a very simple IVR application using FreeSwitch and two Yealink IP > phones and all are in same LAN. For assumption, below are the IP addresses > along with the extension numbers: > > FreeSwitch server: 192.168.1.1 > IP phone1 (Ext. 1456): 192.168.1.2 > IP phone2 (Ext. 1459): 192.168.1.3 > > Based on the input received from IVR, FS is working fine and respective > extension phone is ringing. Till now, everything is working fine. Now, I > entered second phone extension number i.e. 1459 in first phone's "Forward" > configuration settings using web GUI of phone. Simply, I am doing the call > forwarding. If I enter, 1459 at 192.168.1.3 in "Forward" menu option, call > is forwarding to second phone and it is ringing. *But, If I simply enter > the extension number i.e. 1459 with out IP address of the second phone, FS > is adding it's IP address itself like 1459 at 192.168.1.1 <1459 at 192.168.1.1> > and sending invite itself and call is disconnecting.* The objective is > simply I just want to enter the extension number only with out IP address. > Can anybody share the thoughts to overcome this issue? > > Thank you. > > Best Regards, > Chandra. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mouli123 at gmail.com Thu Nov 1 05:54:44 2018 From: mouli123 at gmail.com (Chandramouli P) Date: Thu, 1 Nov 2018 11:24:44 +0530 Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving In-Reply-To: References: <7F36BBCD-D236-45EE-B0A0-F7DE031AC1E6@vallimamod.org> <51dc1d4b2d63413285607f7c6ed801e4@MDF-EXCH1.MDF-Holdings.local> Message-ID: Hello all, Any update would be appreciated. Thank you. Best Regards, Chandramouli. On Mon, Oct 29, 2018 at 12:23 PM Chandramouli P wrote: > Hello all, > > Any update would be appreciated. Thank you. > > Best Regards, > Chandramouli. > > On Wed, Oct 24, 2018 at 11:22 AM Chandramouli P > wrote: > >> Hello Branden, >> >> Thank you for your reply. As you said, I do understand about A-D DTMF >> digits. But, my problem is to regenerate the issue i.e. ("A"). Which key >> (s), I have to press to send "A" as DTMF digit? I am sure that the end user >> is used regular mobile phone dialer only, and not softphone. >> >> Thank you. >> >> Best Regards, >> Chandramouli. >> >> On Wed, Oct 24, 2018 at 2:58 AM Branden Jordan >> wrote: >> >>> A is a valid DTMF digit so I wouldn’t worry about it too much and just >>> handle A-D in your valid TT fields. I see around a million calls a day and >>> see a few hundred A-D DTMF. It’s rare, but some really old phones still >>> have the A-D DTMF on them, and softphones have the ability to send them as >>> well, maybe your user is using a softphone on their end? >>> >>> >>> >>> *From:* FreeSWITCH-users >>> *On Behalf Of *Chandramouli P >>> *Sent:* Monday, October 22, 2018 11:23 PM >>> *To:* FreeSWITCH Users Help >>> *Subject:* Re: [Freeswitch-users] DTMF Issue - Unexpected digits are >>> receiving >>> >>> >>> >>> Hello Abdullah, >>> >>> >>> >>> Thank you for your reply. By following your hint, I tried to regenerate >>> the issue. I tried by getting voice mail beep, other beeps like receiving >>> SMS, WhatsApp messages etc. But, unable to simulate. >>> >>> >>> >>> Thank you. >>> >>> >>> >>> Best regards, >>> >>> Chandramouli. >>> >>> On Mon, Oct 22, 2018 at 8:00 PM Vallimamod Abdullah >>> wrote: >>> >>> Hi, >>> >>> >>> >>> I encountered the same issue in the past and have noticed that it was >>> the mobile voicemail beep tone that was sometimes converted to 'A' dtmf on >>> the isdn gateway. >>> >>> >>> >>> Maybe your case is similar? >>> >>> >>> >>> >>> >>> Best Regards, >>> -- >>> Vallimamod Abdullah >>> SIP Solutions >>> vma at sip.solutions >>> linkedin.com/in/vallimamod >>> . >>> >>> >>> >>> On 22 Oct 2018, at 08:24, Chandramouli P wrote: >>> >>> >>> >>> Hello David, >>> >>> >>> >>> Thanks for your reply and I understand. But, As I explained, we tried to >>> regenerate the issue by pressing different keys on dial pad. But, >>> FreeSwitch is receiving the digits as usual every time. So, which key was >>> pressed by the user on dial pad to send "A"? Any assumptions? >>> >>> >>> >>> Thank you. >>> >>> >>> >>> Best regards, >>> >>> Chandramouli. >>> >>> On Sun, Oct 21, 2018 at 10:53 PM David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>> From what i know, "A" is a multifrequency tone, 697hz+1633hz. Maybe the >>> client is actually sending that. >>> >>> >>> >>> On Sat, Oct 20, 2018, 17:54 Chandramouli P wrote: >>> >>> Hello, >>> >>> >>> >>> We developed an application (FreeSwitch) that receives pressed digit by >>> the end user on mobile phone dialer through DTMF and some action will be >>> performed. Everything is working fine till now. But, we noticed one issue >>> recently. From the FreeSwitch log, we came to know that FreeSwitch is >>> received the letter i.e. "A" from DTMF, instead of digit. We tried to >>> regenerate this issue. But, when we are trying to regenerate, FreeSwitch is >>> receiving the digit from DTMF as usual. I would like to know How FreeSwitch >>> is received the letter i.e. "A", instead of digit. Which key was pressed by >>> the end user on mobile phone dialer? >>> >>> >>> >>> Thank you. >>> >>> >>> >>> Best Regards, >>> >>> Chandramouli. >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From sagarmalam at gmail.com Thu Nov 1 08:51:49 2018 From: sagarmalam at gmail.com (sagar malam) Date: Thu, 1 Nov 2018 14:21:49 +0530 Subject: [Freeswitch-users] Freeswitch crashing : Program terminated with signal 11, Segmentation fault. In-Reply-To: References: Message-ID: I found a way to reproduce the issue. Recently we added support for mobile apps and we are using linphone library of APP.SIP INVITE packet coming from this APP is crashing FS.We also found which headers are leading to this crash. INVITE Causing crash : 2018/11/01 08:37:13.066391 10.50.7.253:5060 -> 10.50.7.251:5070 INVITE sip:1047 at ecosmob.sip.teledge.com SIP/2.0 X-arp : 7206 X-arip : 192.168.2.8 Record-Route: x-video: no x-mac-address: CA6268B1FD88 Record-Route: Via: SIP/2.0/UDP 66.160.227.253:5060 ;branch=z9hG4bK5089.b3e4827194e27326cba1a3d6d0820e60.0 Via: SIP/2.0/UDP 198.136.226.1:5060 ;rport=5060;received=10.50.8.1;branch=z9hG4bK5089.14e5615c7abb8698dfb87f5100ac476d.0 Via: SIP/2.0/UDP 192.168.2.8:50202 ;received=202.131.119.122;branch=z9hG4bK.9gVXpSGbU;rport=35927 From: ;tag=fsvt11Fla To: "1047" CSeq: 20 INVITE Call-ID: vOprfDSdcK Max-Forwards: 70 Supported: replaces, outbound, gruu Content-Type: application/sdp Content-Length: 468 Contact: *;+s* *.instance="";+org.linphone.specs=groupchat* User-Agent: Netcarrier.Telecom_iPhone.SE_iOS12.0.1/4.0.1-97-gf3949dc26 (belle-sip/1.6.3) v=0 o=1001/CA6268B1FD88 767 2345 IN IP4 192.168.2.8 s=Talk c=IN IP4 10.50.7.253 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 21588 RTP/AVP 0 8 9 18 101 a=rtcp-fb:* trr-int 1000 a=rtcp-fb:* ccm tmmbr a=direction:active a=oldmediaip:192.168.2.8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=yes We found contact header was having sip.intance parameter highlighted in above SIP INVITE.We removed it and it fixed the issue.see below INVITE working fine : 2018/11/01 08:39:22.822052 10.50.7.253:5060 -> 10.50.7.251:5070 INVITE sip:1047 at ecosmob.sip.teledge.com SIP/2.0 X-arp : 7226 X-arip : 192.168.2.8 Record-Route: x-video: no x-mac-address: CA6268B1FD88 Record-Route: Via: SIP/2.0/UDP 192.168.2.8:50202 ;received=202.131.119.122;branch=z9hG4bK.UCWmJVLuz;rport=35927 From: ;tag=aB3e~fcu3 To: "1047" CSeq: 20 INVITE Call-ID: d0VM~gqyVj Max-Forwards: 70 Supported: replaces, outbound, gruu Content-Type: application/sdp Content-Length: 469 User-Agent: Netcarrier.Telecom_iPhone.SE_iOS12.0.1/4.0.1-97-gf3949dc26 (belle-sip/1.6.3) *Contact: * v=0 o=1001/CA6268B1FD88 3869 2900 IN IP4 192.168.2.8 s=Talk c=IN IP4 10.50.7.253 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 21640 RTP/AVP 0 8 9 18 101 a=rtcp-fb:* trr-int 1000 a=rtcp-fb:* ccm tmmbr a=direction:active a=oldmediaip:192.168.2.8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=yes a=sendrecv a=rtcp:21641 My issue got resolved with this but i request FS developer team to find cause of this crash and get it fixed. On Wed, Oct 31, 2018 at 6:19 PM sagar malam wrote: > Hello, > > I am using FS 1.6.17 for PBX system.It was working fine since last one > month.Since last two days it has started crashing randomly.All the core > dump show same error as mentioned below : > > #0 hash (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100) at > ./src/include/private/switch_hashtable_private.h:53 > i = > #1 switch_hashtable_search (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100) > at src/switch_hashtable.c:231 > e = > hashvalue = 2433273728 > #2 0x00007f3edfb59885 in switch_core_hash_find (hash=, > key=key at entry=0x7f3ea0001100 "register") at src/switch_core_hash.c:178 > No locals. > #3 0x00007f3e90d28690 in sofia_glue_find_profile__ (file=file at entry=0x7f3e90e2c27a > "sofia_presence.c", > func=func at entry=0x7f3e90e32500 <__func__.29963> > "actual_sofia_presence_mwi_event_handler", line=line at entry=521, > key=0x7f3ea0001100 "register") at sofia_glue.c:1630 > profile = > > Please find whole core dump : > https://pastebin.com/xP2ziRMX > > System is in production.Please help. > -- > Thanks, > > Sagar > -- Thanks, Sagar -------------- next part -------------- An HTML attachment was scrubbed... URL: From ravindra.bhatt at ecosmob.com Thu Nov 1 13:51:18 2018 From: ravindra.bhatt at ecosmob.com (Ravindrakumar Bhatt) Date: Thu, 1 Nov 2018 19:21:18 +0530 Subject: [Freeswitch-users] Can not bypass media if used with tls and srtp Message-ID: Hello, I am using freeswitch 1.6.20 and users are registered with TLS and i am using SRTP for media. In normal scenario calls are working fine but when i try uuid_media off call_uuid , i am getting error message Cannot bypass sofia/internal/user at ip due to secure connection. So is there any way to bypass SRTP from freeswitch after call is answered ? -- *Thanks and Regards,* *Ravindrakumar Bhatt* Jr. Software Developer Ecosmob Technologies Ltd Ahmedabad Mo:*+918460692402* -------------- next part -------------- An HTML attachment was scrubbed... URL: From sagarmalam at gmail.com Thu Nov 1 15:33:26 2018 From: sagarmalam at gmail.com (sagar malam) Date: Thu, 1 Nov 2018 21:03:26 +0530 Subject: [Freeswitch-users] Freeswitch crashing : Program terminated with signal 11, Segmentation fault. In-Reply-To: References: Message-ID: Yes i will test it with latest FS and report back on Jira Thanks On Thu, Nov 1, 2018 at 8:46 PM Brian West wrote: > 1.6.x is EOL and no longer supported, 1.6.20 was the last release of 1.6, > you should try that to see if the issue is fixed. If the problem persists > please try 1.8.2 and then file a JIRA. > > Thanks, > /b > > > On Wed, Oct 31, 2018 at 8:13 AM sagar malam wrote: > >> Hello, >> >> I am using FS 1.6.17 for PBX system.It was working fine since last one >> month.Since last two days it has started crashing randomly.All the core >> dump show same error as mentioned below : >> >> #0 hash (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100) at >> ./src/include/private/switch_hashtable_private.h:53 >> i = >> #1 switch_hashtable_search (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100) >> at src/switch_hashtable.c:231 >> e = >> hashvalue = 2433273728 >> #2 0x00007f3edfb59885 in switch_core_hash_find (hash=, >> key=key at entry=0x7f3ea0001100 "register") at src/switch_core_hash.c:178 >> No locals. >> #3 0x00007f3e90d28690 in sofia_glue_find_profile__ (file=file at entry=0x7f3e90e2c27a >> "sofia_presence.c", >> func=func at entry=0x7f3e90e32500 <__func__.29963> >> "actual_sofia_presence_mwi_event_handler", line=line at entry=521, >> key=0x7f3ea0001100 "register") at sofia_glue.c:1630 >> profile = >> >> Please find whole core dump : >> https://pastebin.com/xP2ziRMX >> >> System is in production.Please help. >> -- >> Thanks, >> >> Sagar >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Thanks, Sagar -------------- next part -------------- An HTML attachment was scrubbed... URL: From sagarmalam at gmail.com Fri Nov 2 05:28:47 2018 From: sagarmalam at gmail.com (sagar malam) Date: Fri, 2 Nov 2018 10:58:47 +0530 Subject: [Freeswitch-users] Freeswitch crashing : Program terminated with signal 11, Segmentation fault. In-Reply-To: References: Message-ID: Yes.That was reported by my colleague. On Thu, Nov 1, 2018 at 10:27 PM Brian West wrote: > Guessing based on the flow of data, this is your issue too > https://freeswitch.org/jira/browse/FS-11493? > > /b > > On Wed, Oct 31, 2018 at 5:17 PM Brian West wrote: > >> 1.6.x is EOL and no longer supported, 1.6.20 was the last release of 1.6, >> you should try that to see if the issue is fixed. If the problem persists >> please try 1.8.2 and then file a JIRA. >> >> Thanks, >> /b >> >> >> On Wed, Oct 31, 2018 at 8:13 AM sagar malam wrote: >> >>> Hello, >>> >>> I am using FS 1.6.17 for PBX system.It was working fine since last one >>> month.Since last two days it has started crashing randomly.All the core >>> dump show same error as mentioned below : >>> >>> #0 hash (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100) at >>> ./src/include/private/switch_hashtable_private.h:53 >>> i = >>> #1 switch_hashtable_search (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100) >>> at src/switch_hashtable.c:231 >>> e = >>> hashvalue = 2433273728 >>> #2 0x00007f3edfb59885 in switch_core_hash_find (hash=, >>> key=key at entry=0x7f3ea0001100 "register") at src/switch_core_hash.c:178 >>> No locals. >>> #3 0x00007f3e90d28690 in sofia_glue_find_profile__ (file=file at entry=0x7f3e90e2c27a >>> "sofia_presence.c", >>> func=func at entry=0x7f3e90e32500 <__func__.29963> >>> "actual_sofia_presence_mwi_event_handler", line=line at entry=521, >>> key=0x7f3ea0001100 "register") at sofia_glue.c:1630 >>> profile = >>> >>> Please find whole core dump : >>> https://pastebin.com/xP2ziRMX >>> >>> System is in production.Please help. >>> -- >>> Thanks, >>> >>> Sagar >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Thanks, Sagar -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Fri Nov 2 07:21:38 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Fri, 2 Nov 2018 16:21:38 +0900 Subject: [Freeswitch-users] debian 9 stable release install : how to put the debug symbols on ? In-Reply-To: References: Message-ID: Try apt install freeswitch-dbg Regards, Alex On Wed, Oct 31, 2018 at 10:59 PM Julien Terrasson < julien.terrasson at gmail.com> wrote: > I'm having coredumps in my call handling scenarios since i moved to debian > 9 + FS Version 1.8.2 -3-a98a958ac3 64bit. > I need to put the debug symbols on so that the coredumps can be > interpreted. > > However : > Installing the stable release with debian package procedure doesn't put > the debug symbols on. > Installing the stable release with sources + compilation procedure (witch > i beleive from the Debugger > > guidelines should put the debug symbols on) ends up in FS Version 1.8.2 git > a98a958 being installed. > > How can i have the latest stable release (1.8.2 -3-a98a958ac3 64bit) > installed with the debug symbols on ? > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From sagarmalam at gmail.com Fri Nov 2 16:12:42 2018 From: sagarmalam at gmail.com (sagar malam) Date: Fri, 2 Nov 2018 21:42:42 +0530 Subject: [Freeswitch-users] Cannot activate late negotiation In-Reply-To: <184568100.424524.1541000616837@webmail.strato.de> References: <184568100.424524.1541000616837@webmail.strato.de> Message-ID: It's late negotiation from sip trace. However I can see 183 session in progress that mean you must have "instant ring back" variable set in dialplan. You can try unsetting it. On Wed 31 Oct, 2018, 10:10 PM Ulrich Backes, wrote: > Hi all, > > how can I switch to late negotiation? FS 1.8, new installed, default > settings! > > internal.xml: > > > default.xml (added 'inherit_codec'): > > > Expected result: late-negotiation. Actual result: early-negotiation (see > image below). > > Thanks and kind regards. > Uli > > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 150338 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 150338 bytes Desc: not available URL: From s.safarov at gmail.com Sat Nov 3 10:04:21 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Sat, 3 Nov 2018 13:04:21 +0300 Subject: [Freeswitch-users] FS 1.8 rpm packaging Message-ID: Hello Now created PR for FS 1.8 rpm packaging. https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1561 If you want package any module that not exist now, then please let us know. Current rpm packages: Wrote: /builddir/build/RPMS/freeswitch-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-devel-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-abstraction-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-avmd-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-blacklist-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-callcenter-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-cidlookup-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-conference-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-curl-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-db-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-directory-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-distributor-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-easyroute-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-enum-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-esf-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-expr-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-fifo-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-fsk-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-fsv-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-hash-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-httapi-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-http-cache-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-lcr-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-limit-1.9.0-1.el7.x86_64.rpm Wrote: 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/builddir/build/RPMS/freeswitch-application-translate-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-valet_parking-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-video_filter-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-voicemail-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-application-voicemail-ivr-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-asrtts-flite-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-asrtts-pocketsphinx-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-asrtts-tts-commandline-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-asrtts-unimrcp-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-codec-passthru-amr-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-codec-av-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-codec-passthru-amrwb-1.9.0-1.el7.x86_64.rpm Wrote: 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/builddir/build/RPMS/freeswitch-endpoint-dingaling-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-endpoint-portaudio-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-endpoint-rtmp-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-endpoint-skinny-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-endpoint-verto-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-endpoint-rtc-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-freetdm-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-event-cdr-mongodb-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-event-cdr-pg-csv-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-event-cdr-sqlite-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-event-erlang-event-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-event-format-cdr-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-event-multicast-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-event-json-cdr-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-event-radius-cdr-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-event-rayo-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-event-snmp-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-event-kazoo-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-logger-graylog2-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-format-local-stream-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-format-native-file-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-format-portaudio-stream-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-format-shell-stream-1.9.0-1.el7.x86_64.rpm Wrote: /builddir/build/RPMS/freeswitch-format-shout-1.9.0-1.el7.x86_64.rpm Wrote: 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URL: From mouli123 at gmail.com Mon Nov 5 06:33:09 2018 From: mouli123 at gmail.com (Chandramouli P) Date: Mon, 5 Nov 2018 12:03:09 +0530 Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving In-Reply-To: References: <7F36BBCD-D236-45EE-B0A0-F7DE031AC1E6@vallimamod.org> <51dc1d4b2d63413285607f7c6ed801e4@MDF-EXCH1.MDF-Holdings.local> Message-ID: Hello Piotr, Thanks for your reply. But, my issue is how to regenerate this issue? That is the main problem as I explained. Thank you. Regards, Chandramouli. Virus-free. www.avg.com <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> On Thu, Nov 1, 2018 at 10:21 PM Piotr Gregor wrote: > Hi Chandramouli, > > This is DTMF association to symbols: > > 569 * > 570 * Tones: > 571 * 1209 Hz | 1336 Hz | 1477 Hz | 1633 Hz > 572 * 697 Hz '1' '2' '3' 'A' > 573 * 770 Hz '4' '5' '6' 'B' > 574 * 852 Hz '7' '8' '9' 'C' > 575 * 941 Hz '*' '0' '#' 'D' > > If the device has different keypad from above then you need to find out > which key it has mapped to DTMF comprised from 697 Hz/1633 Hz frequencies. > > cheers, > Piotr > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From sr at flexmind.de Sun Nov 4 16:47:12 2018 From: sr at flexmind.de (sr at flexmind.de) Date: Sun, 4 Nov 2018 17:47:12 +0100 Subject: [Freeswitch-users] Lose Race instead of missed call when calling multiple phones parallel Message-ID: Hello, we have some Problems with the right signalling of missed calls when calling multiple phones in parallel Here's the scenario: Phone no 49 is calling a group with 2170 and 3275 with the following dialstring Destination dialstrings are seperated by ":_:" ("Enterprise Origination"). We use curly brackets instead of "<" as we sometimes have to insert asserted identy tags into the dialstring. We checked 3 versions of Freeswitch for this Version Feb 2016 shows missed calls on both phones. Even if one phone answers, the other phone one still shows a missed call (reason for upgrading to newer Freeswitch) Version Aug 2017 never shows missed call, see logs and hangup message below Version 10/Dec 2017(yesterday) never shows missed call, as above So for the 2 never Freeswitch Versions, here are the logs at hangup 2017-12-11 17:07:46.022450 [DEBUG] sofia.c:7283 Channel sofia/internal/49 at flex.mydomain.de:5060 entering state [terminated][487] 2017-12-11 17:07:46.022450 [NOTICE] sofia.c:8474 Hangup sofia/internal/49 at flex.mydomain.de:5060 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627 Hangup sofia/internal/2170 at 94.xx.xxx.xx:42170 [CS_CONSUME_MEDIA] [LOSE_RACE] 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866 Cannot create outgoing channel of type [user] cause: [LOSE_RACE] 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863 Originate Resulted in Error Cause: 502 [LOSE_RACE] 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/2170 at 94.xx.xxx.xx:42170) Running State Change CS_HANGUP (Cur 3 Tot 22) 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/2170 at 94.xx.xxx.xx:42170) Callstate Change RINGING -> HANGUP 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/2170 at 94.xx.xxx.xx:42170) State HANGUP 2017-12-11 17:07:46.042331 [DEBUG] mod_sofia.c:449 Channel sofia/internal/2170 at 94.xx.xxx.xx:42170 hanging up, cause: LOSE_RACE 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627 Hangup sofia/internal/3275 at 94.xx.xxx.xx:43275 [CS_CONSUME_MEDIA] [LOSE_RACE] 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866 Cannot create outgoing channel of type [user] cause: [LOSE_RACE] 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863 Originate Resulted in Error Cause: 502 [LOSE_RACE] Here is the Cancel message for one of the called phones: U 2017/12/11 17:07:46.045120 144.xx.xxx.xx:5060 -> 94.xx.xxx.xx:42170 CANCEL sip:2170 at 94.xx.xxx.xx:42170 SIP/2.0. Via: SIP/2.0/UDP 144.xx.xxx.xx;rport;branch=z9hG4bK07r3ce94cvjpp. Max-Forwards: 70. From: "Test" ;tag=mQS1eFDpp1peS. To: . Call-ID: 4238b5a7-5930-1236-b2ab-00505600a1a5. CSeq: 116168167 CANCEL. Reason: SIP;cause=200;text="Call completed elsewhere". Content-Length: 0. Any hints why this happens, or anyone has this scenario working? Best Regard Stephan Reich Leiter Technik foncloud GmbH & Co KG Hahlweg 2a 36093 Künzell Tel: +49 661 968990 20 / Fax: +49 661 968990-99 Email: stephan.reich at foncloud.net Web: www.foncloud.net Registergericht: Amtsgericht Fulda, Persönlich haftende Gesellschafterin der foncloud GmbH&Co.KG: Global Brain Network GmbH Geschäftsführer der Global Brain Network GmbH: Peter Krug Sitz der Gesellschaft: Künzell. Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: 0eb5f48c.e33a6fec.jpg Type: image/jpeg Size: 56822 bytes Desc: not available URL: From fcastelco at gmail.com Mon Nov 5 15:34:55 2018 From: fcastelco at gmail.com (Federico Castro) Date: Mon, 5 Nov 2018 12:34:55 -0300 Subject: [Freeswitch-users] Debian and Centos not officialy supported by server Message-ID: Hi all, I installed FS on a HP DL-120 Gen9 server using Debian but reading server specifications I found that Debian and Centos are not officially supported for that server (and I realized that many others does not support too). Anyway I have everything installed and running except for RAID controller because there’s no drivers for that. So, trying to avoid unexpected behaviors on production I would like to know if any of you have some experience on using hardware that does not officialy support Debian/Centos? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: From flokrrr at gmail.com Mon Nov 5 15:35:18 2018 From: flokrrr at gmail.com (Florent Krieg) Date: Mon, 5 Nov 2018 16:35:18 +0100 Subject: [Freeswitch-users] How to handle acceptance or rejection for SIP SUBSCRIBE requests (subscriptions) Message-ID: Hello everybody, How can I handle acceptance/rejection for subscribe requests? The idea is to apply 'rights' on users, and make for instance user1 able to subscribe to user2's presence (eg a colleague) but unable to subscribe to user3's presence (eg the company boss)? I currently have two ideas of doing so, but none fully satisfies me because they're dirty tricks and not built-in. My ideas: 1/ handle user directory with xml_curl => when sending the request with sip_auth_method=SUBSCRIBE I check From/To and the rights: if allowed I reply the user profile, otherwise I reply "not found" (erf! what sip-level trick...) 2/ write an external script that subscribe to freeswitch events related to presence, accept any subscription, after each successful subscription the script checks the database entry: if allowed don't touch anything, if not allowed then delete the entry from the db (super-erf! db-level trick) Would there be other solutions that I would have missed? I mean, proper and clean solutions. Thanks a lot for your help :) Florent -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmarand at prosodie.com Mon Nov 5 15:46:58 2018 From: rmarand at prosodie.com (MARAND, Remi) Date: Mon, 5 Nov 2018 15:46:58 +0000 Subject: [Freeswitch-users] One way audio case between FS and Chrome on WebRTC. Message-ID: Hello, I am trying to validate FS as a SIP to WebRTC Gateway in our lab environment. I started in middle October and cannot understand this One Way Audio trouble. I must thank the Freeswitch team and contributors for this very impressive work. FreeSWITCH Version 1.8.1-2-4f54cff36a~64bit (-2-4f54cff36a 64bit) On system: SMP Debian 4.9.110-3+deb9u6 (2018-10-08) x86_64 GNU/Linux Openssl version : OpenSSL 1.1.0f 25 May 2017 Chrome version: 69.0 (I tried with different version and with Firefox with the same trouble). The wss part is ok with sip.js and verto.js The Ice negotiation is ok, I use sometimes local networks and sometimes web, I have had to authorize networks in the candidate ACL and domain ACL (acl.conf.xml) The result is the same on both topology. DTLS negotiation is OK, and there is UDP streams between Chrome (or Firefox) and FS in both ways. There is no audio in the direction from FS to Chrome, the other direction is OK. The simplest test is to call the 5000 number from the Chrome client, I send you a paste bin and pcap trace for this call. Should you give me information element to progress on this, what is really mandatory in the sip_profile/internal.xml and external.xml files, and in directory/default/1000.xml for a WebRTC call ?? What should be the good options in fs_cli to see if the encryption of RTP packets is ok or not.. ? Do you think that I have to reinstall a Freeswitch from the current branch ? @IP for FS : 192.168.145.67 @IP for Chrome : 10.70.54.43 Link on the pastebin : https://pastebin.freeswitch.org/view/09a72087 The pcap trace joined is done on Freeswitch on the same call. Thank you for helping me !! Best regards. [Prosodie-signature] Rémi Marand - Product Owner - Pod Connect. PROSODIE - Marketing & Produit Tél. : +33 (0)1.46.84.12.77 / 06.87.72.53.25 rmarand at prosodie.com This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 3368 bytes Desc: image001.gif URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Pros7cMon4.pcap Type: application/octet-stream Size: 3315221 bytes Desc: Pros7cMon4.pcap URL: From chad at apartmentlines.com Mon Nov 5 16:18:05 2018 From: chad at apartmentlines.com (Chad Phillips) Date: Mon, 5 Nov 2018 08:18:05 -0800 Subject: [Freeswitch-users] Debian and Centos not officialy supported by server In-Reply-To: References: Message-ID: I've run both CentOS and Debian on Dell PowerEdge servers for nearly 10 years now, with virtually no issues related to the hardware/software interface, and neither of those are officially supported by Dell. However, Dell does officially support many varieties of Linux, including RHEL (of which CentOS is basically a clone), and after some research I felt comfortable enough with the unofficial support for the operating systems I wanted to use. No idea if that translates to the HP world, though... On Mon, Nov 5, 2018 at 8:01 AM Federico Castro wrote: > Hi all, I installed FS on a HP DL-120 Gen9 server using Debian but > reading server specifications I found that Debian and Centos are not > officially supported for that server (and I realized that many others does > not support too). > Anyway I have everything installed and running except for RAID controller > because there’s no drivers for that. > So, trying to avoid unexpected behaviors on production I would like to > know if any of you have some experience on using hardware that does not > officialy support Debian/Centos? > > Thanks in advance. > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmarand at prosodie.com Mon Nov 5 17:00:30 2018 From: rmarand at prosodie.com (MARAND, Remi) Date: Mon, 5 Nov 2018 17:00:30 +0000 Subject: [Freeswitch-users] One way audio case between FS and Chrome on WebRTC Message-ID: Hello, I am trying to validate FS as a SIP to WebRTC Gateway in our lab environment. I started in middle October and have good result on it, but i cannot understand this One Way Audio trouble. I must thank the Freeswitch team and contributors for this very impressive work. FreeSWITCH Version 1.8.1-2-4f54cff36a~64bit (-2-4f54cff36a 64bit) On system: SMP Debian 4.9.110-3+deb9u6 (2018-10-08) x86_64 GNU/Linux Openssl version : OpenSSL 1.1.0f 25 May 2017 Chrome version: 69.0 (I tried with different version and with Firefox with the same trouble). The wss part is ok with sip.js and verto.js The Ice negotiation is ok, I use sometimes local networks and sometimes web, I have had to authorize networks in the candidate ACL and domain ACL (acl.conf.xml) The result is the same on both topology. DTLS negotiation is OK, and there is UDP streams between Chrome (or Firefox) and FS in both ways. There is no "audible" audio in the direction from FS to Chrome, the other direction is OK. The simplest test is to call the 5000 number from the Chrome client, I send you a paste bin and pcap trace for this call. Should you give me information element to progress on this, what is really mandatory in the sip_profile/internal.xml and external.xml files, and in directory/default/1000.xml for a WebRTC call ?? What should be the good options in fs_cli to see if the encryption of RTP packets is ok or not.. ? Do you think that I have to reinstall a Freeswitch from the current branch ? @IP for FS : 192.168.145.67 @IP for Chrome : 10.70.54.43 Link on the pastebin : https://pastebin.freeswitch.org/view/09a72087 I have a pcap on the same call that I can provide (3 Mb) if necessary.. Thank you for helping me !! Best regards. [Prosodie-signature] Rémi Marand - Product Owner - Pod Connect. PROSODIE - Marketing & Produit Tél. : +33 (0)1.46.84.12.77 / 06.87.72.53.25 rmarand at prosodie.com This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 3368 bytes Desc: image001.gif URL: From s.safarov at gmail.com Mon Nov 5 19:29:09 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Mon, 5 Nov 2018 22:29:09 +0300 Subject: [Freeswitch-users] Debian and Centos not officialy supported by server In-Reply-To: References: Message-ID: CentOS is equal to RHEL dist. I sure HP is supports RHEL. Sergey пн, 5 нояб. 2018 г. в 20:30, Chad Phillips : > I've run both CentOS and Debian on Dell PowerEdge servers for nearly 10 > years now, with virtually no issues related to the hardware/software > interface, and neither of those are officially supported by Dell. > > However, Dell does officially support many varieties of Linux, including > RHEL (of which CentOS is basically a clone), and after some research I felt > comfortable enough with the unofficial support for the operating systems I > wanted to use. > > No idea if that translates to the HP world, though... > > On Mon, Nov 5, 2018 at 8:01 AM Federico Castro > wrote: > >> Hi all, I installed FS on a HP DL-120 Gen9 server using Debian but >> reading server specifications I found that Debian and Centos are not >> officially supported for that server (and I realized that many others does >> not support too). >> Anyway I have everything installed and running except for RAID controller >> because there’s no drivers for that. >> So, trying to avoid unexpected behaviors on production I would like to >> know if any of you have some experience on using hardware that does not >> officialy support Debian/Centos? >> >> Thanks in advance. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From lexxua at gmail.com Tue Nov 6 06:37:27 2018 From: lexxua at gmail.com (Volodymyr Fedorov) Date: Tue, 6 Nov 2018 07:37:27 +0100 Subject: [Freeswitch-users] Debian and Centos not officialy supported by server In-Reply-To: References: Message-ID: Hi guys, well CentOS is not RHEL at least is not supported by RedHat/IBM directly. For Debian you are on your own for sure. The problem comes when you have a problem and basically the first thing which you hear when call hotline in case of HW-issue, this os is not supported. But I think you still can build FreeSWITCH rpm in Oracle/RHEL/SuSE Linux. BR, Volodymyr On Mon, Nov 5, 2018 at 11:08 PM Sergey Safarov wrote: > CentOS is equal to RHEL dist. > I sure HP is supports RHEL. > > Sergey > > пн, 5 нояб. 2018 г. в 20:30, Chad Phillips : > >> I've run both CentOS and Debian on Dell PowerEdge servers for nearly 10 >> years now, with virtually no issues related to the hardware/software >> interface, and neither of those are officially supported by Dell. >> >> However, Dell does officially support many varieties of Linux, including >> RHEL (of which CentOS is basically a clone), and after some research I felt >> comfortable enough with the unofficial support for the operating systems I >> wanted to use. >> >> No idea if that translates to the HP world, though... >> >> On Mon, Nov 5, 2018 at 8:01 AM Federico Castro >> wrote: >> >>> Hi all, I installed FS on a HP DL-120 Gen9 server using Debian but >>> reading server specifications I found that Debian and Centos are not >>> officially supported for that server (and I realized that many others does >>> not support too). >>> Anyway I have everything installed and running except for RAID >>> controller because there’s no drivers for that. >>> So, trying to avoid unexpected behaviors on production I would like to >>> know if any of you have some experience on using hardware that does not >>> officialy support Debian/Centos? >>> >>> Thanks in advance. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Best regards, Volodymyr -------------- next part -------------- An HTML attachment was scrubbed... URL: From satoh.osamu at nttdata-sbc.co.jp Tue Nov 6 08:55:33 2018 From: satoh.osamu at nttdata-sbc.co.jp (satoh.osamu at nttdata-sbc.co.jp) Date: Tue, 6 Nov 2018 08:55:33 +0000 Subject: [Freeswitch-users] H.323 Connection Message-ID: Hello, I am trying to connect to FreeSWITCH by H.323 protcol. I built,installed and loaded the module of mod_h323. but it not work properly. I installed it as follows. (OS, Version info) OS: CentOS Linux release 7.5.1804 (Core) FreeSWITCH: 1.6.2 PTlib: 2.10.9.2 h323plus: 1.27.0 Configure was reference to the following. https://freeswitch.org/confluence/display/FREESWITCH/mod_h323 It is in the following state.(When Call to SIP Extension) - It seems to connection is successfly(Connection established) on Client, but it can not hear any sound. - In Server of FreeSWITCH, the CPU load abnormally increases, and there is the case that service of Freeswith is stopped. I looking for logs of past mailing lists, there are cases of success and failure. http://lists.freeswitch.org/pipermail/freeswitch-users/2016-September/122560.html http://lists.freeswitch.org/pipermail/freeswitch-users/2016-June/120896.html Can you tell me what settings and installation procedures can be successfully connected? Best Regards, Satoh -------------- next part -------------- An HTML attachment was scrubbed... URL: From fcastelco at gmail.com Tue Nov 6 19:29:34 2018 From: fcastelco at gmail.com (Federico Castro) Date: Tue, 6 Nov 2018 16:29:34 -0300 Subject: [Freeswitch-users] Debian and Centos not officialy supported by server In-Reply-To: References: Message-ID: Thanks to all for sharing your experiences and tips. I agree, I think the problem could be if you need some kind of assistance from the vendor. Thanks again. El mar., 6 nov. 2018 a las 14:37, Volodymyr Fedorov () escribió: > Hi guys, well CentOS is not RHEL at least is not supported by RedHat/IBM > directly. For Debian you are on your own for sure. The problem comes when > you have a problem and basically the first thing which you hear when call > hotline in case of HW-issue, this os is not supported. > But I think you still can build FreeSWITCH rpm in Oracle/RHEL/SuSE Linux. > BR, > Volodymyr > > On Mon, Nov 5, 2018 at 11:08 PM Sergey Safarov > wrote: > >> CentOS is equal to RHEL dist. >> I sure HP is supports RHEL. >> >> Sergey >> >> пн, 5 нояб. 2018 г. в 20:30, Chad Phillips : >> >>> I've run both CentOS and Debian on Dell PowerEdge servers for nearly 10 >>> years now, with virtually no issues related to the hardware/software >>> interface, and neither of those are officially supported by Dell. >>> >>> However, Dell does officially support many varieties of Linux, including >>> RHEL (of which CentOS is basically a clone), and after some research I felt >>> comfortable enough with the unofficial support for the operating systems I >>> wanted to use. >>> >>> No idea if that translates to the HP world, though... >>> >>> On Mon, Nov 5, 2018 at 8:01 AM Federico Castro >>> wrote: >>> >>>> Hi all, I installed FS on a HP DL-120 Gen9 server using Debian but >>>> reading server specifications I found that Debian and Centos are not >>>> officially supported for that server (and I realized that many others does >>>> not support too). >>>> Anyway I have everything installed and running except for RAID >>>> controller because there’s no drivers for that. >>>> So, trying to avoid unexpected behaviors on production I would like to >>>> know if any of you have some experience on using hardware that does not >>>> officialy support Debian/Centos? >>>> >>>> Thanks in advance. >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Services >>>> sales at freeswitch.com >>>> https://freeswitch.com >>>> >>>> Official FreeSWITCH Sites >>>> https://freeswitch.com/oss >>>> https://freeswitch.org/confluence >>>> https://cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Best regards, > Volodymyr > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Tue Nov 6 21:41:28 2018 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 6 Nov 2018 18:41:28 -0300 Subject: [Freeswitch-users] mod_fail2ban links Message-ID: I am trying to install mod_fail2ban but both of the links that appear on Confluence are not working. Does anyone have working links? Has this module been abandoned? -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Wed Nov 7 04:30:26 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 7 Nov 2018 07:30:26 +0300 Subject: [Freeswitch-users] mod_fail2ban links In-Reply-To: References: Message-ID: mod_fail2ban in FS source tree. Need to compile FS from sources. ср, 7 нояб. 2018 г., 1:48 Guillermo Ruiz Camauer : > I am trying to install mod_fail2ban but both of the links that appear on > Confluence are not working. Does anyone have working links? Has this > module been abandoned? > > -- > Guillermo Ruiz Camauer > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From andrew at cassidywebservices.co.uk Wed Nov 7 09:06:56 2018 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 7 Nov 2018 09:06:56 +0000 Subject: [Freeswitch-users] Debian and Centos not officialy supported by server In-Reply-To: References: Message-ID: If I remember correctly, the raid controller in the DL120 is a 'fakeraid' one (software raid where the configuration is held in system settings). You may have better luck disabling the controller and settings the SATA ports in AHCI mode and use Linux software raid. If you still have issues, try disabling UEFI boot (change boot mode to Legacy BIOS) as this solves a number of other issues on Gen9/Gen10 with Linux. As always, get hold of the latest firmware updates too, the most recently ProLiant service pack is from September 2018. Kind regards, -- *Andrew Cassidy BSc (Hons) MBCS* Managing Director 0330 44 55 960 <+443304455960> andrew at cassidyweb.co.uk www.cassidyweb.co.uk On Tue, 6 Nov 2018, 22:30 Federico Castro, wrote: > Thanks to all for sharing your experiences and tips. > I agree, I think the problem could be if you need some kind of assistance > from the vendor. > > Thanks again. > > El mar., 6 nov. 2018 a las 14:37, Volodymyr Fedorov () > escribió: > >> Hi guys, well CentOS is not RHEL at least is not supported by RedHat/IBM >> directly. For Debian you are on your own for sure. The problem comes when >> you have a problem and basically the first thing which you hear when call >> hotline in case of HW-issue, this os is not supported. >> But I think you still can build FreeSWITCH rpm in Oracle/RHEL/SuSE Linux. >> BR, >> Volodymyr >> >> On Mon, Nov 5, 2018 at 11:08 PM Sergey Safarov >> wrote: >> >>> CentOS is equal to RHEL dist. >>> I sure HP is supports RHEL. >>> >>> Sergey >>> >>> пн, 5 нояб. 2018 г. в 20:30, Chad Phillips : >>> >>>> I've run both CentOS and Debian on Dell PowerEdge servers for nearly 10 >>>> years now, with virtually no issues related to the hardware/software >>>> interface, and neither of those are officially supported by Dell. >>>> >>>> However, Dell does officially support many varieties of Linux, >>>> including RHEL (of which CentOS is basically a clone), and after some >>>> research I felt comfortable enough with the unofficial support for the >>>> operating systems I wanted to use. >>>> >>>> No idea if that translates to the HP world, though... >>>> >>>> On Mon, Nov 5, 2018 at 8:01 AM Federico Castro >>>> wrote: >>>> >>>>> Hi all, I installed FS on a HP DL-120 Gen9 server using Debian but >>>>> reading server specifications I found that Debian and Centos are not >>>>> officially supported for that server (and I realized that many others does >>>>> not support too). >>>>> Anyway I have everything installed and running except for RAID >>>>> controller because there’s no drivers for that. >>>>> So, trying to avoid unexpected behaviors on production I would like to >>>>> know if any of you have some experience on using hardware that does not >>>>> officialy support Debian/Centos? >>>>> >>>>> Thanks in advance. >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Services >>>>> sales at freeswitch.com >>>>> https://freeswitch.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> https://freeswitch.com/oss >>>>> https://freeswitch.org/confluence >>>>> https://cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> https://freeswitch.com >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Services >>>> sales at freeswitch.com >>>> https://freeswitch.com >>>> >>>> Official FreeSWITCH Sites >>>> https://freeswitch.com/oss >>>> https://freeswitch.org/confluence >>>> https://cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> >> >> -- >> Best regards, >> Volodymyr >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From ap at gen-ip.fr Wed Nov 7 12:39:54 2018 From: ap at gen-ip.fr (Alexis) Date: Wed, 7 Nov 2018 13:39:54 +0100 Subject: [Freeswitch-users] mod_fail2ban links In-Reply-To: References: Message-ID: Hi, You can follow this page : https://freeswitch.org/confluence/display/FREESWITCH/Fail2Ban That doesn't use mod_fail2ban at all but works perfectly. I have no idea if mod_fail2ban is still maintened but last Confluence edit was in 2014. Alexis Le 06/11/2018 à 22:41, Guillermo Ruiz Camauer a écrit : > I am trying to install mod_fail2ban but both of the links that appear > on Confluence are not working.  Does anyone have working links?  Has > this module been abandoned? > > -- > Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From jan at willamowius.de Wed Nov 7 12:41:57 2018 From: jan at willamowius.de (Jan Willamowius) Date: Wed, 7 Nov 2018 13:41:57 +0100 Subject: [Freeswitch-users] H.323 Connection In-Reply-To: References: Message-ID: <20181107134157.0a70fbd7.jan@willamowius.de> Hi Satoh, check if your H.323 endpoints are behind a firewall. Freeswitch doesn't support H.460 NAT traversal natively, so you might have to install a GnuGk in front of it to proxy the media data. Regards, Jan -- Jan Willamowius, Founder of the GNU Gatekeeper Project EMail : jan at willamowius.de Website: https://www.gnugk.org Support: https://www.willamowius.com/gnugk-support.html Relaxed Communications GmbH Frahmredder 91, 22393 Hamburg, Germany Geschäftsführer: Jan Willamowius HRB 125261 (Amtsgericht Hamburg) USt-IdNr: DE286003584 satoh.osamu at nttdata-sbc.co.jp wrote: > Hello, > > I am trying to connect to FreeSWITCH by H.323 protcol. > I built,installed and loaded the module of mod_h323. but it not work properly. > > I installed it as follows. > (OS, Version info) > OS: CentOS Linux release 7.5.1804 (Core) > FreeSWITCH: 1.6.2 > PTlib: 2.10.9.2 > h323plus: 1.27.0 > > Configure was reference to the following. > https://freeswitch.org/confluence/display/FREESWITCH/mod_h323 > > > It is in the following state.(When Call to SIP Extension) > - It seems to connection is successfly(Connection established) on Client, but it can not hear any sound. > - In Server of FreeSWITCH, the CPU load abnormally increases, and there is the case that service of Freeswith is stopped. > > > I looking for logs of past mailing lists, there are cases of success and failure. > http://lists.freeswitch.org/pipermail/freeswitch-users/2016-September/122560.html > http://lists.freeswitch.org/pipermail/freeswitch-users/2016-June/120896.html > > Can you tell me what settings and installation procedures can be successfully connected? > > > Best Regards, > Satoh > From backes at solthor.de Wed Nov 7 12:55:57 2018 From: backes at solthor.de (Ulrich Backes) Date: Wed, 7 Nov 2018 13:55:57 +0100 (CET) Subject: [Freeswitch-users] How to avoid transcoding? Message-ID: <380673553.633982.1541595357082@webmail.strato.de> Hi guys, I've investigated many hours and cannot solve this problem: A and B should use g711u. A: B: But FreeSwitch decides to transcode: The settings: vars.xml internal.xml default.xml (Using the demo-Dialplan) ... Using FS v 1.8 Thanks. Uli -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 3727 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 6279 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 104271 bytes Desc: not available URL: From mail at paulzillmann.de Wed Nov 7 17:58:50 2018 From: mail at paulzillmann.de (Paul Zillmann) Date: Wed, 7 Nov 2018 18:58:50 +0100 Subject: [Freeswitch-users] How to avoid transcoding? In-Reply-To: <380673553.633982.1541595357082@webmail.strato.de> References: <380673553.633982.1541595357082@webmail.strato.de> Message-ID: <0a40d549-a5e7-9ead-324c-5980e336dcc5@paulzillmann.de> Hello Ulrich, take a look at https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation The parameter "disable-transcoding" is for the early codec negotiation. But you want to have a late codec negotiation. > /Note/: it is commonly misunderstood that this parameter disables the > transcoding capability in FS. That is wrong. > > This parameter just changes the outbound codec to match the one > negotiated on the inbound leg so that no transcoding will be required. > (Wiki Quote) Also the codec list is a priority list. When you list PCMU first, it is chosen first. - Paul Am 07.11.18 um 13:55 schrieb Ulrich Backes: > > Hi guys, > > I've investigated many hours and cannot solve this problem: > A and B should use g711u. > > > A: > > > B: > > But FreeSwitch decides to transcode: > > > The settings: > vars.xml > > data="global_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/> > > internal.xml > > > > > > > > > > default.xml (Using the demo-Dialplan) > > ... > > Using FS v 1.8 > > Thanks. > Uli > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 3727 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 6279 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 104271 bytes Desc: not available URL: From sebastian_ml at gmx.net Wed Nov 7 18:26:35 2018 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Wed, 7 Nov 2018 19:26:35 +0100 Subject: [Freeswitch-users] How to avoid transcoding? In-Reply-To: <380673553.633982.1541595357082@webmail.strato.de> References: <380673553.633982.1541595357082@webmail.strato.de> Message-ID: <20181107182634.GA2348@darth.lan> On Wed, Nov 07, 2018 at 01:55:57PM +0100, Ulrich Backes wrote: > The settings: > vars.xml > > > > internal.xml > > > > > > > > > > default.xml (Using the demo-Dialplan) > > ... > > Using FS v 1.8 Hello Uli, FS (.111) starts generating ringback and decides to use G722 on A leg. So remove the ringback for starters. Also, I'd leave "inbound-codec-negotiation" at default. "disable-transcoding" should also not be set, as this is for early negotiation, not late, IIRC. Probably it does nothing when late negotiation is used, but better just remove it. You also need in your dialplan. This is all written down here: https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation Kind regards, Seb From alex at freeswitch.com Thu Nov 8 01:56:19 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Thu, 8 Nov 2018 10:56:19 +0900 Subject: [Freeswitch-users] How to avoid transcoding? In-Reply-To: <380673553.633982.1541595357082@webmail.strato.de> References: <380673553.633982.1541595357082@webmail.strato.de> Message-ID: Hi, try Confluence: https://freeswitch.org/confluence/display/FREESWITCH/inherit_codec https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation Also, you probably don't want 183 in this situation. Regards, Alex On Wed, Nov 7, 2018 at 11:04 PM Ulrich Backes wrote: > Hi guys, > > I've investigated many hours and cannot solve this problem: > A and B should use g711u. > > > A: > > > B: > > But FreeSwitch decides to transcode: > > > The settings: > vars.xml > > data="global_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/> > > internal.xml > > > > > > > > > > default.xml (Using the demo-Dialplan) > > ... > > Using FS v 1.8 > > Thanks. > Uli > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 3727 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 6279 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 104271 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 104271 bytes Desc: not available URL: From asilva at wirelessmundi.com Thu Nov 8 08:42:13 2018 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Thu, 8 Nov 2018 09:42:13 +0100 Subject: [Freeswitch-users] How to avoid transcoding? In-Reply-To: <380673553.633982.1541595357082@webmail.strato.de> References: <380673553.633982.1541595357082@webmail.strato.de> Message-ID: <109e8c98-d14e-9b4b-16fd-a08fa993aee0@wirelessmundi.com> Hi, to understand how codec negotiation work check https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation Also, you may want to enable inherit_codec parameter this way the codec negotiated in bleg is first match against codecs from aleg. Also, another simple way to avoid transcoding is set only one codec in the configuration, set: On 07/11/2018 13:55, Ulrich Backes wrote: > > Hi guys, > > I've investigated many hours and cannot solve this problem: > A and B should use g711u. > > > A: > > > B: > > But FreeSwitch decides to transcode: > > > The settings: > vars.xml > > data="global_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/> > > internal.xml > > > > > > > > > > default.xml (Using the demo-Dialplan) > > ... > > Using FS v 1.8 > > Thanks. > Uli > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Saludos / Regards / Cumprimentos António Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 3727 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 6279 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 104271 bytes Desc: not available URL: From shaun.stokes at itec-support.co.uk Thu Nov 8 08:42:20 2018 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 8 Nov 2018 08:42:20 +0000 Subject: [Freeswitch-users] How to avoid transcoding? In-Reply-To: <380673553.633982.1541595357082@webmail.strato.de> References: <380673553.633982.1541595357082@webmail.strato.de> Message-ID: Setting FreeSWITCH to 'greedy' will cause the codecs you specified in FreeSWITCH to take priority which in your case is G722 as neither of your endpoints support OPUS, you could try setting 'inbound-codec-negotiation' to 'generous'. Even if configured correctly there will still be instances where transcoding may be required in particular if you allow call transfer, for example transferring a G722 call to an endpoint that only supports G711 will require transcoding. ________________________________ From: FreeSWITCH-users on behalf of Ulrich Backes Sent: 07 November 2018 12:55:57 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] How to avoid transcoding? Hi guys, I've investigated many hours and cannot solve this problem: A and B should use g711u. A: [cid:d242e27033084e59a8a877a137eb44ba at Open-Xchange] B: [cid:cd462457d2134dd4950ddf12e9e7ecf8 at Open-Xchange] But FreeSwitch decides to transcode: [cid:61e85c8ab9904d83b9e3553e5745e36d at Open-Xchange] The settings: vars.xml internal.xml default.xml (Using the demo-Dialplan) ... Using FS v 1.8 Thanks. Uli -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 3727 bytes Desc: image.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 6279 bytes Desc: image.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 104271 bytes Desc: image.png URL: From vma at vallimamod.org Thu Nov 8 09:10:11 2018 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Thu, 8 Nov 2018 10:10:11 +0100 Subject: [Freeswitch-users] One way audio case between FS and Chrome on WebRTC. In-Reply-To: References: Message-ID: Hi, Have you tried disabling STUN? Does it work in the same LAN? There are many "Sending RTCP NACK" lines in your logs, meaning that fsw is not receiving all the RTP from your endpoint. It may be related to the stun or you may need to check your network. Hope this helps. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr linkedin.com/in/vallimamod . > On 5 Nov 2018, at 16:46, MARAND, Remi wrote: > > Hello, > > I am trying to validate FS as a SIP to WebRTC Gateway in our lab environment. > I started in middle October and cannot understand this One Way Audio trouble. > > I must thank the Freeswitch team and contributors for this very impressive work. > > FreeSWITCH Version 1.8.1-2-4f54cff36a~64bit (-2-4f54cff36a 64bit) > On system: SMP Debian 4.9.110-3+deb9u6 (2018-10-08) x86_64 GNU/Linux > Openssl version : OpenSSL 1.1.0f 25 May 2017 > Chrome version: 69.0 (I tried with different version and with Firefox with the same trouble). > > The wss part is ok with sip.js and verto.js > > The Ice negotiation is ok, I use sometimes local networks and sometimes web, I have had to authorize networks in the candidate ACL and domain ACL (acl.conf.xml) The result is the same on both topology. > > DTLS negotiation is OK, and there is UDP streams between Chrome (or Firefox) and FS in both ways. > > There is no audio in the direction from FS to Chrome, the other direction is OK. > > The simplest test is to call the 5000 number from the Chrome client, I send you a paste bin and pcap trace for this call. > > Should you give me information element to progress on this, what is really mandatory in the sip_profile/internal.xml and external.xml files, and in directory/default/1000.xml for a WebRTC call ?? What should be the good options in fs_cli to see if the encryption of RTP packets is ok or not.. ? > Do you think that I have to reinstall a Freeswitch from the current branch ? > > @IP for FS : 192.168.145.67 > @IP for Chrome : 10.70.54.43 > > Link on the pastebin : https://pastebin.freeswitch.org/view/09a72087 > > The pcap trace joined is done on Freeswitch on the same call. > > Thank you for helping me !! > Best regards. > > > > Rémi Marand – Product Owner – Pod Connect. > PROSODIE – Marketing & Produit > Tél. : +33 (0)1.46.84.12.77 / 06.87.72.53.25 > rmarand at prosodie.com > > > > > > > > This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message. > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com From Alexander.Haugg at c4b.de Thu Nov 8 13:46:34 2018 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Thu, 8 Nov 2018 13:46:34 +0000 Subject: [Freeswitch-users] mod_verto and verto communicator on Debian 9 -> WS SETUP FAILED Message-ID: <2d61e947e3b74a139b79a51ec297757e@c4b.de> Hi, I try to setup the mod_verto – verto communicator scenario. The Certificate is OK “https://freeswitch.org/confluence/display/FREESWITCH/WebRTC#WebRTC-InstallCertificates” But for the sockets IP:8081 and IP:8082 I get the error message “WS SETUP FAILED” The question… Why the ws setup fails? vhosts is not configured, I am using the mini_httpd as http server. Thanks a lot Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Thu Nov 8 13:58:27 2018 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Thu, 8 Nov 2018 13:58:27 +0000 Subject: [Freeswitch-users] mod_verto and verto communicator on Debian 9 -> WS SETUP FAILED Message-ID: Sorry, here the CLI output 2018-11-08 14:55:20.857925 [DEBUG] mod_verto.c:4250 172.16.103.22:5477 Client Connect from 172.16.103.22:5477 accepted 2018-11-08 14:55:20.857925 [DEBUG] mod_verto.c:2003 172.16.103.22:5477 Starting client thread. 2018-11-08 14:55:20.897905 [WARNING] mod_verto.c:1864 172.16.103.22:5477 WS SETUP FAILED 2018-11-08 14:55:20.897905 [DEBUG] mod_verto.c:2030 172.16.103.22:5477 Ending client thread. 2018-11-08 14:55:20.897905 [DEBUG] mod_verto.c:2038 172.16.103.22:5477 Thread ended Von: Alexander Haugg Gesendet: Donnerstag, 8. November 2018 14:47 An: 'FreeSWITCH Users Help' Betreff: mod_verto and verto communicator on Debian 9 -> WS SETUP FAILED Hi, I try to setup the mod_verto – verto communicator scenario. The Certificate is OK “https://freeswitch.org/confluence/display/FREESWITCH/WebRTC#WebRTC-InstallCertificates” But for the sockets IP:8081 and IP:8082 I get the error message “WS SETUP FAILED” The question… Why the ws setup fails? vhosts is not configured, I am using the mini_httpd as http server. Thanks a lot Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Thu Nov 8 16:09:51 2018 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Thu, 8 Nov 2018 16:09:51 +0000 Subject: [Freeswitch-users] Audio interfacing to POTS port for WebRTC Message-ID: Strange request, so forgive me, but I’m looking to connect a PC running FS Verto/WebRTC to a telephone balancing unit. So basically, stereo audio in and out from a PC, to plug into an FXO port on a mixing console. The line switching on the console is academic as it will be permanently connected, and any dialling controlled from WebRTC. Appreciate this is a more of an electronics question, but if anyone knows a way to achieve this, someone on here will! Or maybe there’s a better way that I haven’t thought of!? Thanks R -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Thu Nov 8 16:11:15 2018 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Thu, 8 Nov 2018 17:11:15 +0100 Subject: Memory increase because of multiple instances of freeswitch Message-ID: <04fc4086-d678-2f65-ca06-059f40b89e22@wirelessmundi.com> Hi all, I notice a strange behaviour on machine due to increase of memory, when i went to see that was the process consuming the memory i notice that freeswitch have multiple  process running: root      2543 90.4  4.8 2178716 1587484 ?     S WS SETUP FAILED In-Reply-To: References: Message-ID: Your certificate is invalid or setup wrong. On Thu, Nov 8, 2018 at 10:13 AM Alexander Haugg wrote: > Sorry, here the CLI output > > > > 2018-11-08 14:55:20.857925 [DEBUG] mod_verto.c:4250 172.16.103.22:5477 > Client Connect from 172.16.103.22:5477 accepted > > 2018-11-08 14:55:20.857925 [DEBUG] mod_verto.c:2003 172.16.103.22:5477 > Starting client thread. > > 2018-11-08 14:55:20.897905 [WARNING] mod_verto.c:1864 172.16.103.22:5477 > WS SETUP FAILED > > 2018-11-08 14:55:20.897905 [DEBUG] mod_verto.c:2030 172.16.103.22:5477 > Ending client thread. > > 2018-11-08 14:55:20.897905 [DEBUG] mod_verto.c:2038 172.16.103.22:5477 > Thread ended > > > > > > *Von:* Alexander Haugg > *Gesendet:* Donnerstag, 8. November 2018 14:47 > *An:* 'FreeSWITCH Users Help' > *Betreff:* mod_verto and verto communicator on Debian 9 -> WS SETUP FAILED > > > > Hi, > > I try to setup the mod_verto – verto communicator scenario. > > The Certificate is OK “ > https://freeswitch.org/confluence/display/FREESWITCH/WebRTC#WebRTC-InstallCertificates > ” > > But for the sockets IP:8081 and IP:8082 I get the error message “WS SETUP > FAILED” > > The question… Why the ws setup fails? > > > > vhosts is not configured, I am using the mini_httpd as http server. > > > > Thanks a lot > > Alex > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Fri Nov 9 16:40:24 2018 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 9 Nov 2018 10:40:24 -0600 Subject: [Freeswitch-users] New FreeSWITCH Community Resources - SignalWire.community Message-ID: Hi, I wanted to let everyone know that, as part of our effort to expand our community resources, we have started building out some resources. We are working on a forum, some github projects for FreeSWITCH and SignalWire tools and we started a slack space. I personally have resisted more modern tools, i've been on irc for decades (and we still are #freeswitch on freenode) but our community is growing and we have all gotten more mobile and using multiple devices and we want to make sure we make our community accessible to everyone. This mailing list will always be here, its running strong for 15 years now! Here is the link to the signalwire.community slack. Come join and lets discuss how we can continue to innovate in the communications industry. https://goo.gl/1mEBwn -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Thu Nov 8 16:53:03 2018 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Thu, 8 Nov 2018 17:53:03 +0100 Subject: Memory increase because of multiple instances of freeswitch In-Reply-To: <04fc4086-d678-2f65-ca06-059f40b89e22@wirelessmundi.com> References: <04fc4086-d678-2f65-ca06-059f40b89e22@wirelessmundi.com> Message-ID: Forget to put my unit configuration: [Service] ; service Type=forking PIDFile=/run/freeswitch/freeswitch.pid ExecStart=/usr/bin/freeswitch -ncwait -nonat -scripts /scripts/fs TimeoutSec=300s Restart=on-failure RestartSec=500ms ; exec User=root Group=daemon LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 LimitSTACK=250000 LimitRTPRIO=infinity LimitRTTIME=infinity IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 On 08/11/2018 17:11, António Silva wrote: > Hi all, > > I notice a strange behaviour on machine due to increase of memory, > when i went to see that was the process consuming the memory i notice > that freeswitch have multiple  process running: > > root      2543 90.4  4.8 2178716 1587484 ?     S /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > root      7858  0.0  5.0 2289392 1649484 ?     SN   Oct19   0:00 \_ > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > root     30626  0.0  4.4 2172632 1464684 ?     SN   Oct23   0:00 \_ > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > root      4505  0.0  4.9 2336544 1621768 ?     SN   10:43   0:00 \_ > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > root     22557  0.0  4.9 2336548 1636604 ?     SN   11:02   0:00 \_ > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > > > and today: > > root     21823 93.9  4.8 2154128 1581064 ?     S /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > root     18417  0.0  4.9 2247264 1610108 ?     SN   11:09   0:00 \_ > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > > > what could be the cause of this? anyone experience the same? > > As for the service everything is running ok, is just that the memory > keeps increasing.. i restart freewitch when i start to reach my memory > limit. > > > I'm running fs 1.8.2 on debian jessie. > > -- Saludos / Regards / Cumprimentos António Silva From mjlopez at smartic.es Thu Nov 8 17:56:40 2018 From: mjlopez at smartic.es (=?iso-8859-1?Q?Miguel_Jes=FAs_L=F3pez_Valverde?=) Date: Thu, 8 Nov 2018 18:56:40 +0100 Subject: [Freeswitch-users] usernames whith special simbols cant recive calls on FS 1.8 Message-ID: <00c001d4778c$67f9add0$37ed0970$@smartic.es> Hello again guys: I'm sending this question in case someone can tell me how to solve the following problem. We currently have a production platform working with FS 1.6 where users have historically been created in the form username~companyname like usernames and the incoming calls to these users work correctly. Also, a request on console like “sofia_contact nameusername~companyname at domain” returns a right result. Now I am considering the migration of these platforms to FS 1.8 and, for this, doing an assembly of initial development. Here I have been able to verify that although these users register correctly, they can issue outgoing calls and appear correctly listed in queries of type “sofia status profile internal reg”, but these users can’t receive incoming calls, in console the information " Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] " is always obtained and when I launch the query “sofia_contact nameusername~companyname at domain” in console, I get the result “error/user_not_registered”; I only have to change the value ~ for example by a dot and the incoming calls works correctly and also the query “sofia_contact nameusername.companyname at domain” return a right result. I guess I need to adjust the configuration of some library of FS 1.8 and recompile it to get back an identical operation to version 1.6, can someone help me in indicating which library or with what adjustment I could get this operation again? Thank you very much and best regards!! Miguel J. Lopez. --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus -------------- next part -------------- An HTML attachment was scrubbed... URL: From ap at gen-ip.fr Thu Nov 8 19:17:45 2018 From: ap at gen-ip.fr (Alexis) Date: Thu, 8 Nov 2018 20:17:45 +0100 Subject: [Freeswitch-users] One way audio case between FS and Chrome on WebRTC. In-Reply-To: References: Message-ID: <6697c43c-5530-cadc-e3da-159c17f8c50f@gen-ip.fr> Hi, We got exaclty the same issue with latest Verto js library. This work with latest Verto communicator. Our developer edited verto lib to fix this issue. I'll ask him tomorrow which modifications he did. Alexis Le 08/11/2018 à 10:10, Vallimamod Abdullah a écrit : > Hi, > > Have you tried disabling STUN? > Does it work in the same LAN? > > There are many "Sending RTCP NACK" lines in your logs, meaning that fsw is not receiving all the RTP from your endpoint. It may be related to the stun or you may need to check your network. > > Hope this helps. > > > Best Regards, From satoh.osamu at nttdata-sbc.co.jp Fri Nov 9 08:36:33 2018 From: satoh.osamu at nttdata-sbc.co.jp (satoh.osamu at nttdata-sbc.co.jp) Date: Fri, 9 Nov 2018 08:36:33 +0000 Subject: [Freeswitch-users] H.323 Connection In-Reply-To: <20181107134157.0a70fbd7.jan@willamowius.de> References: , <20181107134157.0a70fbd7.jan@willamowius.de> Message-ID: Hello Jan, Thank you for your reply. Now I using FreeSWITCH on internal network of company only. Therefore, firewall and H.460 NAT have nothing todo with it. I think that the cause of this problem is FreeSWTICH setting or version of library etc, maybe. I tried other version of PTLib and h323plus, (ptlib-2.8.2 + h323plus-20100525, ptlib-trunk + h323plus-trunk) but this problem is not solve. If you are able to connect of H.323, it will be appreciated if you can share information on the environment, settings. Best Regards, Satoh ________________________________ 差出人: Jan Willamowius 送信日時: 2018年11月7日 21:41 宛先: 佐藤 督; FreeSWITCH Users Help CC: 堀田 忠司 件名: Re: [Freeswitch-users] H.323 Connection Hi Satoh, check if your H.323 endpoints are behind a firewall. Freeswitch doesn't support H.460 NAT traversal natively, so you might have to install a GnuGk in front of it to proxy the media data. Regards, Jan -- Jan Willamowius, Founder of the GNU Gatekeeper Project EMail : jan at willamowius.de Website: https://www.gnugk.org [https://www.gnugk.org/logo-200x200.png] GNU Gatekeeper - a free VOIP Gatekeeper for H.323 www.gnugk.org The GNU Gatekeeper (GnuGk) is a H.323 gatekeeper, available freely under GPL license. It forms the basis for IP telephony (VOIP) or video conferencing systems. Support: https://www.willamowius.com/gnugk-support.html Relaxed Communications GmbH Frahmredder 91, 22393 Hamburg, Germany Geschäftsführer: Jan Willamowius HRB 125261 (Amtsgericht Hamburg) USt-IdNr: DE286003584 satoh.osamu at nttdata-sbc.co.jp wrote: > Hello, > > I am trying to connect to FreeSWITCH by H.323 protcol. > I built,installed and loaded the module of mod_h323. but it not work properly. > > I installed it as follows. > (OS, Version info) > OS: CentOS Linux release 7.5.1804 (Core) > FreeSWITCH: 1.6.2 > PTlib: 2.10.9.2 > h323plus: 1.27.0 > > Configure was reference to the following. > https://freeswitch.org/confluence/display/FREESWITCH/mod_h323 > > > It is in the following state.(When Call to SIP Extension) > - It seems to connection is successfly(Connection established) on Client, but it can not hear any sound. > - In Server of FreeSWITCH, the CPU load abnormally increases, and there is the case that service of Freeswith is stopped. > > > I looking for logs of past mailing lists, there are cases of success and failure. > http://lists.freeswitch.org/pipermail/freeswitch-users/2016-September/122560.html > http://lists.freeswitch.org/pipermail/freeswitch-users/2016-June/120896.html > > Can you tell me what settings and installation procedures can be successfully connected? > > > Best Regards, > Satoh > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mbodbg at gmx.net Fri Nov 9 10:15:04 2018 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Fri, 9 Nov 2018 11:15:04 +0100 Subject: [Freeswitch-users] sip capture Message-ID: <545EA5E6-2EB6-4B60-92E3-FEA79E664E09@gmx.net> Hello, is sip-capture supported in 1.8.2? I’ve enabled sip capturing (as recommended at https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch ) , obj=0x7f71f401a160) at sofia.c:3385 #5 0x00007f7253b06e07 in dummy_worker (opaque=0x7f71f401c258) at threadproc/unix/thread.c:151 #6 0x00007f7252615494 in start_thread (arg=0x7f7249e23700) at pthread_create.c:333 #7 0x00007f7252357acf in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:97 any help or direction would be appreciated ... thank! -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Fri Nov 9 22:37:56 2018 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 9 Nov 2018 16:37:56 -0600 Subject: [Freeswitch-users] New FreeSWITCH Community Resources - SignalWire.community In-Reply-To: <90cb2618-25d7-fa1a-b113-73e7ff39caa7@madovsky.org> References: <90cb2618-25d7-fa1a-b113-73e7ff39caa7@madovsky.org> Message-ID: On Fri, Nov 9, 2018 at 4:14 PM Madovsky wrote: > well, > > https://en.wikipedia.org/wiki/Slack_%28software%29 > > is not itself open source :( > > btw there are plenty of IOS / Android app that connects to IRC > Yes, You're welcome to connect to that as well. We'll be there waiting for you. We don't discriminate against open source or not open source. We don't impose our will on others but really its more about fair or not fair with us. Slack is free for the type of use we have which is fair. There are some trade-offs for not buying it which is left to you to decide on. > On 11/9/2018 8:40 AM, Anthony Minessale wrote: > > Hi, > > I wanted to let everyone know that, as part of our effort to expand our > community resources, we have started building out some resources. We are > working on a forum, some github projects for FreeSWITCH and SignalWire > tools and we started a slack space. > > I personally have resisted more modern tools, i've been on irc for decades > (and we still are #freeswitch on freenode) but our community is growing and > we have all gotten more mobile and using multiple devices and we want to > make sure we make our community accessible to everyone. > > This mailing list will always be here, its running strong for 15 years > now! > > Here is the link to the signalwire.community slack. Come join and lets > discuss how we can continue to innovate in the communications industry. > > https://goo.gl/1mEBwn > > > > -- > Anthony Minessale II > Founder, FreeSWITCH. > http://freeswitch.com > > > https://youtu.be/l_hOxzCt6X4 > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From andywolk at gmail.com Sat Nov 10 09:33:27 2018 From: andywolk at gmail.com (Andrey Wolk) Date: Sat, 10 Nov 2018 13:33:27 +0400 Subject: [Freeswitch-users] TESTING Message-ID: Test message. -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Sat Nov 10 20:28:27 2018 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 10 Nov 2018 14:28:27 -0600 Subject: [Freeswitch-users] segfault when reloading profile In-Reply-To: References: Message-ID: Hi, Post the profile in a jira at freeswitch.org/jira and we can try to reproduce it On Sat, Nov 10, 2018 at 12:33 PM Juan Pablo L wrote: > Hello, i am getting segfaults when i reload a sip profile, i know it has > something to do with the profile itself because when i use the default > vanilla configuration and i reload a profile it does not segfault, i have > checked every single option in the profile file looking for errors or > anything and i find nothing i can fix or change, i m using: > > FS: > > FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git > a98a958 2018-09-26 17:55:25Z 64bit) > > linux: > > Linux sbc02 4.9.0-8-amd64 #1 SMP Debian 4.9.110-3+deb9u6 (2018-10-08) > x86_64 GNU/Linux > > when analysing the core file this is what i get in gdb: > > Core was generated by `freeswitch -nc'. > Program terminated with signal SIGSEGV, Segmentation fault. > #0 switch_ci_hashfunc_default (char_key=0x0, klen=klen at entry=0x7f7249e229c8) > at src/switch_apr.c:120 > 120 for (p = key; *p; p++) { > [Current thread is 1 (Thread 0x7f7249e23700 (LWP 3029))] > (gdb) bt > #0 switch_ci_hashfunc_default (char_key=0x0, klen=klen at entry=0x7f7249e229c8) > at src/switch_apr.c:120 > #1 0x00007f720a5effd6 in xml_directory_db_search (section= out>, tag_name=, key_name=, > key_value=, event_params=0x7f722400ae80, > user_data=) > at xml_directory.c:152 > #2 0x00007f72538973eb in switch_xml_locate (section=section at entry=0x7f724a0570e0 > "directory", tag_name=tag_name at entry=0x0, key_name=key_name at entry=0x0, > key_value=key_value at entry=0x0, > root=root at entry=0x7f7249e22bd0, node=node at entry=0x7f7249e22bd8, > params=0x7f722400ae80, clone=SWITCH_FALSE) at src/switch_xml.c:1695 > #3 0x00007f7249f40f92 in launch_sofia_worker_thread (profile=profile at entry=0x7f71f401a160) > at sofia.c:3062 > #4 0x00007f7249f4ac19 in sofia_profile_thread_run (thread= out>, obj=0x7f71f401a160) at sofia.c:3385 > #5 0x00007f7253b06e07 in dummy_worker (opaque=0x7f71f401c258) at > threadproc/unix/thread.c:151 > #6 0x00007f7252615494 in start_thread (arg=0x7f7249e23700) at > pthread_create.c:333 > #7 0x00007f7252357acf in clone () at > ../sysdeps/unix/sysv/linux/x86_64/clone.S:97 > > > any help or direction would be appreciated ... thank! > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From infos at madovsky.org Sat Nov 10 19:15:33 2018 From: infos at madovsky.org (Madovsky) Date: Sat, 10 Nov 2018 11:15:33 -0800 Subject: [Freeswitch-users] New FreeSWITCH Community Resources - SignalWire.community In-Reply-To: References: <90cb2618-25d7-fa1a-b113-73e7ff39caa7@madovsky.org> Message-ID: That's ok Anthony, I just point it out that I always wondered how a company like slack became so popular in less than 5 years offering their service for free with revenues coming from the sky... what' s the magic behind? On 11/9/2018 2:37 PM, Anthony Minessale wrote: > > > On Fri, Nov 9, 2018 at 4:14 PM Madovsky > wrote: > > well, > > https://en.wikipedia.org/wiki/Slack_%28software%29 > > is not itself open source :( > > btw there are plenty of IOS / Android app that connects to IRC > > Yes, > > You're welcome to connect to that as well.  We'll be there waiting for > you. > > We don't discriminate against open source or not open source. > We don't impose our will on others but really its more about fair or > not fair with us.  Slack is free for the type of use we have which is > fair. > There are some trade-offs for not buying it which is left to you to > decide on. > > > > > > > > > > On 11/9/2018 8:40 AM, Anthony Minessale wrote: >> Hi, >> >> I wanted to let everyone know that, as part of our effort to >> expand our community resources, we have started building out some >> resources.  We are working on a forum, some github projects for >> FreeSWITCH and SignalWire tools and we started a slack space. >> >> I personally have resisted more modern tools, i've been on irc >> for decades (and we still are #freeswitch on freenode) but our >> community is growing and we have all gotten more mobile and using >> multiple devices and we want to make sure we make our community >> accessible to everyone. >> >> This mailing list will always be here, its running strong for 15 >> years now! >> >> Here is the link to the signalwire.community slack.  Come join >> and lets discuss how we can continue to innovate in the >> communications industry. >> >> https://goo.gl/1mEBwn >> >> >> >> -- >> Anthony Minessale II >> Founder, FreeSWITCH. >> http://freeswitch.com >> >> >> https://youtu.be/l_hOxzCt6X4 >> https://www.youtube.com/watch?v=oAxXgyx5jUw >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > > -- > Anthony Minessale II > Founder, FreeSWITCH. > http://freeswitch.com > > > https://youtu.be/l_hOxzCt6X4 > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From hawkins at hawkinsegroup.com Sun Nov 11 03:41:50 2018 From: hawkins at hawkinsegroup.com (Don Hawkins) Date: Sat, 10 Nov 2018 21:41:50 -0600 Subject: [Freeswitch-users] TESTING In-Reply-To: References: Message-ID: It works. On Sat, Nov 10, 2018 at 12:23 PM Andrey Wolk wrote: > Test message. > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- *Sincerely,* Don Hawkins CEO Hawkins Enterprise Group LLC http://corporate.hawkinsegroup.com Zello PTT : push2don P: 469-214-5044 -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Sun Nov 11 06:50:39 2018 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 11 Nov 2018 07:50:39 +0100 Subject: [Freeswitch-users] TESTING In-Reply-To: References: Message-ID: ACK On Sat, Nov 10, 2018, 19:22 Andrey Wolk Test message. > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From freeswitch-users at funke.net Sun Nov 11 14:09:42 2018 From: freeswitch-users at funke.net (Daniel) Date: Sun, 11 Nov 2018 15:09:42 +0100 Subject: [Freeswitch-users] Problem with innovaphone devices Message-ID: <72989e0e-282e-b1f7-2e6a-34243b8f7f80@funke.net> Dear all, I'm working with FS since a week. Hope, it's ok to use it in FusionPBX ;-) I found out many cool things so far but with one thing I'm stuck. I'm not able to initiate a connection from innovaphone devices (1 deskphone IP240 and 1 PBX/GW IP800). Calling to these innovaphones is ok. From other devices to FS and softphones I can connect too. In my first try from the innovaphones I got the message 2018-11-10 23:41:58.763840 [WARNING] switch_core_media.c:4181 NO candidate ACL defined, Defaulting to wan.auto 2018-11-10 23:41:58.763840 [NOTICE] sofia.c:7774 Hangup sofia/internal/11 at pbx.[anonym].net [CS_NEW] [INCOMPATIBLE_DESTINATION] I found the solution to put apply-candidate-acl=localnet.auto to the internal SIP profile. I don't understand why this was necessary only for the innovaphones and not for other devices but after changing this I can establish a connection but cannot hear anything on both ends. After a few seconds the connection is disconnected automatically. Here's the log. I hope, someone can help https://pastebin.freeswitch.org/view/db088714 Best regards Daniel From joel at textplus.com Sun Nov 11 15:06:33 2018 From: joel at textplus.com (Joel Serrano) Date: Sun, 11 Nov 2018 07:06:33 -0800 Subject: [Freeswitch-users] New FreeSWITCH Community Resources - SignalWire.community In-Reply-To: References: <90cb2618-25d7-fa1a-b113-73e7ff39caa7@madovsky.org> Message-ID: Slack is free. Slack has a lot of features. Free tier has limited features. Companies slowly need features. Slack free becomes “not enough” Companies move to paid tiers. Slack makes money. On Sun, Nov 11, 2018 at 02:53 Madovsky wrote: > That's ok Anthony, I just point it out that > > I always wondered how a company like slack became so popular in > > less than 5 years offering their service for free with revenues coming > from the sky... what' s the magic behind? > On 11/9/2018 2:37 PM, Anthony Minessale wrote: > > > > On Fri, Nov 9, 2018 at 4:14 PM Madovsky wrote: > >> well, >> >> https://en.wikipedia.org/wiki/Slack_%28software%29 >> >> is not itself open source :( >> >> btw there are plenty of IOS / Android app that connects to IRC >> > Yes, > > You're welcome to connect to that as well. We'll be there waiting for you. > > We don't discriminate against open source or not open source. > We don't impose our will on others but really its more about fair or not > fair with us. Slack is free for the type of use we have which is fair. > There are some trade-offs for not buying it which is left to you to decide > on. > > > > > > > > > > > >> On 11/9/2018 8:40 AM, Anthony Minessale wrote: >> >> Hi, >> >> I wanted to let everyone know that, as part of our effort to expand our >> community resources, we have started building out some resources. We are >> working on a forum, some github projects for FreeSWITCH and SignalWire >> tools and we started a slack space. >> >> I personally have resisted more modern tools, i've been on irc for >> decades (and we still are #freeswitch on freenode) but our community is >> growing and we have all gotten more mobile and using multiple devices and >> we want to make sure we make our community accessible to everyone. >> >> This mailing list will always be here, its running strong for 15 years >> now! >> >> Here is the link to the signalwire.community slack. Come join and lets >> discuss how we can continue to innovate in the communications industry. >> >> https://goo.gl/1mEBwn >> >> >> >> -- >> Anthony Minessale II >> Founder, FreeSWITCH. >> http://freeswitch.com >> >> >> https://youtu.be/l_hOxzCt6X4 >> https://www.youtube.com/watch?v=oAxXgyx5jUw >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com >> >> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Anthony Minessale II > Founder, FreeSWITCH. > http://freeswitch.com > > > https://youtu.be/l_hOxzCt6X4 > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Sun Nov 11 17:11:29 2018 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 11 Nov 2018 11:11:29 -0600 Subject: [Freeswitch-users] New FreeSWITCH Community Resources - SignalWire.community In-Reply-To: References: <90cb2618-25d7-fa1a-b113-73e7ff39caa7@madovsky.org> Message-ID: On Sun, Nov 11, 2018 at 5:12 AM Madovsky wrote: > That's ok Anthony, I just point it out that > > I always wondered how a company like slack became so popular in > > less than 5 years offering their service for free with revenues coming > from the sky... what' s the magic behind? > I think if you use it for serious productivity on a team for commercial use, its compelling to pay for each user to get access to premium features. If you don’t need that, the free one is still very useable. > On 11/9/2018 2:37 PM, Anthony Minessale wrote: > > > > On Fri, Nov 9, 2018 at 4:14 PM Madovsky wrote: > >> well, >> >> https://en.wikipedia.org/wiki/Slack_%28software%29 >> >> is not itself open source :( >> >> btw there are plenty of IOS / Android app that connects to IRC >> > Yes, > > You're welcome to connect to that as well. We'll be there waiting for you. > > We don't discriminate against open source or not open source. > We don't impose our will on others but really its more about fair or not > fair with us. Slack is free for the type of use we have which is fair. > There are some trade-offs for not buying it which is left to you to decide > on. > > > > > > > > > > > >> On 11/9/2018 8:40 AM, Anthony Minessale wrote: >> >> Hi, >> >> I wanted to let everyone know that, as part of our effort to expand our >> community resources, we have started building out some resources. We are >> working on a forum, some github projects for FreeSWITCH and SignalWire >> tools and we started a slack space. >> >> I personally have resisted more modern tools, i've been on irc for >> decades (and we still are #freeswitch on freenode) but our community is >> growing and we have all gotten more mobile and using multiple devices and >> we want to make sure we make our community accessible to everyone. >> >> This mailing list will always be here, its running strong for 15 years >> now! >> >> Here is the link to the signalwire.community slack. Come join and lets >> discuss how we can continue to innovate in the communications industry. >> >> https://goo.gl/1mEBwn >> >> >> >> -- >> Anthony Minessale II >> Founder, FreeSWITCH. >> http://freeswitch.com >> >> >> https://youtu.be/l_hOxzCt6X4 >> https://www.youtube.com/watch?v=oAxXgyx5jUw >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com >> >> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Anthony Minessale II > Founder, FreeSWITCH. > http://freeswitch.com > > > https://youtu.be/l_hOxzCt6X4 > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Sun Nov 11 17:18:08 2018 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 11 Nov 2018 11:18:08 -0600 Subject: [Freeswitch-users] Problem with innovaphone devices In-Reply-To: <72989e0e-282e-b1f7-2e6a-34243b8f7f80@funke.net> References: <72989e0e-282e-b1f7-2e6a-34243b8f7f80@funke.net> Message-ID: Most SIP devices dont use ice. So that is a reasonable config change if you have a device that has ice enabled. On Sun, Nov 11, 2018 at 8:30 AM Daniel wrote: > Dear all, > > I'm working with FS since a week. Hope, it's ok to use it in FusionPBX > ;-) I found out many cool things so far but > with one thing I'm stuck. > > I'm not able to initiate a connection from innovaphone devices (1 > deskphone IP240 and 1 PBX/GW IP800). Calling to > these innovaphones is ok. From other devices to FS and softphones I can > connect too. > > In my first try from the innovaphones I got the message > > 2018-11-10 23:41:58.763840 [WARNING] switch_core_media.c:4181 NO > candidate ACL defined, Defaulting to wan.auto > 2018-11-10 23:41:58.763840 [NOTICE] sofia.c:7774 Hangup > sofia/internal/11 at pbx.[anonym].net [CS_NEW] > [INCOMPATIBLE_DESTINATION] > > I found the solution to put apply-candidate-acl=localnet.auto to the > internal SIP profile. I don't understand why > this was necessary only for the innovaphones and not for other devices > but after changing this I can establish a > connection but cannot hear anything on both ends. After a few seconds > the connection is disconnected automatically. > > Here's the log. I hope, someone can help > > https://pastebin.freeswitch.org/view/db088714 > > Best regards > Daniel > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Sun Nov 11 17:34:09 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Sun, 11 Nov 2018 20:34:09 +0300 Subject: [Freeswitch-users] Problem with innovaphone devices In-Reply-To: <72989e0e-282e-b1f7-2e6a-34243b8f7f80@funke.net> References: <72989e0e-282e-b1f7-2e6a-34243b8f7f80@funke.net> Message-ID: Try disable ICE protocol on device. вс, 11 нояб. 2018 г. в 17:35, Daniel : > Dear all, > > I'm working with FS since a week. Hope, it's ok to use it in FusionPBX > ;-) I found out many cool things so far but > with one thing I'm stuck. > > I'm not able to initiate a connection from innovaphone devices (1 > deskphone IP240 and 1 PBX/GW IP800). Calling to > these innovaphones is ok. From other devices to FS and softphones I can > connect too. > > In my first try from the innovaphones I got the message > > 2018-11-10 23:41:58.763840 [WARNING] switch_core_media.c:4181 NO > candidate ACL defined, Defaulting to wan.auto > 2018-11-10 23:41:58.763840 [NOTICE] sofia.c:7774 Hangup > sofia/internal/11 at pbx.[anonym].net [CS_NEW] > [INCOMPATIBLE_DESTINATION] > > I found the solution to put apply-candidate-acl=localnet.auto to the > internal SIP profile. I don't understand why > this was necessary only for the innovaphones and not for other devices > but after changing this I can establish a > connection but cannot hear anything on both ends. After a few seconds > the connection is disconnected automatically. > > Here's the log. I hope, someone can help > > https://pastebin.freeswitch.org/view/db088714 > > Best regards > Daniel > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From abalashov at evaristesys.com Sun Nov 11 19:44:05 2018 From: abalashov at evaristesys.com (Alex Balashov) Date: Sun, 11 Nov 2018 14:44:05 -0500 Subject: [Freeswitch-users] New FreeSWITCH Community Resources - SignalWire.community In-Reply-To: References: <90cb2618-25d7-fa1a-b113-73e7ff39caa7@madovsky.org> Message-ID: <20181111194405.GA6512@tlaquepaque.localdomain> On Sun, Nov 11, 2018 at 11:11:29AM -0600, Anthony Minessale wrote: > On Sun, Nov 11, 2018 at 5:12 AM Madovsky wrote: > > > That's ok Anthony, I just point it out that > > > > I always wondered how a company like slack became so popular in > > > > less than 5 years offering their service for free with revenues coming > > from the sky... what' s the magic behind? > > > > I think if you use it for serious productivity on a team for commercial > use, its compelling to pay for each user to get access to premium > features. If you don’t need that, the free one is still very useable. Correct, there's plenty of revenue: https://www.forbes.com/sites/greatspeculations/2018/05/25/breaking-down-slacks-valuation-an-interactive-analysis/#4af960c67616 Including from ourselves, and we're an infinitesimally tiny company. In promoting the recent addition of the Kamailio Slack channel by a member of the community, I made the following arguments in favour of diversifying beyond IRC, which I will recycle here as they speak somewhat to the issue of why it's become so popular: ----- Yes, I am aware of the open-source-purist objection to Slack; it is proprietary and perceived to be a "walled garden" of sorts. Furthermore, for those of us who have been on the Internet and involved in technology for a significant amount of time, the irony is not lost that it is little more than a packaging of the "IRC" experience in the bloated form of a modern JavaScript application. I am also very reluctant to contribute in any way to the Balkanisation or fracturing of project-related communication channels. That thought gives me no comfort. These are all concerns of which I'm mindful. Nevertheless: - A great deal of professional real-time messaging and communication in many organisations has moved to Slack. This is also true of our company, and most companies we work with. Thus, a Slack channel for would be highly complementary to a platform many other people are already using. - Slack provides a polished and cohesive user experience, which is why it has become so popular. There are clients for every platform, and a seamless user experience that is also highly compatible with mobile and tablet. One can shoehorn IRC into these media, but it clearly is not designed for them. - There are already lots of open-source discussion channels and developer forums on Slack. - Slack's inline Markdown and other conveniences make it _much_ easier to have discussions about code, configurations, etc., since it is easy to insert inline monospace and multiline blobs, attachments, etc. This is what inspired my switch to it. - A lot of people positioned within the current wave of thinking about IT and technology, and thus a big part of our candidate user base, have never used IRC. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ From freeswitch-users at funke.net Sun Nov 11 20:46:55 2018 From: freeswitch-users at funke.net (Daniel) Date: Sun, 11 Nov 2018 21:46:55 +0100 Subject: [Freeswitch-users] Problem with innovaphone devices In-Reply-To: References: <72989e0e-282e-b1f7-2e6a-34243b8f7f80@funke.net> Message-ID: Hi Sergey, thank you for this advise. In the telephone I found no option to disable ICE but in the gateway. With ICE disabled in the gateway the call still fails. Do you have other ideas what to change? https://pastebin.freeswitch.org/view/fddd86d4 Best regards Daniel On 11/11/18 6:34 PM, Sergey Safarov wrote: > Try disable ICE protocol on device. > > вс, 11 нояб. 2018 г. в 17:35, Daniel >: > > Dear all, > > I'm working with FS since a week. Hope, it's ok to use it in > FusionPBX > ;-) I found out many cool things so far but > with one thing I'm stuck. > > I'm not able to initiate a connection from innovaphone devices (1 > deskphone IP240 and 1 PBX/GW IP800). Calling to > these innovaphones is ok. From other devices to FS and softphones > I can > connect too. > > In my first try from the innovaphones I got the message > > 2018-11-10 23:41:58.763840 [WARNING] switch_core_media.c:4181 NO > candidate ACL defined, Defaulting to wan.auto > 2018-11-10 23:41:58.763840 [NOTICE] sofia.c:7774 Hangup > sofia/internal/11 at pbx.[anonym].net [CS_NEW] > [INCOMPATIBLE_DESTINATION] > > I found the solution to put apply-candidate-acl=localnet.auto to the > internal SIP profile. I don't understand why > this was necessary only for the innovaphones and not for other > devices > but after changing this I can establish a > connection but cannot hear anything on both ends. After a few seconds > the connection is disconnected automatically. > > Here's the log. I hope, someone can help > > https://pastebin.freeswitch.org/view/db088714 > > Best regards > Daniel > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From jpablolorenzetti at gmail.com Sun Nov 11 21:41:45 2018 From: jpablolorenzetti at gmail.com (Juan Pablo L) Date: Sun, 11 Nov 2018 15:41:45 -0600 Subject: [Freeswitch-users] segfault when reloading profile In-Reply-To: References: Message-ID: thank you very much, i open FS-11519 ... On Sun, 11 Nov 2018 at 06:01, Anthony Minessale wrote: > Hi, > > Post the profile in a jira at freeswitch.org/jira and we can try to > reproduce it > > On Sat, Nov 10, 2018 at 12:33 PM Juan Pablo L > wrote: > >> Hello, i am getting segfaults when i reload a sip profile, i know it has >> something to do with the profile itself because when i use the default >> vanilla configuration and i reload a profile it does not segfault, i have >> checked every single option in the profile file looking for errors or >> anything and i find nothing i can fix or change, i m using: >> >> FS: >> >> FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git >> a98a958 2018-09-26 17:55:25Z 64bit) >> >> linux: >> >> Linux sbc02 4.9.0-8-amd64 #1 SMP Debian 4.9.110-3+deb9u6 (2018-10-08) >> x86_64 GNU/Linux >> >> when analysing the core file this is what i get in gdb: >> >> Core was generated by `freeswitch -nc'. >> Program terminated with signal SIGSEGV, Segmentation fault. >> #0 switch_ci_hashfunc_default (char_key=0x0, klen=klen at entry=0x7f7249e229c8) >> at src/switch_apr.c:120 >> 120 for (p = key; *p; p++) { >> [Current thread is 1 (Thread 0x7f7249e23700 (LWP 3029))] >> (gdb) bt >> #0 switch_ci_hashfunc_default (char_key=0x0, klen=klen at entry=0x7f7249e229c8) >> at src/switch_apr.c:120 >> #1 0x00007f720a5effd6 in xml_directory_db_search (section=> out>, tag_name=, key_name=, >> key_value=, event_params=0x7f722400ae80, >> user_data=) >> at xml_directory.c:152 >> #2 0x00007f72538973eb in switch_xml_locate (section=section at entry=0x7f724a0570e0 >> "directory", tag_name=tag_name at entry=0x0, key_name=key_name at entry=0x0, >> key_value=key_value at entry=0x0, >> root=root at entry=0x7f7249e22bd0, node=node at entry=0x7f7249e22bd8, >> params=0x7f722400ae80, clone=SWITCH_FALSE) at src/switch_xml.c:1695 >> #3 0x00007f7249f40f92 in launch_sofia_worker_thread >> (profile=profile at entry=0x7f71f401a160) at sofia.c:3062 >> #4 0x00007f7249f4ac19 in sofia_profile_thread_run (thread=> out>, obj=0x7f71f401a160) at sofia.c:3385 >> #5 0x00007f7253b06e07 in dummy_worker (opaque=0x7f71f401c258) at >> threadproc/unix/thread.c:151 >> #6 0x00007f7252615494 in start_thread (arg=0x7f7249e23700) at >> pthread_create.c:333 >> #7 0x00007f7252357acf in clone () at >> ../sysdeps/unix/sysv/linux/x86_64/clone.S:97 >> >> >> any help or direction would be appreciated ... thank! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > Anthony Minessale II > Founder, FreeSWITCH. > http://freeswitch.com > > > https://youtu.be/l_hOxzCt6X4 > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From darshanmody at avaya.com Mon Nov 12 09:47:50 2018 From: darshanmody at avaya.com (Mody, Darshan (Darshan)) Date: Mon, 12 Nov 2018 09:47:50 +0000 Subject: [Freeswitch-users] What is the recommended practice to use uuid_bridge? Message-ID: <25D2EC755404B4409F263AC6D050FEBB36F01559@AZ-FFEXMB03.global.avaya.com> Hi We have a FS which receives inbound calls (IB). Using FS's ESL we originate outbound calls (OB). We want to bridge the media of inbound to outbound calls. Should we be using uuid_bridge command with arguments IB and OB (uuid_bridge IB OB) or should we be using (uuid_bridge OB IB). Thanks Darshan -------------- next part -------------- An HTML attachment was scrubbed... URL: From freeswitch-users at funke.net Mon Nov 12 10:04:07 2018 From: freeswitch-users at funke.net (Daniel) Date: Mon, 12 Nov 2018 11:04:07 +0100 Subject: [Freeswitch-users] Problem with innovaphone devices In-Reply-To: References: <72989e0e-282e-b1f7-2e6a-34243b8f7f80@funke.net> Message-ID: <093ad03a-01ab-d1a5-286b-1fdf83caae29@funke.net> Hi Sergey, thank you for this advise. In the telephone I found no option to disable ICE but in the gateway. With ICE disabled in the gateway the call still fails. Do you have other ideas what to change? https://pastebin.freeswitch.org/view/fddd86d4 Best regards Daniel On 11/11/18 6:34 PM, Sergey Safarov wrote: > Try disable ICE protocol on device. > > вс, 11 нояб. 2018 г. в 17:35, Daniel >: > > Dear all, > > I'm working with FS since a week. Hope, it's ok to use it in > FusionPBX > ;-) I found out many cool things so far but > with one thing I'm stuck. > > I'm not able to initiate a connection from innovaphone devices (1 > deskphone IP240 and 1 PBX/GW IP800). Calling to > these innovaphones is ok. From other devices to FS and softphones > I can > connect too. > > In my first try from the innovaphones I got the message > > 2018-11-10 23:41:58.763840 [WARNING] switch_core_media.c:4181 NO > candidate ACL defined, Defaulting to wan.auto > 2018-11-10 23:41:58.763840 [NOTICE] sofia.c:7774 Hangup > sofia/internal/11 at pbx.[anonym].net [CS_NEW] > [INCOMPATIBLE_DESTINATION] > > I found the solution to put apply-candidate-acl=localnet.auto to the > internal SIP profile. I don't understand why > this was necessary only for the innovaphones and not for other > devices > but after changing this I can establish a > connection but cannot hear anything on both ends. After a few seconds > the connection is disconnected automatically. > > Here's the log. I hope, someone can help > > https://pastebin.freeswitch.org/view/db088714 > > Best regards > Daniel > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Mon Nov 12 10:11:06 2018 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Mon, 12 Nov 2018 10:11:06 +0000 Subject: [Freeswitch-users] Audio interfacing to POTS port for WebRTC In-Reply-To: References: Message-ID: No-one any ideas? From: Rick Jarvis Reply: Rick Jarvis Date: 8 November 2018 at 16:09:52 To: FreeSWITCH Help Subject:  Audio interfacing to POTS port for WebRTC Strange request, so forgive me, but I’m looking to connect a PC running FS Verto/WebRTC to a telephone balancing unit. So basically, stereo audio in and out from a PC, to plug into an FXO port on a mixing console. The line switching on the console is academic as it will be permanently connected, and any dialling controlled from WebRTC. Appreciate this is a more of an electronics question, but if anyone knows a way to achieve this, someone on here will! Or maybe there’s a better way that I haven’t thought of!? Thanks R _________________________________________________________________________ Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Mon Nov 12 18:43:26 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Mon, 12 Nov 2018 21:43:26 +0300 Subject: [Freeswitch-users] Problem with innovaphone devices In-Reply-To: References: <72989e0e-282e-b1f7-2e6a-34243b8f7f80@funke.net> Message-ID: Caller try use crypto over UDP transport. 1. a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:X0jxiZ/k7DPggObqIRzhtGG43JCiVNd94vDSguFd|2^31 2. a=fingerprint:sha-256 A9:4C:E4:69:65:05:D5:43:C7:E5:0A:99:E5:5B:2E:01:80:EE:81:49:C0:C3:47:3C:10:66:B5:FF:8B:5D:BA:5F Your FreeSwitch send SDP with duplicated audio media 1. m=audio 21718 RTP/AVP 8 101 13 2. a=rtpmap:8 PCMA/8000 3. a=rtpmap:101 telephone-event/8000 4. a=fmtp:101 0-16 5. a=rtpmap:13 CN/8000 6. a=ptime:30 7. a=sendrecv 8. m=audio 21718 RTP/AVP 8 101 13 9. a=rtpmap:8 PCMA/8000 10. a=rtpmap:101 telephone-event/8000 11. a=fmtp:101 0-16 12. a=rtpmap:13 CN/8000 13. a=ptime:20 14. a=sendrecv To fix need: 1) disable encryption on caller side; 2) upgrade FreeSwitch to 1.6.20 or 1.8.2 version Sergey пн, 12 нояб. 2018 г. в 20:02, Daniel : > Hi Sergey, > > thank you for this advise. In the telephone I found no option to disable > ICE but in the gateway. With ICE disabled in the gateway the call still > fails. Do you have other ideas what to change? > > https://pastebin.freeswitch.org/view/fddd86d4 > > Best regards > > Daniel > > > On 11/11/18 6:34 PM, Sergey Safarov wrote: > > Try disable ICE protocol on device. > > вс, 11 нояб. 2018 г. в 17:35, Daniel : > >> Dear all, >> >> I'm working with FS since a week. Hope, it's ok to use it in FusionPBX >> ;-) I found out many cool things so far but >> with one thing I'm stuck. >> >> I'm not able to initiate a connection from innovaphone devices (1 >> deskphone IP240 and 1 PBX/GW IP800). Calling to >> these innovaphones is ok. From other devices to FS and softphones I can >> connect too. >> >> In my first try from the innovaphones I got the message >> >> 2018-11-10 23:41:58.763840 [WARNING] switch_core_media.c:4181 NO >> candidate ACL defined, Defaulting to wan.auto >> 2018-11-10 23:41:58.763840 [NOTICE] sofia.c:7774 Hangup >> sofia/internal/11 at pbx.[anonym].net [CS_NEW] >> [INCOMPATIBLE_DESTINATION] >> >> I found the solution to put apply-candidate-acl=localnet.auto to the >> internal SIP profile. I don't understand why >> this was necessary only for the innovaphones and not for other devices >> but after changing this I can establish a >> connection but cannot hear anything on both ends. After a few seconds >> the connection is disconnected automatically. >> >> Here's the log. I hope, someone can help >> >> https://pastebin.freeswitch.org/view/db088714 >> >> Best regards >> Daniel >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From andrew at zeta.digital-domain.net Mon Nov 12 21:50:01 2018 From: andrew at zeta.digital-domain.net (Andrew Clayton) Date: Mon, 12 Nov 2018 21:50:01 +0000 Subject: [Freeswitch-users] New FreeSWITCH Community Resources - SignalWire.community In-Reply-To: References: <90cb2618-25d7-fa1a-b113-73e7ff39caa7@madovsky.org> Message-ID: <20181112215001.73ee1613@kappa.digital-domain.net> On Sun, 11 Nov 2018 07:06:33 -0800 Joel Serrano wrote: > Slack is free. (to use) for now... From bjordan at e-teleco.com Tue Nov 13 04:24:39 2018 From: bjordan at e-teleco.com (bjordan at e-teleco.com) Date: Tue, 13 Nov 2018 04:24:39 +0000 Subject: [Freeswitch-users] What is the recommended practice to use uuid_bridge? In-Reply-To: <25D2EC755404B4409F263AC6D050FEBB36F01559@AZ-FFEXMB03.global.avaya.com> References: <25D2EC755404B4409F263AC6D050FEBB36F01559@AZ-FFEXMB03.global.avaya.com> Message-ID: According to the confluence link below, I do not think that it matters as long as at least one of the legs is in an answered state. https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-uuid_bridge From: FreeSWITCH-users On Behalf Of Mody, Darshan (Darshan) Sent: Monday, November 12, 2018 1:48 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] What is the recommended practice to use uuid_bridge? Hi We have a FS which receives inbound calls (IB). Using FS's ESL we originate outbound calls (OB). We want to bridge the media of inbound to outbound calls. Should we be using uuid_bridge command with arguments IB and OB (uuid_bridge IB OB) or should we be using (uuid_bridge OB IB). Thanks Darshan -------------- next part -------------- An HTML attachment was scrubbed... URL: From lexxua at gmail.com Tue Nov 13 06:08:21 2018 From: lexxua at gmail.com (Volodymyr Fedorov) Date: Tue, 13 Nov 2018 07:08:21 +0100 Subject: [Freeswitch-users] Problem with innovaphone devices In-Reply-To: <093ad03a-01ab-d1a5-286b-1fdf83caae29@funke.net> References: <72989e0e-282e-b1f7-2e6a-34243b8f7f80@funke.net> <093ad03a-01ab-d1a5-286b-1fdf83caae29@funke.net> Message-ID: Hi try to configure stun server in the phone. Basically it should help to get list of ip addresses. Br, Volodymyr On Mon, 12 Nov 2018, 17:34 Daniel Hi Sergey, > > thank you for this advise. In the telephone I found no option to disable > ICE but in the gateway. With ICE disabled in the gateway the call still > fails. Do you have other ideas what to change? > > https://pastebin.freeswitch.org/view/fddd86d4 > > Best regards > > Daniel > > > On 11/11/18 6:34 PM, Sergey Safarov wrote: > > Try disable ICE protocol on device. > > вс, 11 нояб. 2018 г. в 17:35, Daniel : > >> Dear all, >> >> I'm working with FS since a week. Hope, it's ok to use it in FusionPBX >> ;-) I found out many cool things so far but >> with one thing I'm stuck. >> >> I'm not able to initiate a connection from innovaphone devices (1 >> deskphone IP240 and 1 PBX/GW IP800). Calling to >> these innovaphones is ok. From other devices to FS and softphones I can >> connect too. >> >> In my first try from the innovaphones I got the message >> >> 2018-11-10 23:41:58.763840 [WARNING] switch_core_media.c:4181 NO >> candidate ACL defined, Defaulting to wan.auto >> 2018-11-10 23:41:58.763840 [NOTICE] sofia.c:7774 Hangup >> sofia/internal/11 at pbx.[anonym].net [CS_NEW] >> [INCOMPATIBLE_DESTINATION] >> >> I found the solution to put apply-candidate-acl=localnet.auto to the >> internal SIP profile. I don't understand why >> this was necessary only for the innovaphones and not for other devices >> but after changing this I can establish a >> connection but cannot hear anything on both ends. After a few seconds >> the connection is disconnected automatically. >> >> Here's the log. I hope, someone can help >> >> https://pastebin.freeswitch.org/view/db088714 >> >> Best regards >> Daniel >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul.muaddib83 at gmail.com Tue Nov 13 07:06:41 2018 From: paul.muaddib83 at gmail.com (Paul Muaddib) Date: Tue, 13 Nov 2018 08:06:41 +0100 Subject: [Freeswitch-users] Since recently no ring back Message-ID: Hello Freeswitch Users, Since recently I am not getting ring back singles for some phones that do inbound calls. Voice connection is working in both direction when the call is established. When someone calls where the ring back is not working then there is only a short beep and then silence until someone answers the phone. Without bridge is is working. Freeswitch sits behinde nat. Fresswitch Version 1.8.2 Best regards, Paul. -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmarand at prosodie.com Tue Nov 13 07:55:18 2018 From: rmarand at prosodie.com (MARAND, Remi) Date: Tue, 13 Nov 2018 07:55:18 +0000 Subject: [Freeswitch-users] One way audio case between FS and Chrome on WebRTC In-Reply-To: References: Message-ID: Hello, I finally was able to solve my problem. For information, on this trouble, the parameter : RTCConstraints: {"optional": [{'DtlsSrtpKeyAgreement': 'true'}]} Is to add at the good place in jquery.FSRTC.js (or verto-min.js (I think that's already the case in this script)) i did not find how to code in the sip.js version but it should be possible. Thanks to those who answered my question, and sorry for the 3 Mb of pcap file i sent to the user-list !!! Perhaps this DtlsSrtpKeyAgreement parameter role should be added and explain in the Verto/WebRTC examples availables on Websites, i suppose that in 2014, it was not mandatory but now with the lasts versions of Chrome and FF it seems to be. Regards. Remi Marand. rmarand at prosodie.com +33687725325. De : MARAND, Remi Envoyé : lundi 5 novembre 2018 18:01 À : freeswitch-users at lists.freeswitch.org Objet : One way audio case between FS and Chrome on WebRTC Hello, I am trying to validate FS as a SIP to WebRTC Gateway in our lab environment. I started in middle October and have good result on it, but i cannot understand this One Way Audio trouble. I must thank the Freeswitch team and contributors for this very impressive work. FreeSWITCH Version 1.8.1-2-4f54cff36a~64bit (-2-4f54cff36a 64bit) On system: SMP Debian 4.9.110-3+deb9u6 (2018-10-08) x86_64 GNU/Linux Openssl version : OpenSSL 1.1.0f 25 May 2017 Chrome version: 69.0 (I tried with different version and with Firefox with the same trouble). The wss part is ok with sip.js and verto.js The Ice negotiation is ok, I use sometimes local networks and sometimes web, I have had to authorize networks in the candidate ACL and domain ACL (acl.conf.xml) The result is the same on both topology. DTLS negotiation is OK, and there is UDP streams between Chrome (or Firefox) and FS in both ways. There is no "audible" audio in the direction from FS to Chrome, the other direction is OK. The simplest test is to call the 5000 number from the Chrome client, I send you a paste bin and pcap trace for this call. Should you give me information element to progress on this, what is really mandatory in the sip_profile/internal.xml and external.xml files, and in directory/default/1000.xml for a WebRTC call ?? What should be the good options in fs_cli to see if the encryption of RTP packets is ok or not.. ? Do you think that I have to reinstall a Freeswitch from the current branch ? @IP for FS : 192.168.145.67 @IP for Chrome : 10.70.54.43 Link on the pastebin : https://pastebin.freeswitch.org/view/09a72087 I have a pcap on the same call that I can provide (3 Mb) if necessary.. Thank you for helping me !! Best regards. [Prosodie-signature] Rémi Marand - Product Owner - Pod Connect. PROSODIE - Marketing & Produit Tél. : +33 (0)1.46.84.12.77 / 06.87.72.53.25 rmarand at prosodie.com This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 3368 bytes Desc: image001.gif URL: From backes at solthor.de Tue Nov 13 19:15:14 2018 From: backes at solthor.de (Ulrich Backes) Date: Tue, 13 Nov 2018 20:15:14 +0100 Subject: [Freeswitch-users] How to avoid transcoding? In-Reply-To: <20181107182634.GA2348@darth.lan> References: <380673553.633982.1541595357082@webmail.strato.de> <20181107182634.GA2348@darth.lan> Message-ID: <000001d47b85$35441c70$9fcc5550$@solthor.de> Hi all, I just want to inform you about the solution of my problem: The principal point: I had to remove the ringback (local_extension in dialplan). Thank's for the advise and thanks for all other advise because I could learn from it. Thank you very much. Kind regards Uli -----Ursprüngliche Nachricht----- Von: Sebastian Kemper Gesendet: Mittwoch, 7. November 2018 19:27 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] How to avoid transcoding? On Wed, Nov 07, 2018 at 01:55:57PM +0100, Ulrich Backes wrote: > The settings: > vars.xml > data="media_mix_inbound_outbound_codecs=false"/> > data="global_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/> > > internal.xml > name="inbound-late-negotiation" value="true"/> name="disable-transcoding" value="true"/> > > name="inbound-bypass-media" value="false"/> name="inbound-codec-prefs" value="$${global_codec_prefs}"/> name="outbound-codec-prefs" value="$${global_codec_prefs}"/> > > default.xml (Using the demo-Dialplan) > > ... > > Using FS v 1.8 Hello Uli, FS (.111) starts generating ringback and decides to use G722 on A leg. So remove the ringback for starters. Also, I'd leave "inbound-codec-negotiation" at default. "disable-transcoding" should also not be set, as this is for early negotiation, not late, IIRC. Probably it does nothing when late negotiation is used, but better just remove it. You also need in your dialplan. This is all written down here: https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation Kind regards, Seb --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. https://www.avast.com/antivirus From darshanmody at avaya.com Tue Nov 13 21:27:18 2018 From: darshanmody at avaya.com (Mody, Darshan (Darshan)) Date: Tue, 13 Nov 2018 21:27:18 +0000 Subject: [Freeswitch-users] Hangup after bridge Message-ID: <25D2EC755404B4409F263AC6D050FEBB36F03A20@AZ-FFEXMB03.global.avaya.com> Hi I find below documentation for hangup after bridge hangup_after_bridge BOOLEAN Controls what happens to a calling (A) party when in a bridge state and the called (B) party hangs up. If true the dialplan will stop processing and the A leg will be terminated when the B leg terminates. If false (default) the dialplan continues to be processed after the B leg terminates. This is checked after park_after_bridge and transfer_after_bridge. In the below code within switch_ivr_bridge.c we are checking park_after_bridge transfer after bridge (highlighted). However we are not checking for hangup_after_bridge setting and hanging up the channel. Is there any reason for missing the check of hangup_after_bridge flag before hanging up the channel? static switch_status_t audio_bridge_on_exchange_media(switch_core_session_t *session) { switch_channel_t *channel = switch_core_session_get_channel(session); switch_ivr_bridge_data_t *bd = switch_channel_get_private(channel, "_bridge_"); switch_channel_state_t state; const char *var; if (bd) { switch_channel_set_private(channel, "_bridge_", NULL); if (bd->session == session && *bd->b_uuid) { audio_bridge_thread(NULL, (void *) bd); switch_core_session_reset(session, SWITCH_TRUE, SWITCH_TRUE); } else { switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER); } } else { switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER); } switch_channel_clear_state_handler(channel, &audio_bridge_peer_state_handlers); state = switch_channel_get_state(channel); if (!switch_channel_test_flag(channel, CF_TRANSFER) && !switch_channel_test_flag(channel, CF_REDIRECT) && !switch_channel_test_flag(channel, CF_XFER_ZOMBIE) && bd && !bd->clean_exit && state != CS_PARK && state != CS_ROUTING && state == CS_EXCHANGE_MEDIA && !switch_channel_test_flag(channel, CF_INNER_BRIDGE)) { if (state < CS_HANGUP && switch_true(switch_channel_get_variable(channel, SWITCH_PARK_AFTER_BRIDGE_VARIABLE))) { switch_ivr_park_session(session); return SWITCH_STATUS_FALSE; } else if (state < CS_HANGUP && (var = switch_channel_get_variable(channel, SWITCH_TRANSFER_AFTER_BRIDGE_VARIABLE))) { transfer_after_bridge(session, var); return SWITCH_STATUS_FALSE; } if (switch_channel_test_flag(channel, CF_INTERCEPTED)) { switch_channel_clear_flag(channel, CF_INTERCEPT); switch_channel_clear_flag(channel, CF_INTERCEPTED); return SWITCH_STATUS_FALSE; } else { if (switch_channel_test_flag(channel, CF_INTERCEPT)) { switch_channel_hangup(channel, SWITCH_CAUSE_PICKED_OFF); } else { if (!switch_channel_test_flag(channel, CF_ANSWERED)) { int x = 0; if (switch_channel_execute_on(channel, "execute_on_orphaned_bleg") == SWITCH_STATUS_SUCCESS) { x++; } if (switch_channel_api_on(channel, "api_on_orphaned_bleg") == SWITCH_STATUS_SUCCESS) { x++; } if (!x) { switch_channel_hangup(channel, SWITCH_CAUSE_ORIGINATOR_CANCEL); } } else { switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING); } } } } if (switch_channel_get_state(channel) == CS_EXCHANGE_MEDIA) { switch_channel_set_variable(channel, "park_timeout", "3"); switch_channel_set_state(channel, CS_PARK); } return SWITCH_STATUS_FALSE; } Thanks Darshan -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Nov 13 10:59:04 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 13 Nov 2018 10:59:04 +0000 Subject: [Freeswitch-users] Problem with innovaphone devices In-Reply-To: References: <72989e0e-282e-b1f7-2e6a-34243b8f7f80@funke.net> Message-ID: Check out the enabled codecs, enable just one codec you know fs supports, and test again. On Mon, 12 Nov 2018 at 16:59, Daniel wrote: > Hi Sergey, > > thank you for this advise. In the telephone I found no option to disable > ICE but in the gateway. With ICE disabled in the gateway the call still > fails. Do you have other ideas what to change? > > https://pastebin.freeswitch.org/view/fddd86d4 > > Best regards > > Daniel > > > On 11/11/18 6:34 PM, Sergey Safarov wrote: > > Try disable ICE protocol on device. > > вс, 11 нояб. 2018 г. в 17:35, Daniel : > >> Dear all, >> >> I'm working with FS since a week. Hope, it's ok to use it in FusionPBX >> ;-) I found out many cool things so far but >> with one thing I'm stuck. >> >> I'm not able to initiate a connection from innovaphone devices (1 >> deskphone IP240 and 1 PBX/GW IP800). Calling to >> these innovaphones is ok. From other devices to FS and softphones I can >> connect too. >> >> In my first try from the innovaphones I got the message >> >> 2018-11-10 23:41:58.763840 [WARNING] switch_core_media.c:4181 NO >> candidate ACL defined, Defaulting to wan.auto >> 2018-11-10 23:41:58.763840 [NOTICE] sofia.c:7774 Hangup >> sofia/internal/11 at pbx.[anonym].net [CS_NEW] >> [INCOMPATIBLE_DESTINATION] >> >> I found the solution to put apply-candidate-acl=localnet.auto to the >> internal SIP profile. I don't understand why >> this was necessary only for the innovaphones and not for other devices >> but after changing this I can establish a >> connection but cannot hear anything on both ends. After a few seconds >> the connection is disconnected automatically. >> >> Here's the log. I hope, someone can help >> >> https://pastebin.freeswitch.org/view/db088714 >> >> Best regards >> Daniel >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Nov 13 11:06:46 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 13 Nov 2018 11:06:46 +0000 Subject: [Freeswitch-users] sip capture In-Reply-To: <545EA5E6-2EB6-4B60-92E3-FEA79E664E09@gmx.net> References: <545EA5E6-2EB6-4B60-92E3-FEA79E664E09@gmx.net> Message-ID: Do you get any error on the fs console/log? On Fri, 9 Nov 2018 at 19:52, Markus Bönke wrote: > Hello, > > is sip-capture supported in 1.8.2? I’ve enabled sip capturing (as > recommended at > https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch > ) > > > wrote: > On Sun, 11 Nov 2018 07:06:33 -0800 > Joel Serrano wrote: > > > Slack is free. > > (to use) for now... > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul.muaddib83 at gmail.com Thu Nov 15 19:32:43 2018 From: paul.muaddib83 at gmail.com (Paul Muaddib) Date: Thu, 15 Nov 2018 20:32:43 +0100 Subject: [Freeswitch-users] NAT Problem with UPNP only for RTP - SIP tcp keept open Message-ID: Hi at the moment I have the setup Internal.xml ext-rtp-ip = auto-nat ext-sip-ip = auto-nat External.xml ext-rtp-ip = “my_local_freeswitch_ip” ext-sip-ip = auto-nat RTP ports (only) are begin opened by UPNP and SIP via TCP is being keep open with “expire-seconds=600” and “register=true” (Is there an alternative for the expire-seconds ? nat-options-ping is only for endpoints registering to freeswitch, right?) The reason why I am doing this is, that I dont want to open up a SIP Port in the Firewall. But this setup up doesn’t seem right. Audio is working, inbound and outbound, both ways. But the echo application is not. When I change External.xml ext-rtp-ip = “auto-nat” I get one-way audio for inbound calls for the callee. The caller can be heard. I tried to analyses it in wireshark, but I don’t understand it. The sip provider keeps sending INVITE messages while freeswitch is confirming it with OK but keeps going on until the connection breaks. Thank you for your help -------------- next part -------------- An HTML attachment was scrubbed... URL: From jpablolorenzetti at gmail.com Thu Nov 15 22:49:33 2018 From: jpablolorenzetti at gmail.com (Juan Pablo L) Date: Thu, 15 Nov 2018 16:49:33 -0600 Subject: [Freeswitch-users] dynamic gateways Message-ID: Hello, i m trying to load gateway information for profile dynamically, at the moment i am trying to do it in the directory section, when the profile is reload/rescan/restarted, upon directory request i m answering this xml:
but it is not working, i have tried like dozens of times with different parameters, values etc but none seems to work , even though the profile is reloaded/rescanned/restarted successfully the gateway(s) do not show in "sofia status" and when i attempt to make a call, naturally, i get an "invalid gateway" error, does anyone know what is the correct format of the xml to be answered ? ... other parameters i have tried are domain,profile,profiles and others alike and all possible combinations of them all .... and yes the gateway information is correct it works if loaded from xml file ... thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Fri Nov 16 07:11:30 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Fri, 16 Nov 2018 16:11:30 +0900 Subject: [Freeswitch-users] Since recently no ring back In-Reply-To: References: Message-ID: Hello, Try to set *ringback* without ring_ready. Or *ring_ready* without ringback. Or even neither of them. Regards, Alex On Thu, Nov 15, 2018 at 8:16 PM Paul Muaddib wrote: > Hello Freeswitch Users, > > Since recently I am not getting ring back singles for some phones that do > inbound calls. Voice connection is working in both direction when the call > is established. When someone calls where the ring back is not working then > there is only a short beep and then silence until someone answers the phone. > > > > > > Without bridge is is working. > > > > > > Freeswitch sits behinde nat. Fresswitch Version 1.8.2 > > Best regards, > Paul. > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Fri Nov 16 07:17:26 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Fri, 16 Nov 2018 16:17:26 +0900 Subject: [Freeswitch-users] RTP Update In-Reply-To: References: Message-ID: If you are using SIP for initial call establishment, not it's not. On Thu, Nov 15, 2018 at 11:08 PM Vishal Dalsania wrote: > Is it possible to update RTP IP address after call is established without > sending any SIP message? > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mickael at winlux.fr Fri Nov 16 08:29:21 2018 From: mickael at winlux.fr (Mickael Hubert) Date: Fri, 16 Nov 2018 09:29:21 +0100 Subject: [Freeswitch-users] Generate uniqueid like asterisk Message-ID: Hi all, currently, I use asterisk uniqueid to lot of internals processes Ex: 001-1542296620.1319230 I want to change asterisk by FS, but I have to keep this uniqueid format. I know the format of the uniqueid : [server id]-[timestamp].[random integer] How can I generate it from FS please ? I already seen with mod_dialplan_asterisk module, but I'm not sure if can I use only UNIQUEID variable (I'm not need the asterisk dialplan) Or maybe use other variables Ex: unixtimestamp and create random interger... thanks in advance ++ -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul.muaddib83 at gmail.com Fri Nov 16 10:44:49 2018 From: paul.muaddib83 at gmail.com (Paul Muaddib) Date: Fri, 16 Nov 2018 11:44:49 +0100 Subject: [Freeswitch-users] Since recently no ring back In-Reply-To: References: Message-ID: Hello Alexey, I had to disable early media by not setting ringback in the dialplan. As soon as I send any early media I get silence. Regards, Paul Am Fr., 16. Nov. 2018 um 09:01 Uhr schrieb Alexey Sibyakin < alex at freeswitch.com>: > Hello, > > Try to set *ringback* without ring_ready. Or *ring_ready* without > ringback. Or even neither of them. > > Regards, > > Alex > > On Thu, Nov 15, 2018 at 8:16 PM Paul Muaddib > wrote: > >> Hello Freeswitch Users, >> >> Since recently I am not getting ring back singles for some phones that do >> inbound calls. Voice connection is working in both direction when the call >> is established. When someone calls where the ring back is not working then >> there is only a short beep and then silence until someone answers the phone. >> >> >> >> >> >> Without bridge is is working. >> >> >> >> >> >> Freeswitch sits behinde nat. Fresswitch Version 1.8.2 >> >> Best regards, >> Paul. >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Alex Sibyakin | Support Engineer > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > Email: alex at freeswitch.com > Website: https://www.FreeSWITCH.com > Need commercial support? Contact sales at freeswitch.com for details. > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From morfair at gmail.com Fri Nov 16 10:47:28 2018 From: morfair at gmail.com (morfair at gmail.com) Date: Fri, 16 Nov 2018 13:47:28 +0300 Subject: [Freeswitch-users] Monitoring mod_odbc_cdr Message-ID: <581ddce4-e557-c9b2-8efa-8e136b51cce3@gmail.com> Hello all! I use mod_odbc_cdr for store CDR in MySQL for billing. Today I saw that no records in DB in 2 weeks. IT IS FAIL!! I went to fs_cli and type `reload mod_odbc_cdr`. After that all work again. But I lost records in two weeks. How to monitor mod_odbc_cdr and prevent risks with its? From francesco at delagarda.com Fri Nov 16 13:08:20 2018 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Fri, 16 Nov 2018 14:08:20 +0100 Subject: [Freeswitch-users] Monitoring mod_odbc_cdr In-Reply-To: <581ddce4-e557-c9b2-8efa-8e136b51cce3@gmail.com> References: <581ddce4-e557-c9b2-8efa-8e136b51cce3@gmail.com> Message-ID: <009701d47dad$7464e310$5d2ea930$@delagarda.com> These are my notes on how I got it working.. Hope they help! NOTE: I just set it up, but don’t really use. It works, but NOWHERE is a timestamp/date time field!!! I think you just have to play with config and db tables to setup F https://freeswitch.org/confluence/display/FREESWITCH/Using+ODBC+in+the+core // mod-odbc-cdr apt-get install libmyodbc apt-get install freeswitch-mod-odbc-cdr // SETUP odbc.. /etc/odbc.ini for MySQL [cdr] Description=description of your DSN Driver=myodbc_mysql Server=localhost Port=3306 Socket=/var/run/mysqld/mysqld.sock Database=cdr Username=******* Password=******** Option=3 ReadOnly=No Test with isql -v cdr SQL> show tables; +-----------------------------------------------------------------+ | Tables_in_cdr | +-----------------------------------------------------------------+ | cdr_table_a_leg | | cdr_table_b_leg | | cdr_table_both | +-----------------------------------------------------------------+ SQLRowCount returns 3 3 rows fetched SQL> https://asterisk-pbx.ru/wiki/freeswitch/freeswitch-cdr-viewer to get it to connect with the right use, use ...
-----Original Message----- From: FreeSWITCH-users On Behalf Of morfair at gmail.com Sent: venerdì 16 novembre 2018 11:47 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Monitoring mod_odbc_cdr Hello all! I use mod_odbc_cdr for store CDR in MySQL for billing. Today I saw that no records in DB in 2 weeks. IT IS FAIL!! I went to fs_cli and type `reload mod_odbc_cdr`. After that all work again. But I lost records in two weeks. How to monitor mod_odbc_cdr and prevent risks with its? _________________________________________________________________________ Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com From fcastro at astelco.com.ar Fri Nov 16 13:17:48 2018 From: fcastro at astelco.com.ar (Federico Castro) Date: Fri, 16 Nov 2018 10:17:48 -0300 Subject: [Freeswitch-users] Debian and Centos not officialy supported by server In-Reply-To: References: Message-ID: Thanks Andrew, I will try that out. El mié., 7 nov. 2018 a las 9:27, Andrew Cassidy (< andrew at cassidywebservices.co.uk>) escribió: > If I remember correctly, the raid controller in the DL120 is a 'fakeraid' > one (software raid where the configuration is held in system settings). You > may have better luck disabling the controller and settings the SATA ports > in AHCI mode and use Linux software raid. > > If you still have issues, try disabling UEFI boot (change boot mode to > Legacy BIOS) as this solves a number of other issues on Gen9/Gen10 with > Linux. > > As always, get hold of the latest firmware updates too, the most recently > ProLiant service pack is from September 2018. > > Kind regards, > -- > *Andrew Cassidy BSc (Hons) MBCS* > Managing Director > > 0330 44 55 960 <+443304455960> andrew at cassidyweb.co.uk > www.cassidyweb.co.uk > > On Tue, 6 Nov 2018, 22:30 Federico Castro, wrote: > >> Thanks to all for sharing your experiences and tips. >> I agree, I think the problem could be if you need some kind of assistance >> from the vendor. >> >> Thanks again. >> >> El mar., 6 nov. 2018 a las 14:37, Volodymyr Fedorov () >> escribió: >> >>> Hi guys, well CentOS is not RHEL at least is not supported by RedHat/IBM >>> directly. For Debian you are on your own for sure. The problem comes when >>> you have a problem and basically the first thing which you hear when call >>> hotline in case of HW-issue, this os is not supported. >>> But I think you still can build FreeSWITCH rpm in Oracle/RHEL/SuSE Linux. >>> BR, >>> Volodymyr >>> >>> On Mon, Nov 5, 2018 at 11:08 PM Sergey Safarov >>> wrote: >>> >>>> CentOS is equal to RHEL dist. >>>> I sure HP is supports RHEL. >>>> >>>> Sergey >>>> >>>> пн, 5 нояб. 2018 г. в 20:30, Chad Phillips : >>>> >>>>> I've run both CentOS and Debian on Dell PowerEdge servers for nearly >>>>> 10 years now, with virtually no issues related to the hardware/software >>>>> interface, and neither of those are officially supported by Dell. >>>>> >>>>> However, Dell does officially support many varieties of Linux, >>>>> including RHEL (of which CentOS is basically a clone), and after some >>>>> research I felt comfortable enough with the unofficial support for the >>>>> operating systems I wanted to use. >>>>> >>>>> No idea if that translates to the HP world, though... >>>>> >>>>> On Mon, Nov 5, 2018 at 8:01 AM Federico Castro >>>>> wrote: >>>>> >>>>>> Hi all, I installed FS on a HP DL-120 Gen9 server using Debian but >>>>>> reading server specifications I found that Debian and Centos are not >>>>>> officially supported for that server (and I realized that many others does >>>>>> not support too). >>>>>> Anyway I have everything installed and running except for RAID >>>>>> controller because there’s no drivers for that. >>>>>> So, trying to avoid unexpected behaviors on production I would like >>>>>> to know if any of you have some experience on using hardware that does not >>>>>> officialy support Debian/Centos? >>>>>> >>>>>> Thanks in advance. >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Services >>>>>> sales at freeswitch.com >>>>>> https://freeswitch.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> https://freeswitch.com/oss >>>>>> https://freeswitch.org/confluence >>>>>> https://cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> https://freeswitch.com >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Services >>>>> sales at freeswitch.com >>>>> https://freeswitch.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> https://freeswitch.com/oss >>>>> https://freeswitch.org/confluence >>>>> https://cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> https://freeswitch.com >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Services >>>> sales at freeswitch.com >>>> https://freeswitch.com >>>> >>>> Official FreeSWITCH Sites >>>> https://freeswitch.com/oss >>>> https://freeswitch.org/confluence >>>> https://cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> >>> >>> -- >>> Best regards, >>> Volodymyr >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Fri Nov 16 13:24:54 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 16 Nov 2018 16:24:54 +0300 Subject: [Freeswitch-users] Monitoring mod_odbc_cdr In-Reply-To: <581ddce4-e557-c9b2-8efa-8e136b51cce3@gmail.com> References: <581ddce4-e557-c9b2-8efa-8e136b51cce3@gmail.com> Message-ID: you can 1) count call in database for last 12 hours using monitoring tools like zabbix. If no calls then rise alarm; 2) configure saving CDR in XML or JSON format on disk. if call is lost in CSL database, then you cann load from CDR files. CDR files older then 2 days may be deleted. пт, 16 нояб. 2018 г. в 16:18, morfair at gmail.com : > Hello all! > > I use mod_odbc_cdr for store CDR in MySQL for billing. Today I saw that > no records in DB in 2 weeks. IT IS FAIL!! > > I went to fs_cli and type `reload mod_odbc_cdr`. After that all work > again. But I lost records in two weeks. > > How to monitor mod_odbc_cdr and prevent risks with its? > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From infos at madovsky.org Fri Nov 16 12:57:51 2018 From: infos at madovsky.org (Madovsky) Date: Fri, 16 Nov 2018 04:57:51 -0800 Subject: [Freeswitch-users] Monitoring mod_odbc_cdr In-Reply-To: <581ddce4-e557-c9b2-8efa-8e136b51cce3@gmail.com> References: <581ddce4-e557-c9b2-8efa-8e136b51cce3@gmail.com> Message-ID: it sounds more a lost connection. did you check the FS logs? On 11/16/2018 2:47 AM, morfair at gmail.com wrote: > Hello all! > > I use mod_odbc_cdr for store CDR in  MySQL for billing. Today I saw > that no records in DB in 2 weeks. IT IS FAIL!! > > I went to fs_cli and type `reload mod_odbc_cdr`. After that all work > again. But I lost records in two weeks. > > How to monitor mod_odbc_cdr and prevent risks with its? > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > From social at bohboh.info Fri Nov 16 18:03:48 2018 From: social at bohboh.info (Social Boh) Date: Fri, 16 Nov 2018 13:03:48 -0500 Subject: [Freeswitch-users] Monitoring mod_odbc_cdr In-Reply-To: References: <581ddce4-e557-c9b2-8efa-8e136b51cce3@gmail.com> Message-ID: Hello, on the odbc_cdr.conf.xml, maybe you can configure: so if database fail you ha ve a file for each call you can process to add to your tables. Regards --- I'm SoCIaL, MayBe El 16/11/2018 a las 08:24, Sergey Safarov escribió: > you can > 1) count call in database for last 12 hours using monitoring tools > like zabbix. If no calls then rise alarm; > 2) configure saving CDR in XML or JSON format on disk. if call is lost > in CSL database, then you cann load from CDR files. CDR files older > then 2 days may be deleted. > > > пт, 16 нояб. 2018 г. в 16:18, morfair at gmail.com > >: > > Hello all! > > I use mod_odbc_cdr for store CDR in  MySQL for billing. Today I > saw that > no records in DB in 2 weeks. IT IS FAIL!! > > I went to fs_cli and type `reload mod_odbc_cdr`. After that all work > again. But I lost records in two weeks. > > How to monitor mod_odbc_cdr and prevent risks with its? > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From allen at praecom.com Fri Nov 16 19:40:12 2018 From: allen at praecom.com (Allen Underdown) Date: Fri, 16 Nov 2018 13:40:12 -0600 Subject: [Freeswitch-users] dialplan xml pre-process include other xml Message-ID: <503907042-74331@mail.praecom.com> I'm having a problem understanding the steps and/or proper formatting of including a xml file inside a "host" xml file within the dial plan. I have limited access to the dial plan, so need to insert a call to a file in a different directory for a customized extension. Example - in /dialplan/somedialplan.xml  
                   
I want to add: so that every time somedialplan.xml is read, custom_dial.xml is included. I've tried adding it after the section name, and after the context, but it's never parsed (according to watching fs_cli). So I can't seem to figure out where to put it and have it parsed. I'm also a bit confused on exactly the XML required in custom_dial.xml. Do I need to specify the context that I want the extension executed in? In custom_dial.xml I've tried using with and without the tags. I've scoured freeswitch.org but can't seem to find the documentation in a way that I can figure this out. Thanks! Allen Underdown 'Progress isn't made by early risers. It's made by lazy men trying to find easier ways to do something.' --- Robert Heinlein -------------- next part -------------- An HTML attachment was scrubbed... URL: From sat at calgaryit.com Sat Nov 17 01:06:08 2018 From: sat at calgaryit.com (George) Date: Fri, 16 Nov 2018 18:06:08 -0700 (MST) Subject: [Freeswitch-users] RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back..... Message-ID: <1610858919.7688.1542416768728.JavaMail.zimbra@calgaryit.com> any way to hack around this issue the carrier is not fixing the issue, any calls to an IVR do do throug after the innitial recorded message, this is especially a problem with 311 calls, this is up in Canada with SHAW Thank you, George From david.villasmil.work at gmail.com Sat Nov 17 23:00:06 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 17 Nov 2018 23:00:06 +0000 Subject: [Freeswitch-users] Memory increase because of multiple instances of freeswitch In-Reply-To: References: <04fc4086-d678-2f65-ca06-059f40b89e22@wirelessmundi.com> Message-ID: I've never seen this... Is it possible the process is being started multiple times? You _can_ run fs multiple times, but only the first would start properly as the rest would try to bind to a port already in use by the first process. There's a parameter on all profiles to shutdown if it can't start, try setting that and see what happens. You should also check your crontab... This is NOT normal FS behaviour, as far as i know. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ᐧ On Fri, Nov 9, 2018 at 7:48 PM António Silva via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: "António Silva" > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Thu, 8 Nov 2018 17:53:03 +0100 > Subject: Re: Memory increase because of multiple instances of freeswitch > Forget to put my unit configuration: > > [Service] > ; service > Type=forking > PIDFile=/run/freeswitch/freeswitch.pid > ExecStart=/usr/bin/freeswitch -ncwait -nonat -scripts /scripts/fs > TimeoutSec=300s > Restart=on-failure > RestartSec=500ms > ; exec > User=root > Group=daemon > LimitCORE=infinity > LimitNOFILE=100000 > LimitNPROC=60000 > LimitSTACK=250000 > LimitRTPRIO=infinity > LimitRTTIME=infinity > IOSchedulingClass=realtime > IOSchedulingPriority=2 > CPUSchedulingPolicy=rr > CPUSchedulingPriority=89 > UMask=0007 > > > > On 08/11/2018 17:11, António Silva wrote: > > Hi all, > > > > I notice a strange behaviour on machine due to increase of memory, > > when i went to see that was the process consuming the memory i notice > > that freeswitch have multiple process running: > > > > root 2543 90.4 4.8 2178716 1587484 ? S > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > > root 7858 0.0 5.0 2289392 1649484 ? SN Oct19 0:00 \_ > > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > > root 30626 0.0 4.4 2172632 1464684 ? SN Oct23 0:00 \_ > > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > > root 4505 0.0 4.9 2336544 1621768 ? SN 10:43 0:00 \_ > > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > > root 22557 0.0 4.9 2336548 1636604 ? SN 11:02 0:00 \_ > > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > > > > > > and today: > > > > root 21823 93.9 4.8 2154128 1581064 ? S > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > > root 18417 0.0 4.9 2247264 1610108 ? SN 11:09 0:00 \_ > > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > > > > > > what could be the cause of this? anyone experience the same? > > > > As for the service everything is running ok, is just that the memory > > keeps increasing.. i restart freewitch when i start to reach my memory > > limit. > > > > > > I'm running fs 1.8.2 on debian jessie. > > > > > -- > Saludos / Regards / Cumprimentos > António Silva > > > > > > ---------- Forwarded message ---------- > From: "António Silva via FreeSWITCH-users" < > freeswitch-users at lists.freeswitch.org> > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Fri, 09 Nov 2018 11:48:33 -0800 (PST) > Subject: Re: [Freeswitch-users] Memory increase because of multiple > instances of freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Nov 17 23:01:54 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 17 Nov 2018 23:01:54 +0000 Subject: [Freeswitch-users] Memory increase because of multiple instances of freeswitch In-Reply-To: References: <04fc4086-d678-2f65-ca06-059f40b89e22@wirelessmundi.com> Message-ID: Also, regarding memory usage, that's how FS works. It takes memory AS NEEDED and simply doesn't returns it, it holds it for future use. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ᐧ On Sat, Nov 17, 2018 at 11:00 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > I've never seen this... > > Is it possible the process is being started multiple times? You _can_ run > fs multiple times, but only the first would start properly as the rest > would try to bind to a port already in use by the first process. > There's a parameter on all profiles to shutdown if it can't start, try > setting that and see what happens. > You should also check your crontab... This is NOT normal FS behaviour, as > far as i know. > > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > ᐧ > > On Fri, Nov 9, 2018 at 7:48 PM António Silva via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> wrote: > >> >> >> >> ---------- Forwarded message ---------- >> From: "António Silva" >> To: FreeSWITCH Users Help >> Cc: >> Bcc: >> Date: Thu, 8 Nov 2018 17:53:03 +0100 >> Subject: Re: Memory increase because of multiple instances of freeswitch >> Forget to put my unit configuration: >> >> [Service] >> ; service >> Type=forking >> PIDFile=/run/freeswitch/freeswitch.pid >> ExecStart=/usr/bin/freeswitch -ncwait -nonat -scripts /scripts/fs >> TimeoutSec=300s >> Restart=on-failure >> RestartSec=500ms >> ; exec >> User=root >> Group=daemon >> LimitCORE=infinity >> LimitNOFILE=100000 >> LimitNPROC=60000 >> LimitSTACK=250000 >> LimitRTPRIO=infinity >> LimitRTTIME=infinity >> IOSchedulingClass=realtime >> IOSchedulingPriority=2 >> CPUSchedulingPolicy=rr >> CPUSchedulingPriority=89 >> UMask=0007 >> >> >> >> On 08/11/2018 17:11, António Silva wrote: >> > Hi all, >> > >> > I notice a strange behaviour on machine due to increase of memory, >> > when i went to see that was the process consuming the memory i notice >> > that freeswitch have multiple process running: >> > >> > root 2543 90.4 4.8 2178716 1587484 ? S> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >> > root 7858 0.0 5.0 2289392 1649484 ? SN Oct19 0:00 \_ >> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >> > root 30626 0.0 4.4 2172632 1464684 ? SN Oct23 0:00 \_ >> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >> > root 4505 0.0 4.9 2336544 1621768 ? SN 10:43 0:00 \_ >> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >> > root 22557 0.0 4.9 2336548 1636604 ? SN 11:02 0:00 \_ >> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >> > >> > >> > and today: >> > >> > root 21823 93.9 4.8 2154128 1581064 ? S> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >> > root 18417 0.0 4.9 2247264 1610108 ? SN 11:09 0:00 \_ >> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >> > >> > >> > what could be the cause of this? anyone experience the same? >> > >> > As for the service everything is running ok, is just that the memory >> > keeps increasing.. i restart freewitch when i start to reach my memory >> > limit. >> > >> > >> > I'm running fs 1.8.2 on debian jessie. >> > >> > >> -- >> Saludos / Regards / Cumprimentos >> António Silva >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: "António Silva via FreeSWITCH-users" < >> freeswitch-users at lists.freeswitch.org> >> To: FreeSWITCH Users Help >> Cc: >> Bcc: >> Date: Fri, 09 Nov 2018 11:48:33 -0800 (PST) >> Subject: Re: [Freeswitch-users] Memory increase because of multiple >> instances of freeswitch >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Mon Nov 19 08:03:56 2018 From: krice at freeswitch.org (Ken Rice) Date: Mon, 19 Nov 2018 02:03:56 -0600 Subject: [Freeswitch-users] Memory increase because of multiple instances of freeswitch In-Reply-To: References: <04fc4086-d678-2f65-ca06-059f40b89e22@wirelessmundi.com> Message-ID: well also keep in mind that depending on how he is looking for freeswitch processes he may be seeing multiple as the threads will show up ad their individual psuedo PID for example in htop Sent from my iPhone > On Nov 17, 2018, at 17:01, David Villasmil wrote: > > Also, regarding memory usage, that's how FS works. It takes memory AS NEEDED and simply doesn't returns it, it holds it for future use. > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > ᐧ > >> On Sat, Nov 17, 2018 at 11:00 PM David Villasmil wrote: >> I've never seen this... >> >> Is it possible the process is being started multiple times? You _can_ run fs multiple times, but only the first would start properly as the rest would try to bind to a port already in use by the first process. >> There's a parameter on all profiles to shutdown if it can't start, try setting that and see what happens. >> You should also check your crontab... This is NOT normal FS behaviour, as far as i know. >> >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> >> ᐧ >> >>> On Fri, Nov 9, 2018 at 7:48 PM António Silva via FreeSWITCH-users wrote: >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: "António Silva" >>> To: FreeSWITCH Users Help >>> Cc: >>> Bcc: >>> Date: Thu, 8 Nov 2018 17:53:03 +0100 >>> Subject: Re: Memory increase because of multiple instances of freeswitch >>> Forget to put my unit configuration: >>> >>> [Service] >>> ; service >>> Type=forking >>> PIDFile=/run/freeswitch/freeswitch.pid >>> ExecStart=/usr/bin/freeswitch -ncwait -nonat -scripts /scripts/fs >>> TimeoutSec=300s >>> Restart=on-failure >>> RestartSec=500ms >>> ; exec >>> User=root >>> Group=daemon >>> LimitCORE=infinity >>> LimitNOFILE=100000 >>> LimitNPROC=60000 >>> LimitSTACK=250000 >>> LimitRTPRIO=infinity >>> LimitRTTIME=infinity >>> IOSchedulingClass=realtime >>> IOSchedulingPriority=2 >>> CPUSchedulingPolicy=rr >>> CPUSchedulingPriority=89 >>> UMask=0007 >>> >>> >>> >>> On 08/11/2018 17:11, António Silva wrote: >>> > Hi all, >>> > >>> > I notice a strange behaviour on machine due to increase of memory, >>> > when i went to see that was the process consuming the memory i notice >>> > that freeswitch have multiple process running: >>> > >>> > root 2543 90.4 4.8 2178716 1587484 ? S>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > root 7858 0.0 5.0 2289392 1649484 ? SN Oct19 0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > root 30626 0.0 4.4 2172632 1464684 ? SN Oct23 0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > root 4505 0.0 4.9 2336544 1621768 ? SN 10:43 0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > root 22557 0.0 4.9 2336548 1636604 ? SN 11:02 0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > >>> > >>> > and today: >>> > >>> > root 21823 93.9 4.8 2154128 1581064 ? S>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > root 18417 0.0 4.9 2247264 1610108 ? SN 11:09 0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > >>> > >>> > what could be the cause of this? anyone experience the same? >>> > >>> > As for the service everything is running ok, is just that the memory >>> > keeps increasing.. i restart freewitch when i start to reach my memory >>> > limit. >>> > >>> > >>> > I'm running fs 1.8.2 on debian jessie. >>> > >>> > >>> -- >>> Saludos / Regards / Cumprimentos >>> António Silva >>> >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: "António Silva via FreeSWITCH-users" >>> To: FreeSWITCH Users Help >>> Cc: >>> Bcc: >>> Date: Fri, 09 Nov 2018 11:48:33 -0800 (PST) >>> Subject: Re: [Freeswitch-users] Memory increase because of multiple instances of freeswitch >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeremy.laine at m4x.org Mon Nov 19 08:17:51 2018 From: jeremy.laine at m4x.org (=?UTF-8?Q?Jeremy_Lain=c3=a9?=) Date: Mon, 19 Nov 2018 09:17:51 +0100 Subject: [Freeswitch-users] One way audio case between FS and Chrome on WebRTC In-Reply-To: References: Message-ID: <21c60857-88ba-c77e-f624-57d6661a12f2@m4x.org> That is very odd, the default behaviour in browsers *is* to use DTLS for SRTP keying instead of the legacy SDES. In fact SDES's status is "MUST NOT implement" since 2013, and the DtlsSrtpKeyAgreement constraint is scheduled for retirement: https://bugs.chromium.org/p/chromium/issues/detail?id=804275 Concerning SIP.js I have never experienced the need to set this constraint, it works fine against freeswitch without it. Jeremy On 11/13/18 8:55 AM, MARAND, Remi wrote: > > Hello, > >   > > I finally was able to solve my problem. > >   > > For information, on this trouble, the parameter : > > /RTCConstraints: {"optional": [{'DtlsSrtpKeyAgreement': 'true'}]}/ > > Is to add at the good place in jquery.FSRTC.js (or verto-min.js (I > think that's already the case in this script)) i did not find how to > code in the sip.js version but it should be possible. > >   > > Thanks to those who answered my question, and sorry for the 3 Mb of > pcap file i sent to the user-list !!! > >   > > Perhaps this DtlsSrtpKeyAgreement parameter role should be added and > explain in the Verto/WebRTC examples availables on Websites, i suppose > that in 2014, it was not mandatory but now with the lasts versions of > Chrome and FF it seems to be. > >   > > Regards. > >   > > Remi Marand. > > rmarand at prosodie.com > > +33687725325. > >   > > *De :*MARAND, Remi > *Envoyé :* lundi 5 novembre 2018 18:01 > *À :* freeswitch-users at lists.freeswitch.org > *Objet :* One way audio case between FS and Chrome on WebRTC > >   > > Hello, > >   > > I am trying to validate FS as a SIP to WebRTC Gateway in our lab > environment. > > I started in middle October and have good result on it, but i cannot > understand this One Way Audio trouble. > >   > > I must thank the Freeswitch team and contributors for this very > impressive work. > >   > > FreeSWITCH Version 1.8.1-2-4f54cff36a~64bit (-2-4f54cff36a 64bit) > > On system: SMP Debian 4.9.110-3+deb9u6 (2018-10-08) x86_64 GNU/Linux > > Openssl version : OpenSSL 1.1.0f  25 May 2017 > > Chrome version: 69.0 (I tried with different version and with Firefox > with the same trouble). > >   > > The wss part is ok with sip.js and verto.js > >   > > The Ice negotiation is ok, I use sometimes local networks and > sometimes web, I have had to authorize networks in the candidate ACL > and domain ACL (acl.conf.xml) The result is the same on both topology. > >   > > DTLS negotiation is OK, and there is UDP streams between Chrome (or > Firefox) and FS in both ways. > >   > > There is no “audible” audio in the direction from FS to Chrome, the > other direction is OK. > >   > > The simplest test is to call the 5000 number from the Chrome client, I > send you a paste bin and pcap trace for this call. > >   > > Should you give me information element to progress on this, what is > really mandatory in the sip_profile/internal.xml and external.xml > files, and in directory/default/1000.xml for a WebRTC call ?? What > should be the good options in fs_cli to see if the encryption of RTP > packets is ok or not.. ? > > Do you think that I have to reinstall a Freeswitch from the current > branch ? > >   > > @IP for FS : 192.168.145.67 > > @IP for Chrome : 10.70.54.43 > >   > > Link on the pastebin : https://pastebin.freeswitch.org/view/09a72087 > >   > > I have a pcap on the same call that I can provide (3 Mb) if necessary.. > >   > > Thank you for helping me !! > >   > > Best regards. > >   > > Prosodie-signature > > > > *Rémi Marand – Product Owner – Pod Connect. > *PROSODIE – Marketing & Produit > Tél. : +33 (0)1.46.84.12.77 / 06.87.72.53.25 > rmarand at prosodie.com > >   > >   > >   > >   > >   > >   > > This message contains information that may be privileged or > confidential and is the property of the Capgemini Group. It is > intended only for the person to whom it is addressed. If you are not > the intended recipient, you are not authorized to read, print, retain, > copy, disseminate, distribute, or use this message or any part > thereof. If you receive this message in error, please notify the > sender immediately and delete all copies of this message. > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 3368 bytes Desc: not available URL: From asilva at wirelessmundi.com Mon Nov 19 11:04:31 2018 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Mon, 19 Nov 2018 12:04:31 +0100 Subject: [Freeswitch-users] Memory increase because of multiple instances of freeswitch In-Reply-To: References: <04fc4086-d678-2f65-ca06-059f40b89e22@wirelessmundi.com> Message-ID: <629c4342-c411-181a-7da3-9f667a4785f7@wirelessmundi.com> Hi David, Thank for the reply. Yes, i've set on sofia profile to shutdown on fail: No, is not started multiple times, i also don't have any cron that would start it, is internally that the forking is happing, also the sofia profiles are not been restarted: the counters doesn't seem to be reset. Based on jira: https://freeswitch.org/jira/browse/FS-10572 Currently i've set a new parameter on switch.conf: So far FS didn't fork to another child process, still waiting a few days to validate that this options solve my issue. On 18/11/2018 00:01, David Villasmil wrote: > Also, regarding memory usage, that's how FS works. It takes memory AS > NEEDED and simply doesn't returns it, it holds it for future use. > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > ᐧ > > On Sat, Nov 17, 2018 at 11:00 PM David Villasmil > > wrote: > > I've never seen this... > > Is it possible the process is being started multiple times? You > _can_ run fs multiple times, but only the first would start > properly as the rest would try to bind to a port already in use by > the first process. > There's a parameter on all profiles to shutdown if it can't start, > try setting that and see what happens. > You should also check your crontab... This is NOT normal FS > behaviour, as far as i know. > > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > ᐧ > > On Fri, Nov 9, 2018 at 7:48 PM António Silva via FreeSWITCH-users > > wrote: > > > > > ---------- Forwarded message ---------- > From: "António Silva" > > To: FreeSWITCH Users Help > > > Cc: > Bcc: > Date: Thu, 8 Nov 2018 17:53:03 +0100 > Subject: Re: Memory increase because of multiple instances of > freeswitch > Forget to put my unit configuration: > > [Service] > ; service > Type=forking > PIDFile=/run/freeswitch/freeswitch.pid > ExecStart=/usr/bin/freeswitch -ncwait -nonat -scripts /scripts/fs > TimeoutSec=300s > Restart=on-failure > RestartSec=500ms > ; exec > User=root > Group=daemon > LimitCORE=infinity > LimitNOFILE=100000 > LimitNPROC=60000 > LimitSTACK=250000 > LimitRTPRIO=infinity > LimitRTTIME=infinity > IOSchedulingClass=realtime > IOSchedulingPriority=2 > CPUSchedulingPolicy=rr > CPUSchedulingPriority=89 > UMask=0007 > > > > On 08/11/2018 17:11, António Silva wrote: > > Hi all, > > > > I notice a strange behaviour on machine due to increase of > memory, > > when i went to see that was the process consuming the memory > i notice > > that freeswitch have multiple  process running: > > > > root      2543 90.4  4.8 2178716 1587484 ? S > /usr/bin/freeswitch -ncwait -nonat -scripts > /opt/commsmundi/scripts/fs > > root      7858  0.0  5.0 2289392 1649484 ?     SN Oct19   > 0:00 \_ > > /usr/bin/freeswitch -ncwait -nonat -scripts > /opt/commsmundi/scripts/fs > > root     30626  0.0  4.4 2172632 1464684 ?     SN Oct23   > 0:00 \_ > > /usr/bin/freeswitch -ncwait -nonat -scripts > /opt/commsmundi/scripts/fs > > root      4505  0.0  4.9 2336544 1621768 ?     SN 10:43   > 0:00 \_ > > /usr/bin/freeswitch -ncwait -nonat -scripts > /opt/commsmundi/scripts/fs > > root     22557  0.0  4.9 2336548 1636604 ?     SN 11:02   > 0:00 \_ > > /usr/bin/freeswitch -ncwait -nonat -scripts > /opt/commsmundi/scripts/fs > > > > > > and today: > > > > root     21823 93.9  4.8 2154128 1581064 ? S > /usr/bin/freeswitch -ncwait -nonat -scripts > /opt/commsmundi/scripts/fs > > root     18417  0.0  4.9 2247264 1610108 ?     SN 11:09   > 0:00 \_ > > /usr/bin/freeswitch -ncwait -nonat -scripts > /opt/commsmundi/scripts/fs > > > > > > what could be the cause of this? anyone experience the same? > > > > As for the service everything is running ok, is just that > the memory > > keeps increasing.. i restart freewitch when i start to reach > my memory > > limit. > > > > > > I'm running fs 1.8.2 on debian jessie. > > > > > -- > Saludos / Regards / Cumprimentos > António Silva > > > > > > ---------- Forwarded message ---------- > From: "António Silva via FreeSWITCH-users" > > > To: FreeSWITCH Users Help > > > Cc: > Bcc: > Date: Fri, 09 Nov 2018 11:48:33 -0800 (PST) > Subject: Re: [Freeswitch-users] Memory increase because of > multiple instances of freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -- Saludos / Regards / Cumprimentos António Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Mon Nov 19 13:37:57 2018 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 19 Nov 2018 14:37:57 +0100 Subject: [Freeswitch-users] Monitoring mod_odbc_cdr In-Reply-To: References: <581ddce4-e557-c9b2-8efa-8e136b51cce3@gmail.com> Message-ID: You NEVER EVER rely on server (freeswitch|asterisk|sipproxy|etc) database connection for accounting and billing. You want to write CDRs on a reliable support, then (maybe even in semirealtime) you use CDRs as input for accounting/billing. If you have advanced needs you can use advanced (and reliable) systems like http://www.cgrates.org/ -giovanni On Fri, Nov 16, 2018 at 8:35 PM Social Boh wrote: > Hello, > > on the odbc_cdr.conf.xml, maybe you can configure: > > > > > > so if database fail you ha ve a file for each call you can process to add > to your tables. > > Regards > > --- > I'm SoCIaL, MayBe > > El 16/11/2018 a las 08:24, Sergey Safarov escribió: > > you can > 1) count call in database for last 12 hours using monitoring tools like > zabbix. If no calls then rise alarm; > 2) configure saving CDR in XML or JSON format on disk. if call is lost in > CSL database, then you cann load from CDR files. CDR files older then 2 > days may be deleted. > > > пт, 16 нояб. 2018 г. в 16:18, morfair at gmail.com : > >> Hello all! >> >> I use mod_odbc_cdr for store CDR in MySQL for billing. Today I saw that >> no records in DB in 2 weeks. IT IS FAIL!! >> >> I went to fs_cli and type `reload mod_odbc_cdr`. After that all work >> again. But I lost records in two weeks. >> >> How to monitor mod_odbc_cdr and prevent risks with its? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Nov 19 18:11:05 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 19 Nov 2018 18:11:05 +0000 Subject: [Freeswitch-users] Memory increase because of multiple instances of freeswitch In-Reply-To: References: <04fc4086-d678-2f65-ca06-059f40b89e22@wirelessmundi.com> Message-ID: Totally true, i didn't think of that... @antonio, can you try `ps -ef`? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ᐧ On Mon, Nov 19, 2018 at 3:49 PM Ken Rice wrote: > well also keep in mind that depending on how he is looking for freeswitch > processes he may be seeing multiple as the threads will show up ad their > individual psuedo PID for example in htop > > Sent from my iPhone > > On Nov 17, 2018, at 17:01, David Villasmil > wrote: > > Also, regarding memory usage, that's how FS works. It takes memory AS > NEEDED and simply doesn't returns it, it holds it for future use. > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > ᐧ > > On Sat, Nov 17, 2018 at 11:00 PM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> I've never seen this... >> >> Is it possible the process is being started multiple times? You _can_ run >> fs multiple times, but only the first would start properly as the rest >> would try to bind to a port already in use by the first process. >> There's a parameter on all profiles to shutdown if it can't start, try >> setting that and see what happens. >> You should also check your crontab... This is NOT normal FS behaviour, as >> far as i know. >> >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> >> ᐧ >> >> On Fri, Nov 9, 2018 at 7:48 PM António Silva via FreeSWITCH-users < >> freeswitch-users at lists.freeswitch.org> wrote: >> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: "António Silva" >>> To: FreeSWITCH Users Help >>> Cc: >>> Bcc: >>> Date: Thu, 8 Nov 2018 17:53:03 +0100 >>> Subject: Re: Memory increase because of multiple instances of freeswitch >>> Forget to put my unit configuration: >>> >>> [Service] >>> ; service >>> Type=forking >>> PIDFile=/run/freeswitch/freeswitch.pid >>> ExecStart=/usr/bin/freeswitch -ncwait -nonat -scripts /scripts/fs >>> TimeoutSec=300s >>> Restart=on-failure >>> RestartSec=500ms >>> ; exec >>> User=root >>> Group=daemon >>> LimitCORE=infinity >>> LimitNOFILE=100000 >>> LimitNPROC=60000 >>> LimitSTACK=250000 >>> LimitRTPRIO=infinity >>> LimitRTTIME=infinity >>> IOSchedulingClass=realtime >>> IOSchedulingPriority=2 >>> CPUSchedulingPolicy=rr >>> CPUSchedulingPriority=89 >>> UMask=0007 >>> >>> >>> >>> On 08/11/2018 17:11, António Silva wrote: >>> > Hi all, >>> > >>> > I notice a strange behaviour on machine due to increase of memory, >>> > when i went to see that was the process consuming the memory i notice >>> > that freeswitch have multiple process running: >>> > >>> > root 2543 90.4 4.8 2178716 1587484 ? S>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > root 7858 0.0 5.0 2289392 1649484 ? SN Oct19 0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > root 30626 0.0 4.4 2172632 1464684 ? SN Oct23 0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > root 4505 0.0 4.9 2336544 1621768 ? SN 10:43 0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > root 22557 0.0 4.9 2336548 1636604 ? SN 11:02 0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > >>> > >>> > and today: >>> > >>> > root 21823 93.9 4.8 2154128 1581064 ? S>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > root 18417 0.0 4.9 2247264 1610108 ? SN 11:09 0:00 \_ >>> > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs >>> > >>> > >>> > what could be the cause of this? anyone experience the same? >>> > >>> > As for the service everything is running ok, is just that the memory >>> > keeps increasing.. i restart freewitch when i start to reach my memory >>> > limit. >>> > >>> > >>> > I'm running fs 1.8.2 on debian jessie. >>> > >>> > >>> -- >>> Saludos / Regards / Cumprimentos >>> António Silva >>> >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: "António Silva via FreeSWITCH-users" < >>> freeswitch-users at lists.freeswitch.org> >>> To: FreeSWITCH Users Help >>> Cc: >>> Bcc: >>> Date: Fri, 09 Nov 2018 11:48:33 -0800 (PST) >>> Subject: Re: [Freeswitch-users] Memory increase because of multiple >>> instances of freeswitch >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From ryharris at airmail.cc Mon Nov 19 23:19:04 2018 From: ryharris at airmail.cc (Ryan Harris) Date: Mon, 19 Nov 2018 18:19:04 -0500 Subject: [Freeswitch-users] Generate uniqueid like asterisk In-Reply-To: References: Message-ID: On 11/16/18 3:29 AM, Mickael Hubert wrote: > Hi all, > currently, I use asterisk uniqueid to lot of internals processes Ex: > 001-1542296620.1319230 > I want to change asterisk by FS, but I have to keep this uniqueid format. > I know the format of the uniqueid : [server id]-[timestamp].[random > integer] > > How can I generate it from FS please ? Touch and edit /usr/share/freeswitch/scripts/uniqueid.lua Write this to the file: local server_id = argv[1] or '001' -- defaults system id to '001' if you don't pass an arg stream:write(server_id .. '-' .. os.time() .. '.' .. math.random(1000000,9999999)) Then use this in your XML dialplan: Now you have a UNIQUEID channel variable in the format you requested. From maarten at youreal.eu Tue Nov 20 08:28:37 2018 From: maarten at youreal.eu (Maarten Ureel) Date: Tue, 20 Nov 2018 08:28:37 +0000 Subject: [Freeswitch-users] scratch sound at beginning of SRTP call Message-ID: Hello I am running a setup where calls from the outside are not encrypted with SRTP (from our SIP trunk), and locally (between phone & FS) they are. I put in the vars.xml: This all works fine, however with one disturbing fact; when you pick up the call you hear a loud "static" noise. I think this is the phase where the suite is negotiated, because it's the same as when we force SRTP on FreeSWITCH and disable it in the phone (but then the call fails afterwards of course). So what I would need to accomplish: * Either the phone needs to negotiate the SRTP before it's answered (is that even possible?), e.g. during ring * If the negotiation has to take place after answering, that it doesn't generate this loud noise in the meantime. It could also be due to the used phone (Yealink, T42G and T46G). With RTX phones we didn't have this issue yet but I only tested it with one device. But if someone has some pointers on what causes this... Thanks for your input! Regards Maarten -------------- next part -------------- An HTML attachment was scrubbed... URL: From michael at mailworks.org Tue Nov 20 15:22:12 2018 From: michael at mailworks.org (Michael Avers) Date: Tue, 20 Nov 2018 08:22:12 -0700 Subject: [Freeswitch-users] Using multiple SSL certificates In-Reply-To: References: Message-ID: <1542727332.2152142.1583296712.228C93B6@webmail.messagingengine.com> Hello, Is there a way to use more than one SSL certificate and have TLS enabled across multiple domains? Is that something that I would need to use a separate SIP profile for? Thank you, Mike From mickael at winlux.fr Tue Nov 20 16:06:54 2018 From: mickael at winlux.fr (Mickael Hubert) Date: Tue, 20 Nov 2018 17:06:54 +0100 Subject: [Freeswitch-users] Generate uniqueid like asterisk In-Reply-To: References: Message-ID: HI Ryan, thanks a lot ! it works like a charm. ++ Le mar. 20 nov. 2018 à 15:40, Ryan Harris a écrit : > On 11/16/18 3:29 AM, Mickael Hubert wrote: > > Hi all, > > currently, I use asterisk uniqueid to lot of internals processes Ex: > > 001-1542296620.1319230 > > I want to change asterisk by FS, but I have to keep this uniqueid format. > > I know the format of the uniqueid : [server id]-[timestamp].[random > > integer] > > > > How can I generate it from FS please ? > > Touch and edit /usr/share/freeswitch/scripts/uniqueid.lua > > Write this to the file: > > local server_id = argv[1] or '001' -- defaults system id to '001' if you > don't pass an arg > stream:write(server_id .. '-' .. os.time() .. '.' .. > math.random(1000000,9999999)) > > Then use this in your XML dialplan: > > > > > > Now you have a UNIQUEID channel variable in the format you requested. > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Tue Nov 20 16:23:42 2018 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Tue, 20 Nov 2018 17:23:42 +0100 Subject: [Freeswitch-users] Memory increase because of multiple instances of freeswitch In-Reply-To: References: <04fc4086-d678-2f65-ca06-059f40b89e22@wirelessmundi.com> Message-ID: <3362de9e-1e03-85f4-b1b6-3721a16d81a6@wirelessmundi.com> Hi, I was getting the process list from the command: "ps auxf | grep freeswitch" Still waiting to see it again on the server, i can't reproduce it, when it happens again i post here the output. Not sure if enabling threaded_system_exec solved the issue. On 19/11/2018 19:11, David Villasmil wrote: > Totally true, i didn't think of that... @antonio, can you try `ps -ef`? > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > ᐧ > > On Mon, Nov 19, 2018 at 3:49 PM Ken Rice > wrote: > > well also keep in mind that depending on how he is looking for > freeswitch processes he may be seeing multiple as the threads will > show up ad their individual psuedo PID for example in htop > > Sent from my iPhone > > On Nov 17, 2018, at 17:01, David Villasmil > > wrote: > >> Also, regarding memory usage, that's how FS works. It takes >> memory AS NEEDED and simply doesn't returns it, it holds it for >> future use. >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> >> phone: +34669448337 >> >> ᐧ >> >> On Sat, Nov 17, 2018 at 11:00 PM David Villasmil >> > > wrote: >> >> I've never seen this... >> >> Is it possible the process is being started multiple times? >> You _can_ run fs multiple times, but only the first would >> start properly as the rest would try to bind to a port >> already in use by the first process. >> There's a parameter on all profiles to shutdown if it can't >> start, try setting that and see what happens. >> You should also check your crontab... This is NOT normal FS >> behaviour, as far as i know. >> >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> >> phone: +34669448337 >> >> ᐧ >> >> On Fri, Nov 9, 2018 at 7:48 PM António Silva via >> FreeSWITCH-users > > wrote: >> >> >> >> >> ---------- Forwarded message ---------- >> From: "António Silva" > > >> To: FreeSWITCH Users Help >> > > >> Cc: >> Bcc: >> Date: Thu, 8 Nov 2018 17:53:03 +0100 >> Subject: Re: Memory increase because of multiple >> instances of freeswitch >> Forget to put my unit configuration: >> >> [Service] >> ; service >> Type=forking >> PIDFile=/run/freeswitch/freeswitch.pid >> ExecStart=/usr/bin/freeswitch -ncwait -nonat -scripts >> /scripts/fs >> TimeoutSec=300s >> Restart=on-failure >> RestartSec=500ms >> ; exec >> User=root >> Group=daemon >> LimitCORE=infinity >> LimitNOFILE=100000 >> LimitNPROC=60000 >> LimitSTACK=250000 >> LimitRTPRIO=infinity >> LimitRTTIME=infinity >> IOSchedulingClass=realtime >> IOSchedulingPriority=2 >> CPUSchedulingPolicy=rr >> CPUSchedulingPriority=89 >> UMask=0007 >> >> >> >> On 08/11/2018 17:11, António Silva wrote: >> > Hi all, >> > >> > I notice a strange behaviour on machine due to increase >> of memory, >> > when i went to see that was the process consuming the >> memory i notice >> > that freeswitch have multiple  process running: >> > >> > root      2543 90.4  4.8 2178716 1587484 ?     S> Sep27 36619:46 >> > /usr/bin/freeswitch -ncwait -nonat -scripts >> /opt/commsmundi/scripts/fs >> > root      7858  0.0  5.0 2289392 1649484 ?     SN   >> Oct19   0:00 \_ >> > /usr/bin/freeswitch -ncwait -nonat -scripts >> /opt/commsmundi/scripts/fs >> > root     30626  0.0  4.4 2172632 1464684 ?     SN   >> Oct23   0:00 \_ >> > /usr/bin/freeswitch -ncwait -nonat -scripts >> /opt/commsmundi/scripts/fs >> > root      4505  0.0  4.9 2336544 1621768 ?     SN   >> 10:43   0:00 \_ >> > /usr/bin/freeswitch -ncwait -nonat -scripts >> /opt/commsmundi/scripts/fs >> > root     22557  0.0  4.9 2336548 1636604 ?     SN   >> 11:02   0:00 \_ >> > /usr/bin/freeswitch -ncwait -nonat -scripts >> /opt/commsmundi/scripts/fs >> > >> > >> > and today: >> > >> > root     21823 93.9  4.8 2154128 1581064 ?     S> Oct26 17781:45 >> > /usr/bin/freeswitch -ncwait -nonat -scripts >> /opt/commsmundi/scripts/fs >> > root     18417  0.0  4.9 2247264 1610108 ?     SN   >> 11:09   0:00 \_ >> > /usr/bin/freeswitch -ncwait -nonat -scripts >> /opt/commsmundi/scripts/fs >> > >> > >> > what could be the cause of this? anyone experience the >> same? >> > >> > As for the service everything is running ok, is just >> that the memory >> > keeps increasing.. i restart freewitch when i start to >> reach my memory >> > limit. >> > >> > >> > I'm running fs 1.8.2 on debian jessie. >> > >> > >> -- >> Saludos / Regards / Cumprimentos >> António Silva >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: "António Silva via FreeSWITCH-users" >> > > >> To: FreeSWITCH Users Help >> > > >> Cc: >> Bcc: >> Date: Fri, 09 Nov 2018 11:48:33 -0800 (PST) >> Subject: Re: [Freeswitch-users] Memory increase because >> of multiple instances of freeswitch >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Saludos / Regards / Cumprimentos António Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Nov 20 17:07:19 2018 From: brian at freeswitch.com (Brian West) Date: Tue, 20 Nov 2018 11:07:19 -0600 Subject: [Freeswitch-users] Using multiple SSL certificates In-Reply-To: <1542727332.2152142.1583296712.228C93B6@webmail.messagingengine.com> References: <1542727332.2152142.1583296712.228C93B6@webmail.messagingengine.com> Message-ID: Not currently. /b On Tue, Nov 20, 2018 at 10:16 AM Michael Avers wrote: > Hello, > > Is there a way to use more than one SSL certificate and have TLS enabled > across multiple domains? Is that something that I would need to use a > separate SIP profile for? > > Thank you, > Mike > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From ko at sv01.de Tue Nov 20 20:53:27 2018 From: ko at sv01.de (Kevin Olbrich) Date: Tue, 20 Nov 2018 21:53:27 +0100 Subject: [Freeswitch-users] Using multiple SSL certificates In-Reply-To: <1542727332.2152142.1583296712.228C93B6@webmail.messagingengine.com> References: <1542727332.2152142.1583296712.228C93B6@webmail.messagingengine.com> Message-ID: Hi! SNI (Server Name Indication), part of TLS could do this but I would not bet that phones actually send this. If you are able to host on different IPs, it would work with internal + external profiles: At least for 1.6.x I am not aware of multi cert for same profile, docs for 1.8.x are limited (as the users does not know what he does not know ;-) ) Maybe nginx can also solve this as a transparent proxy. I did something like this using Kamailio for a large cluster setup. Kevin Am Di., 20. Nov. 2018 um 18:30 Uhr schrieb Michael Avers < michael at mailworks.org>: > Hello, > > Is there a way to use more than one SSL certificate and have TLS enabled > across multiple domains? Is that something that I would need to use a > separate SIP profile for? > > Thank you, > Mike > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From michael at mailworks.org Tue Nov 20 21:18:56 2018 From: michael at mailworks.org (Michael Avers) Date: Tue, 20 Nov 2018 14:18:56 -0700 Subject: [Freeswitch-users] Using multiple SSL certificates In-Reply-To: References: <1542727332.2152142.1583296712.228C93B6@webmail.messagingengine.com> Message-ID: <1542748736.3172761.1583717936.3ADF3921@webmail.messagingengine.com> Hi Brian, I found the following excerpt in Confluence's TLS page - is it no longer relevant?*Multiple Profile TLS** * *If you have multiple Sofia SIP profiles, you may find yourself wanting to enable TLS support for each of the profiles. However, each may be represented to third parties using a different DNS record. In this case, simply create a new directory under /{prefix}/freeswitch/conf/ssl/ for each DNS record . Then place an agent.pem and cafile.pem into each of the directories. Point each profile to the individual directory which contains its specific agent.pem file.* Thank you, Mike On Tue, Nov 20, 2018, at 10:07 AM, Brian West wrote: > Not currently. > /b > > > On Tue, Nov 20, 2018 at 10:16 AM Michael Avers > wrote:>> Hello, >> >> Is there a way to use more than one SSL certificate and have TLS >> enabled across multiple domains? Is that something that I would need >> to use a separate SIP profile for?>> >> Thank you, >> Mike >> >> ___________________________________________________________________- >> ______>> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users>> https://freeswitch.com > > > -- > > Brian West | Co-founder and Developer > Need Commercial support? email sales at freeswitch.com > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045[1]> Email: brian at freeswitch.com > Mobile: 918-424-9378 > Website: https://www.FreeSWITCH.com[2] > https://www.facebook.com/signalwireinc?src=email > https://twitter.com/freeswitch> ___________________________________________________________________- > ________> Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users> https://freeswitch.com Links: 1. https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g 2. https://www.freeswitch.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From ko at sv01.de Tue Nov 20 22:10:17 2018 From: ko at sv01.de (Kevin Olbrich) Date: Tue, 20 Nov 2018 23:10:17 +0100 Subject: [Freeswitch-users] Using multiple SSL certificates In-Reply-To: <1542748736.3172761.1583717936.3ADF3921@webmail.messagingengine.com> References: <1542727332.2152142.1583296712.228C93B6@webmail.messagingengine.com> <1542748736.3172761.1583717936.3ADF3921@webmail.messagingengine.com> Message-ID: > *Point each profile to the individual directory which contains its specific agent.pem file.* For me, that sounds like "you can use different certs for external and internal". I don't think this means multiple certs per profile. Did you verify if your devices can use SNI? If not, this is not possible at all because the URI is sent after TLS handshake where you already have a connection while SNI does this during handshake. Kevin Am Di., 20. Nov. 2018 um 22:29 Uhr schrieb Michael Avers < michael at mailworks.org>: > Hi Brian, I found the following excerpt in Confluence's TLS page - is it > no longer relevant? > *Multiple Profile TLS* > > *If you have multiple Sofia SIP profiles, you may find yourself wanting to > enable TLS support for each of the profiles. However, each may be > represented to third parties using a different DNS record. In this case, > simply create a new directory under /{prefix}/freeswitch/conf/ssl/ for each > DNS record . Then place an agent.pem and cafile.pem into each of the > directories. Point each profile to the individual directory which contains > its specific agent.pem file.* > > Thank you, > Mike > > > > On Tue, Nov 20, 2018, at 10:07 AM, Brian West wrote: > > Not currently. > /b > > > On Tue, Nov 20, 2018 at 10:16 AM Michael Avers > wrote: > > Hello, > > Is there a way to use more than one SSL certificate and have TLS enabled > across multiple domains? Is that something that I would need to use a > separate SIP profile for? > > Thank you, > Mike > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > *_________________________________________________________________________* > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mbodbg at gmx.net Tue Nov 20 22:34:58 2018 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Tue, 20 Nov 2018 23:34:58 +0100 Subject: [Freeswitch-users] sip capture In-Reply-To: References: <545EA5E6-2EB6-4B60-92E3-FEA79E664E09@gmx.net> Message-ID: <6F954C8B-BE04-46D3-8934-B7AA92BCC1E8@gmx.net> No errors, it just not sending anything. > Am 13.11.2018 um 12:06 schrieb David Villasmil : > > Do you get any error on the fs console/log? > On Fri, 9 Nov 2018 at 19:52, Markus Bönke > wrote: > Hello, > > is sip-capture supported in 1.8.2? I’ve enabled sip capturing (as recommended at https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch ) > > > internal". I don't think this means multiple certs per profile. > > Did you verify if your devices can use SNI? If not, this is not possible > at all because the URI is sent after TLS handshake where you already have a > connection while SNI does this during handshake. > > Kevin > > Am Di., 20. Nov. 2018 um 22:29 Uhr schrieb Michael Avers < > michael at mailworks.org>: > >> Hi Brian, I found the following excerpt in Confluence's TLS page - is it >> no longer relevant? >> *Multiple Profile TLS* >> >> *If you have multiple Sofia SIP profiles, you may find yourself wanting >> to enable TLS support for each of the profiles. However, each may be >> represented to third parties using a different DNS record. In this case, >> simply create a new directory under /{prefix}/freeswitch/conf/ssl/ for each >> DNS record . Then place an agent.pem and cafile.pem into each of the >> directories. Point each profile to the individual directory which contains >> its specific agent.pem file.* >> >> Thank you, >> Mike >> >> >> >> On Tue, Nov 20, 2018, at 10:07 AM, Brian West wrote: >> >> Not currently. >> /b >> >> >> On Tue, Nov 20, 2018 at 10:16 AM Michael Avers >> wrote: >> >> Hello, >> >> Is there a way to use more than one SSL certificate and have TLS enabled >> across multiple domains? Is that something that I would need to use a >> separate SIP profile for? >> >> Thank you, >> Mike >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> >> *_________________________________________________________________________* >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From nneul at mst.edu Tue Nov 20 23:43:25 2018 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 20 Nov 2018 17:43:25 -0600 Subject: [Freeswitch-users] Using multiple SSL certificates In-Reply-To: References: <1542727332.2152142.1583296712.228C93B6@webmail.messagingengine.com> <1542748736.3172761.1583717936.3ADF3921@webmail.messagingengine.com> Message-ID: <9f1b1c13-540a-e3ef-1011-130cb639a567@mst.edu> What's the general level of device support for SANs? (i.e. cert with a whole bunch of domains listed in one certificate, like you can easily do with letsencrypt) -- Nathan -------- Original Message -------- From: Brian West Sent: Tue, Nov 20, 2018 5:32 PM CST To: FreeSWITCH Users Help Subject: [Freeswitch-users] Using multiple SSL certificates It can really only do one certificate, most devices don't work with wildcard certs so you're SOL and we don't do SNI that would require some work down in sofia. On Tue, Nov 20, 2018 at 4:36 PM Kevin Olbrich > wrote: > /Point each profile to the individual directory which contains its specific agent.pem file./ For me, that sounds like "you can use different certs for external and internal". I don't think this means multiple certs per profile. Did you verify if your devices can use SNI? If not, this is not possible at all because the URI is sent after TLS handshake where you already have a connection while SNI does this during handshake. Kevin Am Di., 20. Nov. 2018 um 22:29 Uhr schrieb Michael Avers >: __ Hi Brian, I found the following excerpt in Confluence's TLS page - is it no longer relevant? /Multiple Profile TLS// / /If you have multiple Sofia SIP profiles, you may find yourself wanting to enable TLS support for each of the profiles. However, each may be represented to third parties using a different DNS record. In this case, simply create a new directory under /{prefix}/freeswitch/conf/ssl/ for each DNS record . Then place an agent.pem and cafile.pem into each of the directories. Point each profile to the individual directory which contains its specific agent.pem file./ Thank you, Mike On Tue, Nov 20, 2018, at 10:07 AM, Brian West wrote: Not currently. /b On Tue, Nov 20, 2018 at 10:16 AM Michael Avers > wrote: Hello, Is there a way to use more than one SSL certificate and have TLS enabled across multiple domains? Is that something that I would need to use a separate SIP profile for? Thank you, Mike _________________________________________________________________________ Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com https://www.facebook.com/signalwireinc?src=email https://twitter.com/freeswitch ___________________________________________________________________________ Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com https://www.facebook.com/signalwireinc?src=email https://twitter.com/freeswitch _________________________________________________________________________ Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 341-6679 System Administrator - Architect (573) 612-1412 System and Desktop Infrastructure Team Manager From david.villasmil.work at gmail.com Wed Nov 21 00:46:18 2018 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 21 Nov 2018 00:46:18 +0000 Subject: [Freeswitch-users] sip capture In-Reply-To: <6F954C8B-BE04-46D3-8934-B7AA92BCC1E8@gmx.net> References: <545EA5E6-2EB6-4B60-92E3-FEA79E664E09@gmx.net> <6F954C8B-BE04-46D3-8934-B7AA92BCC1E8@gmx.net> Message-ID: can you execute on the fs box: ngrep -qW byline port 9060 You SHOULD see packets going out, if not, then remove that "hep=3;capture_id=2008" from the capture server and try again, just in case. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ᐧ On Tue, Nov 20, 2018 at 11:11 PM Markus Bönke wrote: > No errors, it just not sending anything. > > Am 13.11.2018 um 12:06 schrieb David Villasmil < > david.villasmil.work at gmail.com>: > > Do you get any error on the fs console/log? > On Fri, 9 Nov 2018 at 19:52, Markus Bönke wrote: > >> Hello, >> >> is sip-capture supported in 1.8.2? I’ve enabled sip capturing (as >> recommended at >> https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch >> ) >> >> >> > internal". I don't think this means multiple certs per profile. >> >> Did you verify if your devices can use SNI? If not, this is not possible >> at all because the URI is sent after TLS handshake where you already have a >> connection while SNI does this during handshake. >> >> Kevin >> >> Am Di., 20. Nov. 2018 um 22:29 Uhr schrieb Michael Avers < >> michael at mailworks.org>: >> >>> Hi Brian, I found the following excerpt in Confluence's TLS page - is it >>> no longer relevant? >>> *Multiple Profile TLS* >>> >>> *If you have multiple Sofia SIP profiles, you may find yourself wanting >>> to enable TLS support for each of the profiles. However, each may be >>> represented to third parties using a different DNS record. In this case, >>> simply create a new directory under /{prefix}/freeswitch/conf/ssl/ for each >>> DNS record . Then place an agent.pem and cafile.pem into each of the >>> directories. Point each profile to the individual directory which contains >>> its specific agent.pem file.* >>> >>> Thank you, >>> Mike >>> >>> >>> >>> On Tue, Nov 20, 2018, at 10:07 AM, Brian West wrote: >>> >>> Not currently. >>> /b >>> >>> >>> On Tue, Nov 20, 2018 at 10:16 AM Michael Avers >>> wrote: >>> >>> Hello, >>> >>> Is there a way to use more than one SSL certificate and have TLS enabled >>> across multiple domains? Is that something that I would need to use a >>> separate SIP profile for? >>> >>> Thank you, >>> Mike >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >>> >>> >>> >>> -- >>> >>> Brian West | Co-founder and Developer >>> >>> Need Commercial support? email sales at freeswitch.com >>> >>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >>> >>> >>> Email: brian at freeswitch.com >>> >>> Mobile: 918-424-9378 >>> >>> Website: https://www.FreeSWITCH.com >>> >>> [image: https://www.facebook.com/signalwireinc?src=email] >>> [image: >>> https://twitter.com/freeswitch] >>> >>> *_________________________________________________________________________* >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From moshe.rosenberg at gmail.com Wed Nov 21 14:33:27 2018 From: moshe.rosenberg at gmail.com (Moshe Rosenberg) Date: Wed, 21 Nov 2018 09:33:27 -0500 Subject: [Freeswitch-users] Effective_Caller_ID_Name set to Dialed name for internal Ext. In-Reply-To: References: Message-ID: so for example we have 2 users Sam, 101 Susan , 102 now when Sam (101) dials 102 he will see on his phone display screen "calling 102" in many other system that i worked with when a user dials an internal call the screen will display "Susan 102" any ideas or hint on how to go about it with Freeswitch -- Moshe Rosenberg Tel. 718 633 1444 Cel. 347 678 3993 www.data phone.cloud Moshe at dataphone.cloud -- Moshe Rosenberg Tel. 718 633 1444 Cel. 347 678 3993 www.data phone.cloud Moshe at dataphone.cloud -------------- next part -------------- An HTML attachment was scrubbed... URL: From nunoferreirabessa at gmail.com Wed Nov 21 15:32:52 2018 From: nunoferreirabessa at gmail.com (Nuno Bessa) Date: Wed, 21 Nov 2018 15:32:52 +0000 Subject: [Freeswitch-users] mod_verto dialplan variables Message-ID: Hi all, I'm using mod_verto with the javascript library and it's working great except for one thing I haven't quite figured out yet. 95% of our calls are outbound, from mod_verto to PSTN and my outbound dialplan is set to record all sessions. Before mod_verto, we used a desktop app developed by me, wich in turn received a variable from the dialplan on A-leg set like this: My application would store that variable along with all client details, for later supervision. Recordings are stored using YYYY/MM/DD/HH/MM/LEG-B-NUMBER_LEG-A-EXTENSION.WAV format, and I need to acess this variable, because client's clock may slightly differ from freeswitch server, and the path to the recording would be wrong when stored in MySQL. With mod_verto and the javascript library, I can't read custom variables and get them on the javascript dialog object, more specifically on dialog.params, which is the place to read them I believe. This is my complete dialplan: How can I set the verto_h_recording_filename variable in order to acess it with verto javascript client being the A-leg a mod_verto client? I searched all over but couldn't find a solution to my specific case, and documention is still a little sparse unfortunatelly. Thanks in advance for your help! -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexandr.popov at iqoption.com Wed Nov 21 17:01:27 2018 From: alexandr.popov at iqoption.com (Alexandr Popov) Date: Wed, 21 Nov 2018 19:01:27 +0200 Subject: [Freeswitch-users] mod_verto dialplan variables In-Reply-To: References: Message-ID: verto_h_ is only for incoming verto invites. why not generate filename at client side as you got any needed data at it anyway ср, 21 нояб. 2018 г. в 18:17, Nuno Bessa : > Hi all, > > I'm using mod_verto with the javascript library and it's working great > except for one thing I haven't quite figured out yet. > 95% of our calls are outbound, from mod_verto to PSTN and my outbound > dialplan is set to record all sessions. > Before mod_verto, we used a desktop app developed by me, wich in turn > received a variable from the dialplan on A-leg set like this: > > data="sip_rh_X-RecordingFilename=${strftime(%Y)}/${strftime(%m)}/${strftime(%d)}/${strftime(%H)}/${strftime(%M)}/${destination_number}_${caller_id_number}.wav"/> > > My application would store that variable along with all client details, > for later supervision. Recordings are stored using > YYYY/MM/DD/HH/MM/LEG-B-NUMBER_LEG-A-EXTENSION.WAV format, and I need to > acess this variable, because client's clock may slightly differ from > freeswitch server, and the path to the recording would be wrong when stored > in MySQL. > > With mod_verto and the javascript library, I can't read custom variables > and get them on the javascript dialog object, more specifically on > dialog.params, which is the place to read them I believe. > > This is my complete dialplan: > > > > > > > > > > data="verto_h_recording_filename=${strftime(%Y)}/${strftime(%m)}/${strftime(%d)}/${strftime(%H)}/${strftime(%M)}/${destination_number}_${caller_id_number}.wav"/> > > > > > > > How can I set the verto_h_recording_filename variable in order to acess it > with verto javascript client being the A-leg a mod_verto client? > > I searched all over but couldn't find a solution to my specific case, and > documention is still a little sparse unfortunatelly. > > Thanks in advance for your help! > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mbodbg at gmx.net Wed Nov 21 17:33:10 2018 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Wed, 21 Nov 2018 18:33:10 +0100 Subject: [Freeswitch-users] sip capture In-Reply-To: References: <545EA5E6-2EB6-4B60-92E3-FEA79E664E09@gmx.net> <6F954C8B-BE04-46D3-8934-B7AA92BCC1E8@gmx.net> Message-ID: <8FA05C7F-7435-4509-91B3-8146EA543D15@gmx.net> Hello David, ngrep -qW byline port 9060 gives me no output. Also when I remove "hep=3;capture_id=2008“ from the capture-server parameter. If I start captagent on that server I can see the output, but then I cannot trace traffic over wss. Does it work at your site with 1.8.2? Thanks and regards Markus > Am 21.11.2018 um 01:46 schrieb David Villasmil : > > can you execute on the fs box: > > ngrep -qW byline port 9060 > > You SHOULD see packets going out, if not, then remove that "hep=3;capture_id=2008" from the capture server and try again, just in case. > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > ᐧ > > On Tue, Nov 20, 2018 at 11:11 PM Markus Bönke > wrote: > No errors, it just not sending anything. > >> Am 13.11.2018 um 12:06 schrieb David Villasmil >: >> >> Do you get any error on the fs console/log? >> On Fri, 9 Nov 2018 at 19:52, Markus Bönke > wrote: >> Hello, >> >> is sip-capture supported in 1.8.2? I’ve enabled sip capturing (as recommended at https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch ) >> >> >> : > Sounds like your gateway does SIP ALG which in my experience breaks SIP in > 99 percent of all communication (because it fixes where nothing is broken). > Had such problems when I specified a STUN server for clients - NAT > traversal should be left to the server alone. It's perfectly fine to use > private IPs in SIP/SDN and let RPORT handle it all. > > Kevin > > > Am Do., 15. Nov. 2018 um 23:51 Uhr schrieb Paul Muaddib < > paul.muaddib83 at gmail.com>: > >> Hi >> >> at the moment I have the setup >> >> Internal.xml >> ext-rtp-ip = auto-nat >> ext-sip-ip = auto-nat >> >> External.xml >> ext-rtp-ip = “my_local_freeswitch_ip” >> ext-sip-ip = auto-nat >> >> RTP ports (only) are begin opened by UPNP and SIP via TCP is being keep >> open with “expire-seconds=600” and “register=true” (Is there an alternative >> for the expire-seconds ? nat-options-ping is only for endpoints registering >> to freeswitch, right?) >> >> The reason why I am doing this is, that I dont want to open up a SIP Port >> in the Firewall. >> >> But this setup up doesn’t seem right. Audio is working, inbound and >> outbound, both ways. But the echo application is not. >> >> When I change >> External.xml >> ext-rtp-ip = “auto-nat” >> >> I get one-way audio for inbound calls for the callee. The caller can be >> heard. >> >> I tried to analyses it in wireshark, but I don’t understand it. The sip >> provider keeps sending INVITE messages while freeswitch is confirming it >> with OK but keeps going on until the connection breaks. >> >> Thank you for your help >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From heedfeld at gmail.com Wed Nov 21 19:09:46 2018 From: heedfeld at gmail.com (Henning Heedfeld) Date: Wed, 21 Nov 2018 20:09:46 +0100 Subject: [Freeswitch-users] sip capture In-Reply-To: <8FA05C7F-7435-4509-91B3-8146EA543D15@gmx.net> References: <545EA5E6-2EB6-4B60-92E3-FEA79E664E09@gmx.net> <6F954C8B-BE04-46D3-8934-B7AA92BCC1E8@gmx.net> <8FA05C7F-7435-4509-91B3-8146EA543D15@gmx.net> Message-ID: Hi, I can confirm that sipcapture works with 1.8.2. Did you enable capture via console? „sofia global capture on“ Regards, Henning > Am 21.11.2018 um 18:33 schrieb Markus Bönke : > > Hello David, > > ngrep -qW byline port 9060 gives me no output. Also when I remove "hep=3;capture_id=2008“ from the capture-server parameter. If I start captagent on that server I can see the output, but then I cannot trace traffic over wss. Does it work at your site with 1.8.2? > > Thanks and regards > > Markus > > >> Am 21.11.2018 um 01:46 schrieb David Villasmil : >> >> can you execute on the fs box: >> >> ngrep -qW byline port 9060 >> >> You SHOULD see packets going out, if not, then remove that "hep=3;capture_id=2008" from the capture server and try again, just in case. >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> >> ᐧ >> >>> On Tue, Nov 20, 2018 at 11:11 PM Markus Bönke wrote: >>> No errors, it just not sending anything. >>> >>>> Am 13.11.2018 um 12:06 schrieb David Villasmil : >>>> >>>> Do you get any error on the fs console/log? >>>> On Fri, 9 Nov 2018 at 19:52, Markus Bönke wrote: >>>>> Hello, >>>>> >>>>> is sip-capture supported in 1.8.2? I’ve enabled sip capturing (as recommended at https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch) >>>>> >>>>> >>>>> captagent on that server I can see the output, but then I cannot trace > traffic over wss. Does it work at your site with 1.8.2? > > Thanks and regards > > Markus > > > Am 21.11.2018 um 01:46 schrieb David Villasmil < > david.villasmil.work at gmail.com>: > > can you execute on the fs box: > > ngrep -qW byline port 9060 > > You SHOULD see packets going out, if not, then remove that > "hep=3;capture_id=2008" from the capture server and try again, just in case. > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > ᐧ > > On Tue, Nov 20, 2018 at 11:11 PM Markus Bönke wrote: > >> No errors, it just not sending anything. >> >> Am 13.11.2018 um 12:06 schrieb David Villasmil < >> david.villasmil.work at gmail.com>: >> >> Do you get any error on the fs console/log? >> On Fri, 9 Nov 2018 at 19:52, Markus Bönke wrote: >> >>> Hello, >>> >>> is sip-capture supported in 1.8.2? I’ve enabled sip capturing (as >>> recommended at >>> https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch >>> ) >>> >>> >>> Am 21.11.2018 um 20:29 schrieb David Villasmil : > > It shouldn't matter, as long as you set the trace on the right profile. > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > ᐧ > > On Wed, Nov 21, 2018 at 6:50 PM Markus Bönke > wrote: > Hello David, > > ngrep -qW byline port 9060 gives me no output. Also when I remove "hep=3;capture_id=2008“ from the capture-server parameter. If I start captagent on that server I can see the output, but then I cannot trace traffic over wss. Does it work at your site with 1.8.2? > > Thanks and regards > > Markus > > >> Am 21.11.2018 um 01:46 schrieb David Villasmil >: >> >> can you execute on the fs box: >> >> ngrep -qW byline port 9060 >> >> You SHOULD see packets going out, if not, then remove that "hep=3;capture_id=2008" from the capture server and try again, just in case. >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> >> ᐧ >> >> On Tue, Nov 20, 2018 at 11:11 PM Markus Bönke > wrote: >> No errors, it just not sending anything. >> >>> Am 13.11.2018 um 12:06 schrieb David Villasmil >: >>> >>> Do you get any error on the fs console/log? >>> On Fri, 9 Nov 2018 at 19:52, Markus Bönke > wrote: >>> Hello, >>> >>> is sip-capture supported in 1.8.2? I’ve enabled sip capturing (as recommended at https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch ) >>> >>> >>> Am 21.11.2018 um 20:29 schrieb David Villasmil >: > > It shouldn't matter, as long as you set the trace on the right profile. > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > ᐧ > > On Wed, Nov 21, 2018 at 6:50 PM Markus Bönke > wrote: > Hello David, > > ngrep -qW byline port 9060 gives me no output. Also when I remove "hep=3;capture_id=2008“ from the capture-server parameter. If I start captagent on that server I can see the output, but then I cannot trace traffic over wss. Does it work at your site with 1.8.2? > > Thanks and regards > > Markus > > >> Am 21.11.2018 um 01:46 schrieb David Villasmil >: >> >> can you execute on the fs box: >> >> ngrep -qW byline port 9060 >> >> You SHOULD see packets going out, if not, then remove that "hep=3;capture_id=2008" from the capture server and try again, just in case. >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> >> ᐧ >> >> On Tue, Nov 20, 2018 at 11:11 PM Markus Bönke > wrote: >> No errors, it just not sending anything. >> >>> Am 13.11.2018 um 12:06 schrieb David Villasmil >: >>> >>> Do you get any error on the fs console/log? >>> On Fri, 9 Nov 2018 at 19:52, Markus Bönke > wrote: >>> Hello, >>> >>> is sip-capture supported in 1.8.2? I’ve enabled sip capturing (as recommended at https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch ) >>> >>> >>> > in many other system that i worked with when a user dials an internal call > the screen will display "Susan 102" > > any ideas or hint on how to go about it with Freeswitch > > -- > Moshe Rosenberg > Tel. 718 633 1444 > Cel. 347 678 3993 > www.data phone.cloud > Moshe at dataphone.cloud > > > > > -- > Moshe Rosenberg > Tel. 718 633 1444 > Cel. 347 678 3993 > www.data phone.cloud > Moshe at dataphone.cloud > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Fri Nov 23 08:11:22 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Fri, 23 Nov 2018 17:11:22 +0900 Subject: [Freeswitch-users] Effective_Caller_ID_Name set to Dialed name for internal Ext. In-Reply-To: References: Message-ID: Try to set *origination_callee_id_name* for both users in the Directory. Regards, Alex On Thu, Nov 22, 2018 at 12:42 AM Moshe Rosenberg wrote: > > > > > so for example we have 2 users > > Sam, 101 > Susan , 102 > > now when Sam (101) dials 102 he will see on his phone display screen > "calling 102" > > in many other system that i worked with when a user dials an internal call > the screen will display "Susan 102" > > any ideas or hint on how to go about it with Freeswitch > > -- > Moshe Rosenberg > Tel. 718 633 1444 > Cel. 347 678 3993 > www.data phone.cloud > Moshe at dataphone.cloud > > > > > -- > Moshe Rosenberg > Tel. 718 633 1444 > Cel. 347 678 3993 > www.data phone.cloud > Moshe at dataphone.cloud > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ko at sv01.de Fri Nov 23 08:44:22 2018 From: ko at sv01.de (Kevin Olbrich) Date: Fri, 23 Nov 2018 09:44:22 +0100 Subject: [Freeswitch-users] Replace User-Agent and Server headers Message-ID: Hi! How can I replace User-Agent and Server headers in freeswitch? This is some kind of hardening (why tell people which version I use if the don't need to know...). Sure, I use the latest releases but I would like to hide the version I use or that I use freeswitch at all. Kind regards Kevin From ko at sv01.de Fri Nov 23 09:09:40 2018 From: ko at sv01.de (Kevin Olbrich) Date: Fri, 23 Nov 2018 10:09:40 +0100 Subject: [Freeswitch-users] T.38 UDP port range Message-ID: Hi! Which ports do I need to open to allow T.38? The RTP range can be set but I was unable to find such a setting for T.38. Asterisk for example needs range 4000:4999. Kind regards Kevin From vma at vallimamod.org Fri Nov 23 09:18:54 2018 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Fri, 23 Nov 2018 10:18:54 +0100 Subject: [Freeswitch-users] sip capture In-Reply-To: <200CABDB-AA96-448D-BF84-DB7A7F84F003@gmx.net> References: <545EA5E6-2EB6-4B60-92E3-FEA79E664E09@gmx.net> <6F954C8B-BE04-46D3-8934-B7AA92BCC1E8@gmx.net> <8FA05C7F-7435-4509-91B3-8146EA543D15@gmx.net> <200CABDB-AA96-448D-BF84-DB7A7F84F003@gmx.net> Message-ID: Hi, The correct command for sip tracing is: sofia global siptrace on Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sip.solutions linkedin.com/in/vallimamod . > On 22 Nov 2018, at 13:24, Markus Bönke wrote: > > I also tried "sofia global capture on", it returns: > > +OK Global capture on > nua.c:575 nua_set_params() nua: nua_set_params: entering > nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params > nua.c:575 nua_set_params() nua: nua_set_params: entering > nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params > nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params > nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params > nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering > nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering > soa.c:403 soa_set_params() soa_set_params(static::0x7fd4f0001b50, ...) called > soa.c:403 soa_set_params() soa_set_params(static::0x7fd4f8001ae0, ...) called > nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK > nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK > nua_stack.c:359 nua_application_event() nua: nua_application_event: entering > nua_stack.c:359 nua_application_event() nua: nua_application_event: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > But still nothing to see in the network trace. > > I’m using version FreeSWITCH Version 1.8.2-3-a98a958~64bit (-3-a98a958 64bit) installed from the debian packages on Debian Jessie. Is it possible that capturing requires a special compiler switch, or is dependent on another freeswitch or kernel module? > > Thanks and regards > > Markus > > >> Am 21.11.2018 um 20:29 schrieb David Villasmil >: >> >> It shouldn't matter, as long as you set the trace on the right profile. >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> >> ᐧ >> >> On Wed, Nov 21, 2018 at 6:50 PM Markus Bönke > wrote: >> Hello David, >> >> ngrep -qW byline port 9060 gives me no output. Also when I remove "hep=3;capture_id=2008“ from the capture-server parameter. If I start captagent on that server I can see the output, but then I cannot trace traffic over wss. Does it work at your site with 1.8.2? >> >> Thanks and regards >> >> Markus >> >> >>> Am 21.11.2018 um 01:46 schrieb David Villasmil >: >>> >>> can you execute on the fs box: >>> >>> ngrep -qW byline port 9060 >>> >>> You SHOULD see packets going out, if not, then remove that "hep=3;capture_id=2008" from the capture server and try again, just in case. >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> >>> ᐧ >>> >>> On Tue, Nov 20, 2018 at 11:11 PM Markus Bönke > wrote: >>> No errors, it just not sending anything. >>> >>>> Am 13.11.2018 um 12:06 schrieb David Villasmil >: >>>> >>>> Do you get any error on the fs console/log? >>>> On Fri, 9 Nov 2018 at 19:52, Markus Bönke > wrote: >>>> Hello, >>>> >>>> is sip-capture supported in 1.8.2? I’ve enabled sip capturing (as recommended at https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch ) >>>> >>>> >>>> to your Sofia-sip profile. Regards, Alex On Fri, Nov 23, 2018 at 7:45 PM Kevin Olbrich wrote: > Hi! > > How can I replace User-Agent and Server headers in freeswitch? > This is some kind of hardening (why tell people which version I use if > the don't need to know...). > > Sure, I use the latest releases but I would like to hide the version I > use or that I use freeswitch at all. > > Kind regards > Kevin > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Fri Nov 23 11:39:23 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 23 Nov 2018 14:39:23 +0300 Subject: [Freeswitch-users] T.38 UDP port range In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/Firewall пт, 23 нояб. 2018 г. в 12:53, Kevin Olbrich : > Hi! > > Which ports do I need to open to allow T.38? > The RTP range can be set but I was unable to find such a setting for T.38. > > Asterisk for example needs range 4000:4999. > > Kind regards > Kevin > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mbodbg at gmx.net Fri Nov 23 12:18:05 2018 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Fri, 23 Nov 2018 13:18:05 +0100 Subject: [Freeswitch-users] sip capture In-Reply-To: References: <545EA5E6-2EB6-4B60-92E3-FEA79E664E09@gmx.net> <6F954C8B-BE04-46D3-8934-B7AA92BCC1E8@gmx.net> <8FA05C7F-7435-4509-91B3-8146EA543D15@gmx.net> <200CABDB-AA96-448D-BF84-DB7A7F84F003@gmx.net> Message-ID: <09F15DC5-62BE-42A6-A40E-D380D7711CA3@gmx.net> If you type "sofia help" in fs_cli you can see there is both: sofia global siptrace to enable/disable the sip trace sofia capture to enable/disable capturing, e.g. sending the traces to a caputue server (kamailio/homer). The Sip trace is working at my side, but no capture ... Thanks and regards Markus > Am 23.11.2018 um 10:18 schrieb Vallimamod Abdullah : > > Hi, > > The correct command for sip tracing is: sofia global siptrace on > > > Best Regards, > -- > Vallimamod Abdullah > SIP Solutions > vma at sip.solutions > linkedin.com/in/vallimamod > . > > >> On 22 Nov 2018, at 13:24, Markus Bönke > wrote: >> >> I also tried "sofia global capture on", it returns: >> >> +OK Global capture on >> nua.c:575 nua_set_params() nua: nua_set_params: entering >> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params >> nua.c:575 nua_set_params() nua: nua_set_params: entering >> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params >> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params >> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params >> nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering >> nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering >> soa.c:403 soa_set_params() soa_set_params(static::0x7fd4f0001b50, ...) called >> soa.c:403 soa_set_params() soa_set_params(static::0x7fd4f8001ae0, ...) called >> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK >> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK >> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering >> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering >> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >> >> But still nothing to see in the network trace. >> >> I’m using version FreeSWITCH Version 1.8.2-3-a98a958~64bit (-3-a98a958 64bit) installed from the debian packages on Debian Jessie. Is it possible that capturing requires a special compiler switch, or is dependent on another freeswitch or kernel module? >> >> Thanks and regards >> >> Markus >> >> >>> Am 21.11.2018 um 20:29 schrieb David Villasmil >: >>> >>> It shouldn't matter, as long as you set the trace on the right profile. >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> >>> ᐧ >>> >>> On Wed, Nov 21, 2018 at 6:50 PM Markus Bönke > wrote: >>> Hello David, >>> >>> ngrep -qW byline port 9060 gives me no output. Also when I remove "hep=3;capture_id=2008“ from the capture-server parameter. If I start captagent on that server I can see the output, but then I cannot trace traffic over wss. Does it work at your site with 1.8.2? >>> >>> Thanks and regards >>> >>> Markus >>> >>> >>>> Am 21.11.2018 um 01:46 schrieb David Villasmil >: >>>> >>>> can you execute on the fs box: >>>> >>>> ngrep -qW byline port 9060 >>>> >>>> You SHOULD see packets going out, if not, then remove that "hep=3;capture_id=2008" from the capture server and try again, just in case. >>>> >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 >>>> >>>> ᐧ >>>> >>>> On Tue, Nov 20, 2018 at 11:11 PM Markus Bönke > wrote: >>>> No errors, it just not sending anything. >>>> >>>>> Am 13.11.2018 um 12:06 schrieb David Villasmil >: >>>>> >>>>> Do you get any error on the fs console/log? >>>>> On Fri, 9 Nov 2018 at 19:52, Markus Bönke > wrote: >>>>> Hello, >>>>> >>>>> is sip-capture supported in 1.8.2? I’ve enabled sip capturing (as recommended at https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch ) >>>>> >>>>> >>>>> > >> in many other system that i worked with when a user dials an internal >> call the screen will display "Susan 102" >> >> any ideas or hint on how to go about it with Freeswitch >> >> -- >> Moshe Rosenberg >> Tel. 718 633 1444 >> Cel. 347 678 3993 >> www.data phone.cloud >> Moshe at dataphone.cloud >> >> >> >> >> -- >> Moshe Rosenberg >> Tel. 718 633 1444 >> Cel. 347 678 3993 >> www.data phone.cloud >> Moshe at dataphone.cloud >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Alex Sibyakin | Support Engineer > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > Email: alex at freeswitch.com > Website: https://www.FreeSWITCH.com > Need commercial support? Contact sales at freeswitch.com for details. > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Moshe Rosenberg Tel. 718 633 1444 Cel. 347 678 3993 www.data phone.cloud Moshe at dataphone.cloud -------------- next part -------------- An HTML attachment was scrubbed... URL: From mbodbg at gmx.net Fri Nov 23 12:34:20 2018 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Fri, 23 Nov 2018 13:34:20 +0100 Subject: [Freeswitch-users] Replace User-Agent and Server headers In-Reply-To: References: Message-ID: <94A74EA3-A0EA-4F1E-9CFF-A685C5F0CFC0@gmx.net> You can set: in the sipprofile config. > Am 23.11.2018 um 09:44 schrieb Kevin Olbrich : > > Hi! > > How can I replace User-Agent and Server headers in freeswitch? > This is some kind of hardening (why tell people which version I use if > the don't need to know...). > > Sure, I use the latest releases but I would like to hide the version I > use or that I use freeswitch at all. > > Kind regards > Kevin > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com From brian at freeswitch.com Fri Nov 23 13:13:57 2018 From: brian at freeswitch.com (Brian West) Date: Fri, 23 Nov 2018 07:13:57 -0600 Subject: [Freeswitch-users] T.38 UDP port range In-Reply-To: References: Message-ID: There is no special range it’s the same as RTP /b On Fri, Nov 23, 2018 at 7:13 AM Sergey Safarov wrote: > https://freeswitch.org/confluence/display/FREESWITCH/Firewall > > пт, 23 нояб. 2018 г. в 12:53, Kevin Olbrich : > >> Hi! >> >> Which ports do I need to open to allow T.38? >> The RTP range can be set but I was unable to find such a setting for T.38. >> >> Asterisk for example needs range 4000:4999. >> >> Kind regards >> Kevin >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From ch.chhatra at gmail.com Fri Nov 23 14:48:56 2018 From: ch.chhatra at gmail.com (Chhorm Chhatra) Date: Fri, 23 Nov 2018 21:48:56 +0700 Subject: [Freeswitch-users] Setup Verto in Local to Replace SIP Message-ID: Hi, I am not sure if this approach is still being used these days, but I'd like to know if there is any guide out there to set up Verto in Local (No SSL certificates required). My use case is that we don't want to involve with the Internet yet we still want the call to be established via Verto rather than SIP since SIP is not supported by the latest version of Android out of the box. Any help would be appreciated. Best regards, Chhatra Chhorm -------------- next part -------------- An HTML attachment was scrubbed... URL: From abaci64 at gmail.com Fri Nov 23 14:49:41 2018 From: abaci64 at gmail.com (Abaci B) Date: Fri, 23 Nov 2018 09:49:41 -0500 Subject: [Freeswitch-users] Effective_Caller_ID_Name set to Dialed name for internal Ext. In-Reply-To: References: Message-ID: Just add it to the dialplan you use to call users. On Fri, Nov 23, 2018 at 8:13 AM Moshe Rosenberg wrote: > I need this as a global setting for all users > > On Fri, Nov 23, 2018 at 5:08 AM Alexey Sibyakin > wrote: > >> Try to set *origination_callee_id_name* for both users in the Directory. >> >> Regards, >> >> Alex >> >> On Thu, Nov 22, 2018 at 12:42 AM Moshe Rosenberg < >> moshe.rosenberg at gmail.com> wrote: >> >>> >>> >>> >>> >>> so for example we have 2 users >>> >>> Sam, 101 >>> Susan , 102 >>> >>> now when Sam (101) dials 102 he will see on his phone display screen >>> "calling 102" >>> >>> in many other system that i worked with when a user dials an internal >>> call the screen will display "Susan 102" >>> >>> any ideas or hint on how to go about it with Freeswitch >>> >>> -- >>> Moshe Rosenberg >>> Tel. 718 633 1444 >>> Cel. 347 678 3993 >>> www.data phone.cloud >>> Moshe at dataphone.cloud >>> >>> >>> >>> >>> -- >>> Moshe Rosenberg >>> Tel. 718 633 1444 >>> Cel. 347 678 3993 >>> www.data phone.cloud >>> Moshe at dataphone.cloud >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> >> >> -- >> Alex Sibyakin | Support Engineer >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> Email: alex at freeswitch.com >> Website: https://www.FreeSWITCH.com >> Need commercial support? Contact sales at freeswitch.com for details. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > Moshe Rosenberg > Tel. 718 633 1444 > Cel. 347 678 3993 > www.data phone.cloud > Moshe at dataphone.cloud > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From ch.chhatra at gmail.com Fri Nov 23 14:51:00 2018 From: ch.chhatra at gmail.com (Chhorm Chhatra) Date: Fri, 23 Nov 2018 21:51:00 +0700 Subject: [Freeswitch-users] Get Active User Message-ID: Hi, I'd like to know if there is any possible way to get all active users (users that are reachable and being to be bridged to). Any help would be appreciated. Best regards, Chhatra Chhorm -------------- next part -------------- An HTML attachment was scrubbed... URL: From nunoferreirabessa at gmail.com Fri Nov 23 21:10:27 2018 From: nunoferreirabessa at gmail.com (Nuno Bessa) Date: Fri, 23 Nov 2018 21:10:27 +0000 Subject: [Freeswitch-users] mod_verto dialplan variables In-Reply-To: References: Message-ID: Hi Alexander, thanks for your reply. Actually, that's what I feared. As for your suggestion, recordings are stored in Freeswitch server, not in client's PC. So, the filename has to be defined by freeswitch dialplan itself. I actually used another approach to solve this. Freeswitch server is also a web server, so I set up a php script which returns the filename according to server time. So whenever there's an outbound call, the client queries the Freeswitch server's php script for the filename and everything works as expected. Anyway, it would be nice if mod_verto would allow what I mentioned in my first post. It would be much simpler to just define a variable in the dialplan and then access it from verto.js on verto outbound calls. Anyway, I also thought that any dialplan variable would be accessible on the SDP itself, which in turn could be somehow parsed by verto.js. Isn't that possible in any way? Nuno Bessa escreveu no dia quarta, 21/11/2018 à(s) 15:32: > Hi all, > > I'm using mod_verto with the javascript library and it's working great > except for one thing I haven't quite figured out yet. > 95% of our calls are outbound, from mod_verto to PSTN and my outbound > dialplan is set to record all sessions. > Before mod_verto, we used a desktop app developed by me, wich in turn > received a variable from the dialplan on A-leg set like this: > > data="sip_rh_X-RecordingFilename=${strftime(%Y)}/${strftime(%m)}/${strftime(%d)}/${strftime(%H)}/${strftime(%M)}/${destination_number}_${caller_id_number}.wav"/> > > My application would store that variable along with all client details, > for later supervision. Recordings are stored using > YYYY/MM/DD/HH/MM/LEG-B-NUMBER_LEG-A-EXTENSION.WAV format, and I need to > acess this variable, because client's clock may slightly differ from > freeswitch server, and the path to the recording would be wrong when stored > in MySQL. > > With mod_verto and the javascript library, I can't read custom variables > and get them on the javascript dialog object, more specifically on > dialog.params, which is the place to read them I believe. > > This is my complete dialplan: > > > > > > > > > > data="verto_h_recording_filename=${strftime(%Y)}/${strftime(%m)}/${strftime(%d)}/${strftime(%H)}/${strftime(%M)}/${destination_number}_${caller_id_number}.wav"/> > > > > > > > How can I set the verto_h_recording_filename variable in order to acess it > with verto javascript client being the A-leg a mod_verto client? > > I searched all over but couldn't find a solution to my specific case, and > documention is still a little sparse unfortunatelly. > > Thanks in advance for your help! > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Sat Nov 24 06:13:21 2018 From: gregor at infomedia.si (Gregor Nanger) Date: Sat, 24 Nov 2018 07:13:21 +0100 Subject: [Freeswitch-users] mod_verto dialplan variables In-Reply-To: References: Message-ID: Hi! verto_h works only on invite, where you can send variables to verto client and access it in Javascript. Just one thought. Try to use chat functionality of verto where you can send message from server to verto client. https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/6586512 You can subscribe on client side on incoming message event and define your own protocol in message how will you handle it in client side. It might works. Best regards, Gregor On Fri, 23 Nov 2018, 22:25 Nuno Bessa Hi Alexander, thanks for your reply. > > Actually, that's what I feared. > As for your suggestion, recordings are stored in Freeswitch server, not in > client's PC. So, the filename has to be defined by freeswitch dialplan > itself. > I actually used another approach to solve this. Freeswitch server is also > a web server, so I set up a php script which returns the filename according > to server time. > So whenever there's an outbound call, the client queries the Freeswitch > server's php script for the filename and everything works as expected. > > Anyway, it would be nice if mod_verto would allow what I mentioned in my > first post. It would be much simpler to just define a variable in the > dialplan and then access it from verto.js on verto outbound calls. Anyway, > I also thought that any dialplan variable would be accessible on the SDP > itself, which in turn could be somehow parsed by verto.js. > > Isn't that possible in any way? > > Nuno Bessa escreveu no dia quarta, > 21/11/2018 à(s) 15:32: > >> Hi all, >> >> I'm using mod_verto with the javascript library and it's working great >> except for one thing I haven't quite figured out yet. >> 95% of our calls are outbound, from mod_verto to PSTN and my outbound >> dialplan is set to record all sessions. >> Before mod_verto, we used a desktop app developed by me, wich in turn >> received a variable from the dialplan on A-leg set like this: >> >> > data="sip_rh_X-RecordingFilename=${strftime(%Y)}/${strftime(%m)}/${strftime(%d)}/${strftime(%H)}/${strftime(%M)}/${destination_number}_${caller_id_number}.wav"/> >> >> My application would store that variable along with all client details, >> for later supervision. Recordings are stored using >> YYYY/MM/DD/HH/MM/LEG-B-NUMBER_LEG-A-EXTENSION.WAV format, and I need to >> acess this variable, because client's clock may slightly differ from >> freeswitch server, and the path to the recording would be wrong when stored >> in MySQL. >> >> With mod_verto and the javascript library, I can't read custom variables >> and get them on the javascript dialog object, more specifically on >> dialog.params, which is the place to read them I believe. >> >> This is my complete dialplan: >> >> >> >> >> >> >> >> >> >> > data="verto_h_recording_filename=${strftime(%Y)}/${strftime(%m)}/${strftime(%d)}/${strftime(%H)}/${strftime(%M)}/${destination_number}_${caller_id_number}.wav"/> >> >> >> >> >> >> >> How can I set the verto_h_recording_filename variable in order to acess >> it with verto javascript client being the A-leg a mod_verto client? >> >> I searched all over but couldn't find a solution to my specific case, and >> documention is still a little sparse unfortunatelly. >> >> Thanks in advance for your help! >> > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From tiagoggsouza at gmail.com Sat Nov 24 15:03:38 2018 From: tiagoggsouza at gmail.com (=?UTF-8?Q?Tiago_Galv=C3=A3o_Gomes_de_Souza?=) Date: Sat, 24 Nov 2018 13:03:38 -0200 Subject: [Freeswitch-users] early media + sip trunk Message-ID: Hello friends, I would like to know how is the better way to block outbound calls with early media and sip trunk, I have a system for callcenter and I have a big problem with calls in voicemail entering to agents , I tried to use monitor_early_media_fail with ignore_early_media but I didn't have a sucessfull, it is possible to use this function to block it? How can i do it? My wish is to stop the call when i knew that is a voicemail call. -- Atenciosamente, Tiago Galvão Gomes de Souza. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ryandelgrosso at gmail.com Sat Nov 24 23:57:25 2018 From: ryandelgrosso at gmail.com (Ryan Delgrosso) Date: Sat, 24 Nov 2018 15:57:25 -0800 Subject: [Freeswitch-users] Call recording and intercept interplays Message-ID: Greetings all, I am looking to trigger session recording when the A leg of a call is actually bridged to an endpoint. I have been successful when using execute_on_answer or when using bridge_pre_execute_aleg when the call is answerd by the original endpoint, but when the original A leg is intercepted with the intercept application I'm not able to get any of these triggers to fire as desired. I don't want to simply begin recording without a bridge as I then just end up recording the call unnecessarily if it goes to voicemail. Can anybody give me some guidance on what callbacks are fired on a leg when it is intercepted and bridged to the intercepting leg or perhaps another way to accomplish the goal of only recording upon bridge. Thanks in advance -Ryan From vma at vallimamod.org Sun Nov 25 13:23:15 2018 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Sun, 25 Nov 2018 14:23:15 +0100 Subject: [Freeswitch-users] sip capture In-Reply-To: <09F15DC5-62BE-42A6-A40E-D380D7711CA3@gmx.net> References: <545EA5E6-2EB6-4B60-92E3-FEA79E664E09@gmx.net> <6F954C8B-BE04-46D3-8934-B7AA92BCC1E8@gmx.net> <8FA05C7F-7435-4509-91B3-8146EA543D15@gmx.net> <200CABDB-AA96-448D-BF84-DB7A7F84F003@gmx.net> <09F15DC5-62BE-42A6-A40E-D380D7711CA3@gmx.net> Message-ID: Hi, You are correct, I actually misunderstood your question as I missed the thread history about homer. Sorry for the noise. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sip.solutions linkedin.com/in/vallimamod . > On 23 Nov 2018, at 13:18, Markus Bönke wrote: > > If you type "sofia help" in fs_cli you can see there is both: > > sofia global siptrace > > to enable/disable the sip trace > > sofia capture > > to enable/disable capturing, e.g. sending the traces to a caputue server (kamailio/homer). The Sip trace is working at my side, but no capture ... > > Thanks and regards > > Markus > > >> Am 23.11.2018 um 10:18 schrieb Vallimamod Abdullah >: >> >> Hi, >> >> The correct command for sip tracing is: sofia global siptrace on >> >> >> Best Regards, >> -- >> Vallimamod Abdullah >> SIP Solutions >> vma at sip.solutions >> linkedin.com/in/vallimamod >> . >> >> >>> On 22 Nov 2018, at 13:24, Markus Bönke > wrote: >>> >>> I also tried "sofia global capture on", it returns: >>> >>> +OK Global capture on >>> nua.c:575 nua_set_params() nua: nua_set_params: entering >>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params >>> nua.c:575 nua_set_params() nua: nua_set_params: entering >>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params >>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params >>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params >>> nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering >>> nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering >>> soa.c:403 soa_set_params() soa_set_params(static::0x7fd4f0001b50, ...) called >>> soa.c:403 soa_set_params() soa_set_params(static::0x7fd4f8001ae0, ...) called >>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK >>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK >>> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering >>> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering >>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >>> >>> But still nothing to see in the network trace. >>> >>> I’m using version FreeSWITCH Version 1.8.2-3-a98a958~64bit (-3-a98a958 64bit) installed from the debian packages on Debian Jessie. Is it possible that capturing requires a special compiler switch, or is dependent on another freeswitch or kernel module? >>> >>> Thanks and regards >>> >>> Markus >>> >>> >>>> Am 21.11.2018 um 20:29 schrieb David Villasmil >: >>>> >>>> It shouldn't matter, as long as you set the trace on the right profile. >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 >>>> >>>> ᐧ >>>> >>>> On Wed, Nov 21, 2018 at 6:50 PM Markus Bönke > wrote: >>>> Hello David, >>>> >>>> ngrep -qW byline port 9060 gives me no output. Also when I remove "hep=3;capture_id=2008“ from the capture-server parameter. If I start captagent on that server I can see the output, but then I cannot trace traffic over wss. Does it work at your site with 1.8.2? >>>> >>>> Thanks and regards >>>> >>>> Markus >>>> >>>> >>>>> Am 21.11.2018 um 01:46 schrieb David Villasmil >: >>>>> >>>>> can you execute on the fs box: >>>>> >>>>> ngrep -qW byline port 9060 >>>>> >>>>> You SHOULD see packets going out, if not, then remove that "hep=3;capture_id=2008" from the capture server and try again, just in case. >>>>> >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work at gmail.com >>>>> phone: +34669448337 >>>>> >>>>> ᐧ >>>>> >>>>> On Tue, Nov 20, 2018 at 11:11 PM Markus Bönke > wrote: >>>>> No errors, it just not sending anything. >>>>> >>>>>> Am 13.11.2018 um 12:06 schrieb David Villasmil >: >>>>>> >>>>>> Do you get any error on the fs console/log? >>>>>> On Fri, 9 Nov 2018 at 19:52, Markus Bönke > wrote: >>>>>> Hello, >>>>>> >>>>>> is sip-capture supported in 1.8.2? I’ve enabled sip capturing (as recommended at https://github.com/sipcapture/homer/wiki/Examples%3A-FreeSwitch ) >>>>>> >>>>>> >>>>>> From jurijs.ivolga at gmail.com Wed Nov 28 12:27:10 2018 From: jurijs.ivolga at gmail.com (Jurijs Ivolga) Date: Wed, 28 Nov 2018 14:27:10 +0200 Subject: [Freeswitch-users] Inviting both normal and webrtc users In-Reply-To: References: Message-ID: Hi, So question is, shouldn't it be possible to offer both secure and insecure > RTP in the INVITE from FS? > Yes it is possible, it took me a while to figure out this too, please check bridge command, for DTLS/WebRTC just use: when calling and it works fine. > But once we do this, many of our other sip clients complain because they > only support/accept insecure RTP (RTP/AVP). > > So question is, shouldn't it be possible to offer both secure and insecure > RTP in the INVITE from FS? > > If not then, would it advisable/possible to not send SDP in the initial > invite from FS (we don't need early media for inbound anyway) and then wait > for the client to offer the SDP instead ? > (I'm afraid the ice process might cause a little silence interval at the > start of the call because of the late media setup and user would always > expect media to be up when answering the call) > > We really want to have generic way of handling of clients if possible, to > avoid client detection (as currently only kamailio has this info, doing the > location services) or any special config items on sip accounts. > > Best regards, > Allan > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From jurijs.ivolga at gmail.com Wed Nov 28 12:57:46 2018 From: jurijs.ivolga at gmail.com (Jurijs Ivolga) Date: Wed, 28 Nov 2018 14:57:46 +0200 Subject: [Freeswitch-users] Inviting both normal and webrtc users In-Reply-To: References: Message-ID: Hi, Sorry I bit misunderstood your question, so you can ignore my first reply. :) In my implementation I have similar problem when some clients are connecting via WebRTC and some via SIP. I solved it in a way that when calls needs to go to WebRTC, then call are routed to dialplan what is dedicated to WebRTC and plain SIP are routed to regular dialplan. This do not answer your question, but I hope it will help. :) Jurijs On Wed, Nov 28, 2018 at 2:21 PM Allan Kristensen wrote: > Hello, > > I'm trying to implement calling Webrtc clients (through kamailio proxy), > which needs ICE, DTLS, etc. in SDP and so we are adding "media_webrtc=true" > when calling and it works fine. > But once we do this, many of our other sip clients complain because they > only support/accept insecure RTP (RTP/AVP). > > So question is, shouldn't it be possible to offer both secure and insecure > RTP in the INVITE from FS? > > If not then, would it advisable/possible to not send SDP in the initial > invite from FS (we don't need early media for inbound anyway) and then wait > for the client to offer the SDP instead ? > (I'm afraid the ice process might cause a little silence interval at the > start of the call because of the late media setup and user would always > expect media to be up when answering the call) > > We really want to have generic way of handling of clients if possible, to > avoid client detection (as currently only kamailio has this info, doing the > location services) or any special config items on sip accounts. > > Best regards, > Allan > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From imfanee at gmail.com Wed Nov 28 13:18:41 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Wed, 28 Nov 2018 18:18:41 +0500 Subject: [Freeswitch-users] Conference Video Mux Option doesn't display video but Video Muted (but audio not affected) instead In-Reply-To: References: Message-ID: Hi Geeks, I am trying to implement a conference in mux mode and FreeSWITCH send canvas properly but never show video on but a pic "Video Muted (but audio not affected)" pic in place of every member's video on canvas. I tried a lot with no success :( OS : Ubuntu 14.04.5 LTS trusty FreeSWITCH Version 1.9.0+git~20181120T210412Z~968c76b29c~64bit (git 968c76b 2018-11-20 21:04:12Z 64bit) My conference profile is can anyone please help me. Regards, Faisal -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.witham at netsip.com.au Wed Nov 28 23:14:44 2018 From: david.witham at netsip.com.au (David Witham) Date: Thu, 29 Nov 2018 09:14:44 +1000 Subject: [Freeswitch-users] Access to specific B leg headers on failed bridge Message-ID: I have a problem where an upstream provider will correctly send a SIP 486 Busy Here from their voice gateway with a Reason header containing Q.850 cause=17 when a B party is genuinely busy. However if we exceed certain traffic thresholds, instead of receiving a SIP 503 with a retryable cause code, their proxy sitting in front of their voice switch sends us a SIP 486 Busy Here and no reason header. Different user agent and also contains a Server: header that the voice gateway responses do not. Therefore we start failing calls that we would otherwise route advance to another upstream. I want to be able to discern the difference between the two responses and route advance on the false user busy. I've tried using variations of the example LUA code found here https://freeswitch.org/confluence/display/FREESWITCH/Cause+Code+Substitution+Example but what I have found is that, of the variables I seem to have access to at this point in the call flow, sip_invite_failure_status=486 and last_bridge_hangup_cause=USER_BUSY are set the same in both cases. I think I need to be able to check for the absence of a Reason header or presence of the Server header to know that I have a false user busy. My question is then, is it possible to obtain the specific headers that are present in the B leg response after a failed bridge? I've tried addressing sip_h*, sip_bye_h* type variables but none seem to be populated. I'm guessing that I'm trying to address them in the A leg but they're only present in the B leg. Any tips or advice would be greatly appreciated. David -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Thu Nov 29 06:14:00 2018 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Thu, 29 Nov 2018 06:14:00 +0000 Subject: [Freeswitch-users] mod_verto and verto communicator on Debian 9 -> WS SETUP FAILED In-Reply-To: References: Message-ID: Hi Brian, thanks for the answer. You are right, it was the certificate. I fix it with the following links: https://freeswitch.org/confluence/display/FREESWITCH/WebRTC#WebRTC-Aquickhowtofrombkw(BrianK.West): https://manpages.debian.org/stretch/ca-certificates/update-ca-certificates.8.en.html have a nice time Alex Von: FreeSWITCH-users Im Auftrag von Brian West Gesendet: Donnerstag, 8. November 2018 17:21 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] mod_verto and verto communicator on Debian 9 -> WS SETUP FAILED Your certificate is invalid or setup wrong. On Thu, Nov 8, 2018 at 10:13 AM Alexander Haugg > wrote: Sorry, here the CLI output 2018-11-08 14:55:20.857925 [DEBUG] mod_verto.c:4250 172.16.103.22:5477 Client Connect from 172.16.103.22:5477 accepted 2018-11-08 14:55:20.857925 [DEBUG] mod_verto.c:2003 172.16.103.22:5477 Starting client thread. 2018-11-08 14:55:20.897905 [WARNING] mod_verto.c:1864 172.16.103.22:5477 WS SETUP FAILED 2018-11-08 14:55:20.897905 [DEBUG] mod_verto.c:2030 172.16.103.22:5477 Ending client thread. 2018-11-08 14:55:20.897905 [DEBUG] mod_verto.c:2038 172.16.103.22:5477 Thread ended Von: Alexander Haugg Gesendet: Donnerstag, 8. November 2018 14:47 An: 'FreeSWITCH Users Help' > Betreff: mod_verto and verto communicator on Debian 9 -> WS SETUP FAILED Hi, I try to setup the mod_verto – verto communicator scenario. The Certificate is OK “https://freeswitch.org/confluence/display/FREESWITCH/WebRTC#WebRTC-InstallCertificates” But for the sockets IP:8081 and IP:8082 I get the error message “WS SETUP FAILED” The question… Why the ws setup fails? vhosts is not configured, I am using the mini_httpd as http server. Thanks a lot Alex _________________________________________________________________________ Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- [https://hipchat.freeswitch.org/files/1/9111/w0eGOzyOVyZQdMg/email_logo.png] Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [https://www.facebook.com/signalwireinc?src=email][https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From ak at hejdu.dk Thu Nov 29 08:28:39 2018 From: ak at hejdu.dk (Allan Kristensen) Date: Thu, 29 Nov 2018 09:28:39 +0100 Subject: [Freeswitch-users] Inviting both normal and webrtc users In-Reply-To: References: Message-ID: Hello Jurijs, Thank you for your answer. So you are at the same place as us, that you require to know what client you are calling. We are trying to make a super user friendly system adding additional config items for clients is not the way we want to go. So we are considering connecting our Kamailio to our Rabbitmq (AMQP), relaying the register requests and storing this information in the database, so we can look at the User Agent string before doing the call and perform the correct invite. But maaaan, just so much complexity to overcome this (little) problem. We tried rtp_secure_media=optional, but AFAIK that's only for SDES (a=crypto) and will not work. If we find a way I will let you know :-) Have a nice day.. Allan On Wed, Nov 28, 2018 at 5:14 PM Jurijs Ivolga wrote: > Hi, > > Sorry I bit misunderstood your question, so you can ignore my first reply. > :) > > In my implementation I have similar problem when some clients are > connecting via WebRTC and some via SIP. > > I solved it in a way that when calls needs to go to WebRTC, then call are > routed to dialplan what is dedicated to WebRTC and plain SIP are routed to > regular dialplan. > > This do not answer your question, but I hope it will help. :) > > Jurijs > > On Wed, Nov 28, 2018 at 2:21 PM Allan Kristensen wrote: > >> Hello, >> >> I'm trying to implement calling Webrtc clients (through kamailio proxy), >> which needs ICE, DTLS, etc. in SDP and so we are adding "media_webrtc=true" >> when calling and it works fine. >> But once we do this, many of our other sip clients complain because they >> only support/accept insecure RTP (RTP/AVP). >> >> So question is, shouldn't it be possible to offer both secure and >> insecure RTP in the INVITE from FS? >> >> If not then, would it advisable/possible to not send SDP in the initial >> invite from FS (we don't need early media for inbound anyway) and then wait >> for the client to offer the SDP instead ? >> (I'm afraid the ice process might cause a little silence interval at the >> start of the call because of the late media setup and user would always >> expect media to be up when answering the call) >> >> We really want to have generic way of handling of clients if possible, to >> avoid client detection (as currently only kamailio has this info, doing the >> location services) or any special config items on sip accounts. >> >> Best regards, >> Allan >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From ynasida at gmail.com Thu Nov 29 11:22:43 2018 From: ynasida at gmail.com (Yuriy Nasida) Date: Thu, 29 Nov 2018 14:22:43 +0300 Subject: [Freeswitch-users] odbc issues with Deb9. how FS community members solve them ? In-Reply-To: References: Message-ID: Thank you Joel! I didn't try your advice for #1 but will keep it in mind in case of extra issues with pgsq-odbc. My issue #1 was solved by changing in postgresql.conf client_min_messages = debug5 to client_min_messages = notice One great guy - Hiroshi Inoue (pgsql-odbc community) adviced this trick. Thanks! On Wed, 28 Nov 2018 at 13:47, Joel Serrano wrote: > For #1... I would first try with debian unixodbc + latest pgsql driver > from their web. If that doesn’t work, then I would try with unixodbc latest > + latest driver. > > For both those test you will need to remove the related OS installed > packages. > > > On Tue, Nov 27, 2018 at 09:49 Yuriy Nasida wrote: > >> Thanks a lot Joel! >> >> You advice about #2 works great for me. >> >> As for #1 I am ready to provide any exta info if this will help. I >> currently think I should downgrade pgsql10 to pgsql9 because of this >> segfault issue. I also asked in pgsql-odbc community but looks like nobody >> read it :( >> >> >> >> On Tue, 27 Nov 2018 at 18:57, Joel Serrano wrote: >> >>> Not sure for #1. >>> >>> But for #2 you should download the official ODBC MySQL connector for >>> your OS (Debian/ubuntu available), that includes libmyodbc.so: >>> >>> https://dev.mysql.com/downloads/connector/odbc/ >>> >>> NOTE: you will have to register with oracle.com to download it. >>> >>> >>> Cheers, >>> Joel. >>> >>> >>> On Mon, Nov 26, 2018 at 13:53 Yuriy Nasida wrote: >>> >>>> Hi guys, >>>> >>>> I was very glad to know that Deb9 is supported officially by FS team >>>> and did some tests. Unfortunatelly I noted a lot of odbc problems with >>>> Deb9. Freeswitch is NOT the reason but I would like to ask other comunity >>>> members how they solve such problems. A lot of FS modules can not work with >>>> pgsql directly (like mod_lcr for example) and that is why I can not just >>>> stop to use odbc. >>>> >>>> Which issues ? >>>> >>>> 1) Well, 1st of them is odbc-postgresql in case I try to reach >>>> pgsql10. Please look below. I can send odbc.ini file but don't want to send >>>> too many info:) >>>> # isql -v pgsqldsn >>>> DEBUG: CommitTransaction(1) name: unnamed; blockState: STARTED; state: >>>> INPROGR, xid/subid/cid: 0/1/0 >>>> *Segmentation fault* >>>> >>>> 2) Second issue with mysql-percona. I suppose everybody knows >>>> that libmyodbc.so doesn't exist in Deb9. There is possibility to install >>>> manually odbc connector for mariadb but... what should I do for percona ? >>>> Also I saw very sad feedbacks from asterisk guys who tried to use mariadb >>>> odbc connecter. >>>> >>>> Please advce that you use to get odbc working with Deb9 >>>> Thanks. >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Services >>>> sales at freeswitch.com >>>> https://freeswitch.com >>>> >>>> Official FreeSWITCH Sites >>>> https://freeswitch.com/oss >>>> https://freeswitch.org/confluence >>>> https://cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From jurijs.ivolga at gmail.com Thu Nov 29 12:22:06 2018 From: jurijs.ivolga at gmail.com (Jurijs Ivolga) Date: Thu, 29 Nov 2018 14:22:06 +0200 Subject: [Freeswitch-users] Inviting both normal and webrtc users In-Reply-To: References: Message-ID: Hi Allan, Not sure if it might work, but I had an idea: When you need to send invite from freeswitch then you trigger a regular call with Bridge and then if it fails send other Bridge with "media_webrtc=true"(continue_on_fail). On WebRTC client you should reject call if it sees invite without DTLS and do not show anything to customer. Because first calls fails second bridge will be triggered from Freeswitch and that must have "media_webrtc=true". Again if first bridge succeed then you do not trigger second bridge with "media_webrtc=true". So for WebRTC you will always send call without DTLS and it will fails and after you send right one. I did'n check but for me it sounds like solution what might work for some cases. It is definitely not very elegant, but it might work... Off cause it adds more problems with call-logs, but again it is solvable. Jurijs On Thu, Nov 29, 2018 at 1:27 PM Allan Kristensen wrote: > Hello Jurijs, > > Thank you for your answer. So you are at the same place as us, that you > require to know what client you are calling. > We are trying to make a super user friendly system adding additional > config items for clients is not the way we want to go. > > So we are considering connecting our Kamailio to our Rabbitmq (AMQP), > relaying the register requests and storing this information in the > database, so we can look at the User Agent string before doing the call and > perform the correct invite. But maaaan, just so much complexity to overcome > this (little) problem. > > We tried rtp_secure_media=optional, but AFAIK that's only for SDES > (a=crypto) and will not work. > > If we find a way I will let you know :-) > > Have a nice day.. > Allan > > On Wed, Nov 28, 2018 at 5:14 PM Jurijs Ivolga > wrote: > >> Hi, >> >> Sorry I bit misunderstood your question, so you can ignore my first >> reply. :) >> >> In my implementation I have similar problem when some clients are >> connecting via WebRTC and some via SIP. >> >> I solved it in a way that when calls needs to go to WebRTC, then call are >> routed to dialplan what is dedicated to WebRTC and plain SIP are routed to >> regular dialplan. >> >> This do not answer your question, but I hope it will help. :) >> >> Jurijs >> >> On Wed, Nov 28, 2018 at 2:21 PM Allan Kristensen wrote: >> >>> Hello, >>> >>> I'm trying to implement calling Webrtc clients (through kamailio proxy), >>> which needs ICE, DTLS, etc. in SDP and so we are adding "media_webrtc=true" >>> when calling and it works fine. >>> But once we do this, many of our other sip clients complain because they >>> only support/accept insecure RTP (RTP/AVP). >>> >>> So question is, shouldn't it be possible to offer both secure and >>> insecure RTP in the INVITE from FS? >>> >>> If not then, would it advisable/possible to not send SDP in the initial >>> invite from FS (we don't need early media for inbound anyway) and then wait >>> for the client to offer the SDP instead ? >>> (I'm afraid the ice process might cause a little silence interval at the >>> start of the call because of the late media setup and user would always >>> expect media to be up when answering the call) >>> >>> We really want to have generic way of handling of clients if possible, >>> to avoid client detection (as currently only kamailio has this info, doing >>> the location services) or any special config items on sip accounts. >>> >>> Best regards, >>> Allan >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From steveayre at gmail.com Thu Nov 29 12:42:16 2018 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 29 Nov 2018 12:42:16 +0000 Subject: [Freeswitch-users] odbc issues with Deb9. how FS community members solve them ? In-Reply-To: References: Message-ID: Historically I've found MyODBC has always been prone to segfaults, even with threading=2. Are the latest connectors from Oracle and MariaDB any better? On Tue, 27 Nov 2018 at 15:52, Joel Serrano wrote: > Not sure for #1. > > But for #2 you should download the official ODBC MySQL connector for your > OS (Debian/ubuntu available), that includes libmyodbc.so: > > https://dev.mysql.com/downloads/connector/odbc/ > > NOTE: you will have to register with oracle.com to download it. > > > Cheers, > Joel. > > > On Mon, Nov 26, 2018 at 13:53 Yuriy Nasida wrote: > >> Hi guys, >> >> I was very glad to know that Deb9 is supported officially by FS team and >> did some tests. Unfortunatelly I noted a lot of odbc problems with Deb9. >> Freeswitch is NOT the reason but I would like to ask other comunity members >> how they solve such problems. A lot of FS modules can not work with pgsql >> directly (like mod_lcr for example) and that is why I can not just stop to >> use odbc. >> >> Which issues ? >> >> 1) Well, 1st of them is odbc-postgresql in case I try to reach pgsql10. >> Please look below. I can send odbc.ini file but don't want to send too many >> info:) >> # isql -v pgsqldsn >> DEBUG: CommitTransaction(1) name: unnamed; blockState: STARTED; state: >> INPROGR, xid/subid/cid: 0/1/0 >> *Segmentation fault* >> >> 2) Second issue with mysql-percona. I suppose everybody knows >> that libmyodbc.so doesn't exist in Deb9. There is possibility to install >> manually odbc connector for mariadb but... what should I do for percona ? >> Also I saw very sad feedbacks from asterisk guys who tried to use mariadb >> odbc connecter. >> >> Please advce that you use to get odbc working with Deb9 >> Thanks. >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mbodbg at gmx.net Thu Nov 29 13:39:26 2018 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Thu, 29 Nov 2018 14:39:26 +0100 Subject: [Freeswitch-users] Register to freeswitch with 2 IPs on the same interface Message-ID: <56A4F202-2299-4B72-92FD-4C657FF40552@gmx.net> Hello all, we have a server with 2 external IPs (eth1 .204 and eth1.1 .210 at the end). The sip profile is listening on .210 and clients are sending REGISTER requests to .210. But in table sip_registrations field hostname and orig_server_host there is the IP .204. Is there a setting to force freeswitch to store the IP where the REGISTER was sent to? Or is there an issue when the IP is configured as second IP on the same interface? Thanks and regards Markus From adevin at legos.io Thu Nov 29 16:44:42 2018 From: adevin at legos.io (Alexis DEVIN) Date: Thu, 29 Nov 2018 16:44:42 +0000 Subject: [Freeswitch-users] Diversion Header in Dialplan Message-ID: Hello freeswitch users, I’m discovering freeswitch, and I want to understand how to extract and use the variable « sip_h_Diversion » in the Dialplan. I know this question has been asked more times in the mailing-list but I didn’t understand the issue of the diverse topic who evocate this variable. Also I read the information on the wiki but I wasn’t able to know what I have to do in my case. In my Dialplan , I tried to put : to apply a modification on the Diversion Header, but this condition is unrecognized. I hope someone could explain me, if I have to extract this variable before to use it and how to simply do it ? Thanks, Bests Regards, Alexis Devin -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Nov 29 20:13:43 2018 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Nov 2018 15:13:43 -0500 Subject: [Freeswitch-users] Register to freeswitch with 2 IPs on the same interface In-Reply-To: <56A4F202-2299-4B72-92FD-4C657FF40552@gmx.net> References: <56A4F202-2299-4B72-92FD-4C657FF40552@gmx.net> Message-ID: <90E406D6-D7DA-48F8-8869-62BD28BE9058@jerris.com> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/autoload_configs/switch.conf.xml#39 > On Nov 29, 2018, at 8:39 AM, Markus Bönke wrote: > > Hello all, > > we have a server with 2 external IPs (eth1 .204 and eth1.1 .210 at the end). > The sip profile is listening on .210 and clients are sending REGISTER requests to .210. > But in table sip_registrations field hostname and orig_server_host there is the IP .204. > > Is there a setting to force freeswitch to store the IP where the REGISTER was sent to? > Or is there an issue when the IP is configured as second IP on the same interface? > > Thanks and regards > > Markus -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Nov 29 20:21:21 2018 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Nov 2018 15:21:21 -0500 Subject: [Freeswitch-users] Get Active User In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/mod_commands?focusedCommentId=16351380#mod_commands-show > On Nov 23, 2018, at 9:51 AM, Chhorm Chhatra wrote: > > Hi, > I'd like to know if there is any possible way to get all active users (users that are reachable and being to be bridged to). > Any help would be appreciated. > Best regards, > Chhatra Chhorm From mike at jerris.com Thu Nov 29 20:22:47 2018 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Nov 2018 15:22:47 -0500 Subject: [Freeswitch-users] Setup Verto in Local to Replace SIP In-Reply-To: References: Message-ID: <77108710-F3BE-4D83-86D6-229220E98FE2@jerris.com> The browsers generally require secure connections or significant extra hoops to jump through to start a call (approving connection on every call for example). Even for local you will want to use certs. > On Nov 23, 2018, at 9:48 AM, Chhorm Chhatra wrote: > > Hi, > I am not sure if this approach is still being used these days, but I'd like to know if there is any guide out there to set up Verto in Local (No SSL certificates required). > My use case is that we don't want to involve with the Internet yet we still want the call to be established via Verto rather than SIP since SIP is not supported by the latest version of Android out of the box. > Any help would be appreciated. > Best regards, > Chhatra Chhorm From mike at jerris.com Thu Nov 29 20:24:24 2018 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Nov 2018 15:24:24 -0500 Subject: [Freeswitch-users] Transfer/Merge Call In-Reply-To: References: Message-ID: <5C9D546D-33CF-415A-8B9F-57682D7CF265@jerris.com> https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-sched_transfer > On Nov 23, 2018, at 4:33 AM, Vishal Dalsania wrote: > > I have forwarded a call to another gateway and after certain period of time i want to transfer that call to an extension and hangup call to the gateway how can i do that? > > Basically an outside call lands to FreeSwitch, i forward it to another FreeSwitch at different location but after say 2 minutes i want the call to routed back to internal extension and hangup ongoing call to second FreeSwitch. > > What is the best way to achieve this? From mike at jerris.com Thu Nov 29 20:28:10 2018 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Nov 2018 15:28:10 -0500 Subject: [Freeswitch-users] Inability of External SIP Phone to Send IM (Instant Messaging) In-Reply-To: References: Message-ID: <43FB98A0-F328-4F2C-A118-D60ACC1FC324@jerris.com> https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files#SofiaConfigurationFiles-force-register-domain > On Nov 20, 2018, at 9:22 PM, yosin sun wrote: > > Dear Sir/Mme, > > I have a question concerning the inability of an external SIP phone to send IM (Instant Messaging) as follows: > > An external SIP phone (Zoiper) was able to connect to and talk with another SIP phone, but could not send IM (Instant Messaging). My FreeSWITCH version was 1.6.12. The private IP of my FreeSWITCH was 192.168.1.84 and the public IP was 220.128.137.70. When the external SIP phone (phone number 1001) sent text messages to another SIP phone, the following error message was displayed: > > sofia_presence.c:225 Can't find registered user 1001 at 220.128.137.70 > > The current solution was to modify sofia_presence.c:216 from: > > } else if (!(list = sofia_reg_find_reg_url_multi(profile, user, *host ))) { > > to: > > } else if (!(list = sofia_reg_find_reg_url_multi(profile, user, NULL))) {/*ignore host value*/ > > or another solution was for the SIP phone to register with domain, but it was too troublesome. > > Is there any solution other than modifying the source code as shown above? > > Your great help is highly appreciated. > > Best regards, > > Jack Su -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Nov 29 20:33:02 2018 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Nov 2018 15:33:02 -0500 Subject: [Freeswitch-users] RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back..... In-Reply-To: <1610858919.7688.1542416768728.JavaMail.zimbra@calgaryit.com> References: <1610858919.7688.1542416768728.JavaMail.zimbra@calgaryit.com> Message-ID: <6421F2C7-903A-46C0-98CC-A04D7069E053@jerris.com> There shouldn’t be an issue, its just a log to notify you. > On Nov 16, 2018, at 8:06 PM, George wrote: > > any way to hack around this issue the carrier is not fixing the issue, any calls to an IVR do do throug after the innitial recorded message, this is especially a problem with 311 calls, this is up in Canada with SHAW > > Thank you, George From mike at jerris.com Thu Nov 29 20:36:09 2018 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Nov 2018 15:36:09 -0500 Subject: [Freeswitch-users] dialplan xml pre-process include other xml In-Reply-To: <503907042-74331@mail.praecom.com> References: <503907042-74331@mail.praecom.com> Message-ID: <01C46FBD-53D3-47E9-B4DB-21FD0C7BD4BD@jerris.com> https://freeswitch.org/confluence/display/FREESWITCH/Understanding+the+Configuration+Files#UnderstandingtheConfigurationFiles-include > On Nov 16, 2018, at 2:40 PM, Allen Underdown wrote: > > I'm having a problem understanding the steps and/or proper formatting of including a xml file inside a "host" xml file within the dial plan. > > I have limited access to the dial plan, so need to insert a call to a file in a different directory for a customized extension. > > Example - in /dialplan/somedialplan.xml > > > >
> > > > > >
>
> > I want to add: > so that every time somedialplan.xml is > read, custom_dial.xml is included. > > I've tried adding it after the section name, and after the context, but it's never parsed (according to watching fs_cli). > > So I can't seem to figure out where to put it and have it parsed. I'm also a bit confused on exactly the XML required > in custom_dial.xml. Do I need to specify the context that I want the extension executed in? > > In custom_dial.xml I've tried using with and without the tags. I've scoured freeswitch.org but can't seem to > find the documentation in a way that I can figure this out. From mike at jerris.com Thu Nov 29 20:45:53 2018 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Nov 2018 15:45:53 -0500 Subject: [Freeswitch-users] usernames whith special simbols cant recive calls on FS 1.8 In-Reply-To: <00c001d4778c$67f9add0$37ed0970$@smartic.es> References: <00c001d4778c$67f9add0$37ed0970$@smartic.es> Message-ID: <8D79C314-B2F2-4CE9-A854-C4AEE2F4DFE5@jerris.com> This issue is cause by the patch adding in: https://freeswitch.org/jira/browse/FS-9791 And is specific to calling sip registered endpoints with ~ in the username. I’ve reached out to the original author of that patch, and if they don’t respond with a fix I’ll have to revert that patch. Please remind me if I dont follow up on that issue soon by commenting on that jira. > On Nov 8, 2018, at 12:56 PM, Miguel Jesús López Valverde wrote: > > Hello again guys: > > I'm sending this question in case someone can tell me how to solve the following problem. > > We currently have a production platform working with FS 1.6 where users have historically been created in the form username~companyname like usernames and the incoming calls to these users work correctly. Also, a request on console like “sofia_contact nameusername~companyname at domain” returns a right result. > > Now I am considering the migration of these platforms to FS 1.8 and, for this, doing an assembly of initial development. Here I have been able to verify that although these users register correctly, they can issue outgoing calls and appear correctly listed in queries of type “sofia status profile internal reg”, but these users can’t receive incoming calls, in console the information " Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] " is always obtained and when I launch the query “sofia_contact nameusername~companyname at domain” in console, I get the result “error/user_not_registered”; I only have to change the value ~ for example by a dot and the incoming calls works correctly and also the query “sofia_contact nameusername.companyname at domain” return a right result. > > I guess I need to adjust the configuration of some library of FS 1.8 and recompile it to get back an identical operation to version 1.6, can someone help me in indicating which library or with what adjustment I could get this operation again? > > Thank you very much and best regards!! > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Fri Nov 30 03:17:09 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Fri, 30 Nov 2018 12:17:09 +0900 Subject: [Freeswitch-users] Conference Video Mux Option doesn't display video but Video Muted (but audio not affected) instead In-Reply-To: References: Message-ID: Hi, Try Debian 9 and FreeSWITCH 1.8.2 from official packages. If you are going to use dev version on nonsupported OS you have to handle it yourself. Regards, Alex On Thu, Nov 29, 2018 at 1:16 AM Faisal Hanif wrote: > Hi Geeks, > > I am trying to implement a conference in mux mode and FreeSWITCH send > canvas properly but never show video on but a pic "Video Muted (but audio > not affected)" pic in place of every member's video on canvas. I tried a > lot with no success :( > > OS : Ubuntu 14.04.5 LTS trusty > FreeSWITCH Version 1.9.0+git~20181120T210412Z~968c76b29c~64bit (git > 968c76b 2018-11-20 21:04:12Z 64bit) > > My conference profile is > > > > > > > > > value="tone_stream://%(200,0,500,600,700)"/> > value="tone_stream://%(500,0,300,200,100,50,25)"/> > > > > > value="audio-always|livearray-json-status"/> > > > > > > > > > > > value="/usr/local/freeswitch/conf/images/video-muted.png"/> > value="/usr/local/freeswitch/conf/images/video-muted.png"/> > > > > can anyone please help me. > Regards, > > Faisal > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From fs at voice2net.ca Fri Nov 30 02:04:35 2018 From: fs at voice2net.ca (Darcy Primrose) Date: Thu, 29 Nov 2018 21:04:35 -0500 Subject: [Freeswitch-users] Mutli-lingual ivr Message-ID: <1dd84c16-9337-711c-c2a3-344a3774a347@voice2net.ca> We are long time freeswitch users in an area of  Canada where there is a strong english french mix.  We have for some time be providing a multi-lingual experience for the extensions/users quite successfully but have not been able to get the ivr to switch to a second language.   Has anyone had experience or success with this. We use en as our prime language code but when specific DIDs are dialed, we would like the IVR responses to be in french.  First, is this possible in freeswitch and if so, does anyone have any hints on how to accomplish this. Thanks Darcy Primrose Voice2Net From mbodbg at gmx.net Fri Nov 30 10:22:50 2018 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Fri, 30 Nov 2018 11:22:50 +0100 Subject: [Freeswitch-users] Register to freeswitch with 2 IPs on the same interface In-Reply-To: <90E406D6-D7DA-48F8-8869-62BD28BE9058@jerris.com> References: <56A4F202-2299-4B72-92FD-4C657FF40552@gmx.net> <90E406D6-D7DA-48F8-8869-62BD28BE9058@jerris.com> Message-ID: <1E0049F2-B314-42DD-B664-0946A0ED9E6F@gmx.net> setting the switchname writes the value into sip_registrations.hostname, but in sip_registrations.server_host and sip_registration.orig_server_host is still the ip from eth1 .204. I’ve swapped the IPs, so .210 is now the main IP on eth1. Now I see .210 in all fields. Thanks and regards Markus > Am 29.11.2018 um 21:13 schrieb Michael Jerris : > > https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/autoload_configs/switch.conf.xml#39 > > >> On Nov 29, 2018, at 8:39 AM, Markus Bönke > wrote: >> >> Hello all, >> >> we have a server with 2 external IPs (eth1 .204 and eth1.1 .210 at the end). >> The sip profile is listening on .210 and clients are sending REGISTER requests to .210. >> But in table sip_registrations field hostname and orig_server_host there is the IP .204. >> >> Is there a setting to force freeswitch to store the IP where the REGISTER was sent to? >> Or is there an issue when the IP is configured as second IP on the same interface? >> >> Thanks and regards >> >> Markus > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From imfanee at gmail.com Fri Nov 30 10:44:33 2018 From: imfanee at gmail.com (Faisal Hanif) Date: Fri, 30 Nov 2018 15:44:33 +0500 Subject: [Freeswitch-users] Mutli-lingual ivr In-Reply-To: <1dd84c16-9337-711c-c2a3-344a3774a347@voice2net.ca> References: <1dd84c16-9337-711c-c2a3-344a3774a347@voice2net.ca> Message-ID: This as simple you just set channel variable language={the-language-shortcode} before playing IVR and install that language sound files. On Fri, Nov 30, 2018, 3:42 PM Darcy Primrose We are long time freeswitch users in an area of Canada where there is a > strong english french mix. We have for some time be providing a > multi-lingual experience for the extensions/users quite successfully but > have not been able to get the ivr to switch to a second language. Has > anyone had experience or success with this. > > We use en as our prime language code but when specific DIDs are dialed, > we would like the IVR responses to be in french. First, is this > possible in freeswitch and if so, does anyone have any hints on how to > accomplish this. > > > Thanks > > Darcy Primrose > > Voice2Net > > > _________________________________________________________________________ > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Fri Nov 30 11:05:51 2018 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Fri, 30 Nov 2018 12:05:51 +0100 Subject: [Freeswitch-users] Diversion Header in Dialplan In-Reply-To: References: Message-ID: Hi, It is not clear what you are trying to achieve. If you want to set the diversion header on the b-leg, you need to use the export application instead of set. Also, you are using break-on-true, which requires a bridge call or an extension with continue=true to be useful. If the export does not work, can you please post your full extension and the corresponding freeswitch debug log? Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr linkedin.com/in/vallimamod . > On 29 Nov 2018, at 17:44, Alexis DEVIN wrote: > > Hello freeswitch users, > > I’m discovering freeswitch, and I want to understand how to extract and use the variable « sip_h_Diversion » in the Dialplan. > I know this question has been asked more times in the mailing-list but I didn’t understand the issue of the diverse topic who evocate this variable. > Also I read the information on the wiki but I wasn’t able to know what I have to do in my case. > > In my Dialplan , I tried to put : > > > > > > to apply a modification on the Diversion Header, but this condition is unrecognized. > I hope someone could explain me, if I have to extract this variable before to use it and how to simply do it ? > > Thanks, > Bests Regards, > Alexis Devin -------------- next part -------------- An HTML attachment was scrubbed... URL: From caioebassis at hotmail.com Fri Nov 30 12:30:51 2018 From: caioebassis at hotmail.com (Caio Assis) Date: Fri, 30 Nov 2018 12:30:51 +0000 Subject: [Freeswitch-users] Change Gateway Context to receive calls Message-ID: Hello! I have to configure a Gateway to receive calls from it and process it correctly. The problem is that I haven't found a way to change the context, and, when I receive a call from a Peer with the same IP as the Gateway, FreeSwitch ignores the authentication from the Peer and checks if the IP is allowed in ACL domains. Call comes from authenticated Peer 1000 though context CUSTOM -> OK I have to configure a Gateway which has the same IP as the Peer 1000, so I allow it in ACL. Then, all calls from Peer 1000 come through PUBLIC context. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From tiagoggsouza at gmail.com Fri Nov 30 15:33:12 2018 From: tiagoggsouza at gmail.com (=?UTF-8?Q?Tiago_Galv=C3=A3o_Gomes_de_Souza?=) Date: Fri, 30 Nov 2018 13:33:12 -0200 Subject: [Freeswitch-users] early media + sip trunk In-Reply-To: References: <8CADAD5B-2D85-4B50-9C49-BA3C6CCB3FCB@freeswitch.org> Message-ID: some can help me? Em ter, 27 de nov de 2018 às 20:36, Tiago Galvão Gomes de Souza < tiagoggsouza at gmail.com> escreveu: > I'm from brazil and here we have a lot of early media calls with machine > answer(voicemail), I want to detect early media + machine answer and hangup > this call , I read about avmd mod and it is a way to detect after answer > call and i would like to detect machine before answer, it is possible with > freeswitch? I tought that monitor_early_media_fail was a better way to do > it but I don't know how use it because when i tried to use doesn't work, > problably I'm using in a wrong way, I used AMD from asterisk to detect > Answer machine after Answer... some can help me? > > > > Em seg, 26 de nov de 2018 às 08:00, Sergey Safarov > escreveu: > >> Need to try detect voicemail tone. >> https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd >> >> Or try play announcement after answer and check voice activity. If voice >> activity is high, then voicemail >> https://github.com/seanbright/mod_amd >> >> >> пн, 26 нояб. 2018 г. в 03:34, Ken Rice : >> >>> Voicemail is not earlier media. the call is answed. early media is >>> things like ringing or other in band signaling methods like fast busy or >>> ‘not in service’ recordings. >>> >>> Sent from my iPhone >>> >>> On Nov 24, 2018, at 09:03, Tiago Galvão Gomes de Souza < >>> tiagoggsouza at gmail.com> wrote: >>> >>> Hello friends, >>> >>> I would like to know how is the better way to block outbound calls with >>> early media and sip trunk, I have a system for callcenter and I have a big >>> problem with calls in voicemail entering to agents , I tried to >>> use monitor_early_media_fail with ignore_early_media but I didn't have >>> a sucessfull, it is possible to use this function to block it? How can i do >>> it? >>> >>> My wish is to stop the call when i knew that is a voicemail call. >>> >>> >>> -- >>> Atenciosamente, >>> >>> Tiago Galvão Gomes de Souza. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Atenciosamente, > > Tiago Galvão Gomes de Souza. > -- Atenciosamente, Tiago Galvão Gomes de Souza. -------------- next part -------------- An HTML attachment was scrubbed... URL: From igor.potjevlesch at gmail.com Fri Nov 30 15:49:46 2018 From: igor.potjevlesch at gmail.com (Igor Potjevlesch) Date: Fri, 30 Nov 2018 16:49:46 +0100 Subject: [Freeswitch-users] Diversion Header in Dialplan Message-ID: Hello freeswitch users, I’m discovering freeswitch, and I want to understand how to extract and use the variable « sip_h_Diversion » in the Dialplan. I know this question has been asked more times in the mailing-list but I didn’t understand the issue of the diverse topic who evocate this variable. Also I read the information on the wiki but I wasn’t able to know what I have to do in my case. In my Dialplan , I tried to put : to apply a modification on the Diversion Header, but this condition is unrecognized. I hope someone could explain me, if I have to extract this variable before to use it and how to simply do it ? Thanks, Bests Regards, Igor -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Nov 30 18:02:58 2018 From: mike at jerris.com (Michael Jerris) Date: Fri, 30 Nov 2018 13:02:58 -0500 Subject: [Freeswitch-users] Register to freeswitch with 2 IPs on the same interface In-Reply-To: <1E0049F2-B314-42DD-B664-0946A0ED9E6F@gmx.net> References: <56A4F202-2299-4B72-92FD-4C657FF40552@gmx.net> <90E406D6-D7DA-48F8-8869-62BD28BE9058@jerris.com> <1E0049F2-B314-42DD-B664-0946A0ED9E6F@gmx.net> Message-ID: <32C14128-5A2B-4EFA-AF62-A40B731FEDCF@jerris.com> Those should match what the profile is binding to. > On Nov 30, 2018, at 5:22 AM, Markus Bönke wrote: > > setting the switchname writes the value into sip_registrations.hostname, but in sip_registrations.server_host and sip_registration.orig_server_host is still the ip from eth1 .204. I’ve swapped the IPs, so .210 is now the main IP on eth1. Now I see .210 in all fields. > > Thanks and regards > > Markus > >> Am 29.11.2018 um 21:13 schrieb Michael Jerris >: >> >> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/autoload_configs/switch.conf.xml#39 >> >> >>> On Nov 29, 2018, at 8:39 AM, Markus Bönke > wrote: >>> >>> Hello all, >>> >>> we have a server with 2 external IPs (eth1 .204 and eth1.1 .210 at the end). >>> The sip profile is listening on .210 and clients are sending REGISTER requests to .210. >>> But in table sip_registrations field hostname and orig_server_host there is the IP .204. >>> >>> Is there a setting to force freeswitch to store the IP where the REGISTER was sent to? >>> Or is there an issue when the IP is configured as second IP on the same interface? >>> >>> Thanks and regards >>> >>> Markus >> -------------- next part -------------- An HTML attachment was scrubbed... URL: